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Subject: Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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On 02/25/2014 11:23 AM, Magnus Westerlund wrote:
> On 2014-02-19 19:11, Harald Alvestrand wrote:
>> On 02/19/2014 09:16 AM, Magnus Westerlund wrote:
>>> On 2014-02-18 23:58, Harald Alvestrand wrote:
>>>> I may be a little simple-minded, but if we have a recommendation that
>>>> receivers MUST be able to receive all packetizations of G.711 and OPUS
>>>> up to 200 ms per packet, and that receivers should signal this by
>>>> sending "a=maxptime:200" in their SDP, what situations exist where
>>>> interoperability is not going to be achieved?
>>> Interoperability is going to be achieve in the direction towards this
>>> endpoint. The question is if we achieve interoperability in the
>>> direction from that endpoint.
>> So, given that there is nothing in the G.711 specification about which
>> sample sizes a G.711 receiver has to support - the right approach seems
>> to be to amend the G.711 MIME registration with this information.
> Yes, it would for the future. But considering the wide-spread deployment
> of this payload format I don't see that having any short term effect on
> the interoperability.
>
> Also, the recommendations are actually context dependent. Thus, it is
> not obvious that a single recommendation is suitable.
>
>> This is not a WebRTC issue; it is an absence of specification for the
>> non-WebRTC devices that use G.711. The RTCWEB specifications can
>> reasonably be expected to point to existing information about this
>> issue; it is not reasonable (in my mind) that the RTCWEB WG decide what
>> these values should be.
>>
> I would argue for this consideration based on that this is done in the
> WebRTC context, not any other context.

For which value of "this" is that true?

All other contexts that use G.711 send packets with a certain ptime 
(which determines the number of samples per packet).

It seems that all other contexts have managed to survive without a 
specification of which ptime it makes sense to send, except for an upper 
bound (maxptime).

When this situation occurs, usually one of two things is happening:

- There's an agreement not captured in the standard, which all 
participants follow.
- The choice doesn't matter for interoperability.

If the second is the case, I think the RTCWEB specification needs to say 
nothing.
If the first is the case, the agreement not captured in the standard 
needs to be captured in a standard. But this is not an RTCWEB standard.

        Harald




From nobody Sun Mar  2 01:38:35 2014
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Subject: Re: [rtcweb] Open data channel issues
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On 2/26/2014 2:50 AM, Magnus Westerlund wrote:
> Thanks Michael,
>
> WG, please consider these open issues and try to form your own position
> on them. They are intended to be discussed at the meeting. If you have
> proposals on how to resolve them or other input you are very welcome to
> provide that on the list.

One more big issue.  I realize this is very late for pre-meeting 
discussion; I'd hoped to hash this out earlier but for various reasons 
(including power outages and my own workload) this didn't happen.


We discussed a way to deal with the issues surrounding maximum message 
sizes at the last IETF.  Right now we have a proposal in the MMUSIC 
draft for limiting the maximum message size via the SDP.

There is a problem with this: it's at odds with the definition of 
DataChannels in the W3 and with the "duck-typing" of DataChannels to 
work largely as a superset of WebSockets (outside of channel creation), 
and the WebAPI folk at Mozilla I talked to don't like the direction 
we're taking.

I've been having talks with the WebAPI people at Mozilla, in particular 
Jonas Sicking, our WebAPI lead, and they strongly dislike encouraging 
applications to try to implement their large-data/blob transfer 
protocols; browsers have considerably more tools available to them to 
avoid memory hits and to make use of off-main-thread resources than the 
JS apps do.  "Having Send(blob) fail for any size of blob is crazy and 
non-sensical" was one comment made when I described the impacts of the 
current plan.


Manual chunking in the application means poorly-implemented congestion 
control in the app to keep the channel running efficiently (the only 
feedback available directly is either having the far-end ack at the user 
level, or trying to estimate sleep times via setTimeout() and 
bufferedAmount() (which is simply not a great solution), or simply 
dumping a large amount of smaller transfers into Send() and causing the 
browser to have to buffer them in memory).  Also GC or other pauses in 
JS execution may cause hiccups in the transfer and mis-estimation of 
available bandwidth.  And of course this is being run over a 
congestion-controlled channel in the first place.

Unless and until the W3 side makes DataChannels (and by extension, 
PeerConnection) APIs available from JS workers (and this is 
implemented), there will be compromises with packet-level protocols in 
JS.  One of those will be "it's hard to implement your own congestion 
control well".  Even with worker support, considerable extension of the 
APIs would be needed to make it work really well there.  I'll also note 
that DataChannels-from-worker support is nowhere near implementation in 
browsers.


Another BIG problem as it's currently defined is that there's no lower 
bound for this limit, so all DataChannel users will need to implement 
manual chunking even if they use small fixed messages to guarantee spec 
compliance.  Of course they won't do so...  and even if they did, they 
wouldn't test it (another big problem).  You might say "ok, fine, lets 
set some small lower bound on this value, say 2 or 4 or 16K".  That 
doesn't really help much either.  Many will send variable-sized messages 
(because it's easy), and again won't test what happens when the messages 
trip over the spec limit (or the actual browser implementation limit!)  
Those with fixed-size messages larger than the spec lower-bound won't 
test the against that; they'll test against what Firefox and Chrome 
implement at the moment.  So the net result is they'll ship applications 
that can break randomly in the field for no obvious reason (say if IE 
implements and uses 16K when Chrome used 32K and Firefox used 100MB).

Why hand the application a footgun?


Jonas Sicking suggested if the IETF insists on not supporting arbitrary 
Send(blob), we'll need to push in the W3 for a W3 protocol run on top of 
IETF DataChannels that handles fragmentation and reassembly in order to 
preserve the JS API for Send().  We can do this, but part of the whole 
partnership between the IETF and W3 on WebRTC was to try to avoid having 
the W3 define network protocols and keep them in the IETF where they belong.

Note: abandoning Send(blob) in W3 doesn't help much, as the comments I 
made above about arbitrary limits and almost-certain lack of testing of 
messages violating the negotiated size would still apply.  Send(blob) 
just makes it easier to trip over the problem (and in fact more likely 
that the application will test very large sizes).


Our options are:

A) Just accept this complexity and just hope that people write good code 
or use good libraries. (See above about testing...)
Note: we'd need to set *some* lower bound for the value.

B) Make the W3 API implementation add a level of protocol on top of the 
underlying IETF network protocol. This protocol could then deal with 
fragmenting messages on the sending side and reassembling them on the 
receiving side.

C) Convince IETF WG to support arbitrarily sized messages at a protocol 
level, at least in the WebRTC context, similar to WebSockets.


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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I'm not sure I understand the actual problem here. We *are* going to handle
arbitrary-sized messages once ndata gets done. The max-msg-size stuff is
just a solution until ndata is deployed. (To cite your options, this is A
followed by C).

That said, Send(blob) seems like a bit of a footgun to me anyway; I think
apps are going to typically avoid it, a) to avoid memory bloat, and b) to
get progress indications. The only way out of that situation is to provide
stream semantics, which seems like a lot to take on given that WebSockets
hasn't gone that route. I also agree that we shouldn't try to go the (B)
route that you mention.

So I still think manual chunking seems like a reasonable thing to support.
I don't quite get the congestion control concerns; just because there is a
max chunk size doesn't mean the impl can't buffer multiple chunks in
bufferedAmount; the app could let that fill up to a degree to avoid having
to poll constantly to prevent underrun.


On Sun, Mar 2, 2014 at 1:37 AM, Randell Jesup <randell-ietf@jesup.org>wrote:

> On 2/26/2014 2:50 AM, Magnus Westerlund wrote:
>
>> Thanks Michael,
>>
>> WG, please consider these open issues and try to form your own position
>> on them. They are intended to be discussed at the meeting. If you have
>> proposals on how to resolve them or other input you are very welcome to
>> provide that on the list.
>>
>
> One more big issue.  I realize this is very late for pre-meeting
> discussion; I'd hoped to hash this out earlier but for various reasons
> (including power outages and my own workload) this didn't happen.
>
>
> We discussed a way to deal with the issues surrounding maximum message
> sizes at the last IETF.  Right now we have a proposal in the MMUSIC draft
> for limiting the maximum message size via the SDP.
>
> There is a problem with this: it's at odds with the definition of
> DataChannels in the W3 and with the "duck-typing" of DataChannels to work
> largely as a superset of WebSockets (outside of channel creation), and the
> WebAPI folk at Mozilla I talked to don't like the direction we're taking.
>
> I've been having talks with the WebAPI people at Mozilla, in particular
> Jonas Sicking, our WebAPI lead, and they strongly dislike encouraging
> applications to try to implement their large-data/blob transfer protocols;
> browsers have considerably more tools available to them to avoid memory
> hits and to make use of off-main-thread resources than the JS apps do.
>  "Having Send(blob) fail for any size of blob is crazy and non-sensical"
> was one comment made when I described the impacts of the current plan.
>
>
> Manual chunking in the application means poorly-implemented congestion
> control in the app to keep the channel running efficiently (the only
> feedback available directly is either having the far-end ack at the user
> level, or trying to estimate sleep times via setTimeout() and
> bufferedAmount() (which is simply not a great solution), or simply dumping
> a large amount of smaller transfers into Send() and causing the browser to
> have to buffer them in memory).  Also GC or other pauses in JS execution
> may cause hiccups in the transfer and mis-estimation of available
> bandwidth.  And of course this is being run over a congestion-controlled
> channel in the first place.
>
> Unless and until the W3 side makes DataChannels (and by extension,
> PeerConnection) APIs available from JS workers (and this is implemented),
> there will be compromises with packet-level protocols in JS.  One of those
> will be "it's hard to implement your own congestion control well".  Even
> with worker support, considerable extension of the APIs would be needed to
> make it work really well there.  I'll also note that
> DataChannels-from-worker support is nowhere near implementation in browsers.
>
>
> Another BIG problem as it's currently defined is that there's no lower
> bound for this limit, so all DataChannel users will need to implement
> manual chunking even if they use small fixed messages to guarantee spec
> compliance.  Of course they won't do so...  and even if they did, they
> wouldn't test it (another big problem).  You might say "ok, fine, lets set
> some small lower bound on this value, say 2 or 4 or 16K".  That doesn't
> really help much either.  Many will send variable-sized messages (because
> it's easy), and again won't test what happens when the messages trip over
> the spec limit (or the actual browser implementation limit!)  Those with
> fixed-size messages larger than the spec lower-bound won't test the against
> that; they'll test against what Firefox and Chrome implement at the moment.
>  So the net result is they'll ship applications that can break randomly in
> the field for no obvious reason (say if IE implements and uses 16K when
> Chrome used 32K and Firefox used 100MB).
>
> Why hand the application a footgun?
>
>
> Jonas Sicking suggested if the IETF insists on not supporting arbitrary
> Send(blob), we'll need to push in the W3 for a W3 protocol run on top of
> IETF DataChannels that handles fragmentation and reassembly in order to
> preserve the JS API for Send().  We can do this, but part of the whole
> partnership between the IETF and W3 on WebRTC was to try to avoid having
> the W3 define network protocols and keep them in the IETF where they belong.
>
> Note: abandoning Send(blob) in W3 doesn't help much, as the comments I
> made above about arbitrary limits and almost-certain lack of testing of
> messages violating the negotiated size would still apply.  Send(blob) just
> makes it easier to trip over the problem (and in fact more likely that the
> application will test very large sizes).
>
>
> Our options are:
>
> A) Just accept this complexity and just hope that people write good code
> or use good libraries. (See above about testing...)
> Note: we'd need to set *some* lower bound for the value.
>
> B) Make the W3 API implementation add a level of protocol on top of the
> underlying IETF network protocol. This protocol could then deal with
> fragmenting messages on the sending side and reassembling them on the
> receiving side.
>
> C) Convince IETF WG to support arbitrarily sized messages at a protocol
> level, at least in the WebRTC context, similar to WebSockets.
>
>
>
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b675bf070f99404f3a65894
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I&#39;m not sure I understand the actual problem here. We =
*are* going to handle arbitrary-sized messages once ndata gets done. The ma=
x-msg-size stuff is just a solution until ndata is deployed. (To cite your =
options, this is A followed by C).<div>

<br></div><div>That said, Send(blob) seems like a bit of a footgun to me an=
yway; I think apps are going to typically avoid it, a) to avoid memory bloa=
t, and b) to get progress indications. The only way out of that situation i=
s to provide stream semantics, which seems like a lot to take on given that=
 WebSockets hasn&#39;t gone that route. I also agree that we shouldn&#39;t =
try to go the (B) route that you mention.</div>

<div><br></div><div>So I still think manual chunking seems like a reasonabl=
e thing to support. I don&#39;t quite get the congestion control concerns; =
just because there is a max chunk size doesn&#39;t mean the impl can&#39;t =
buffer multiple chunks in bufferedAmount; the app could let that fill up to=
 a degree to avoid having to poll constantly to prevent underrun.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Sun,=
 Mar 2, 2014 at 1:37 AM, Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@jesup.org</a>&gt=
;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 2/26/2014 2:50 AM, Magnus=
 Westerlund wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Thanks Michael,<br>
<br>
WG, please consider these open issues and try to form your own position<br>
on them. They are intended to be discussed at the meeting. If you have<br>
proposals on how to resolve them or other input you are very welcome to<br>
provide that on the list.<br>
</blockquote>
<br></div>
One more big issue. =C2=A0I realize this is very late for pre-meeting discu=
ssion; I&#39;d hoped to hash this out earlier but for various reasons (incl=
uding power outages and my own workload) this didn&#39;t happen.<br>
<br>
<br>
We discussed a way to deal with the issues surrounding maximum message size=
s at the last IETF. =C2=A0Right now we have a proposal in the MMUSIC draft =
for limiting the maximum message size via the SDP.<br>
<br>
There is a problem with this: it&#39;s at odds with the definition of DataC=
hannels in the W3 and with the &quot;duck-typing&quot; of DataChannels to w=
ork largely as a superset of WebSockets (outside of channel creation), and =
the WebAPI folk at Mozilla I talked to don&#39;t like the direction we&#39;=
re taking.<br>


<br>
I&#39;ve been having talks with the WebAPI people at Mozilla, in particular=
 Jonas Sicking, our WebAPI lead, and they strongly dislike encouraging appl=
ications to try to implement their large-data/blob transfer protocols; brow=
sers have considerably more tools available to them to avoid memory hits an=
d to make use of off-main-thread resources than the JS apps do. =C2=A0&quot=
;Having Send(blob) fail for any size of blob is crazy and non-sensical&quot=
; was one comment made when I described the impacts of the current plan.<br=
>


<br>
<br>
Manual chunking in the application means poorly-implemented congestion cont=
rol in the app to keep the channel running efficiently (the only feedback a=
vailable directly is either having the far-end ack at the user level, or tr=
ying to estimate sleep times via setTimeout() and bufferedAmount() (which i=
s simply not a great solution), or simply dumping a large amount of smaller=
 transfers into Send() and causing the browser to have to buffer them in me=
mory). =C2=A0Also GC or other pauses in JS execution may cause hiccups in t=
he transfer and mis-estimation of available bandwidth. =C2=A0And of course =
this is being run over a congestion-controlled channel in the first place.<=
br>


<br>
Unless and until the W3 side makes DataChannels (and by extension, PeerConn=
ection) APIs available from JS workers (and this is implemented), there wil=
l be compromises with packet-level protocols in JS. =C2=A0One of those will=
 be &quot;it&#39;s hard to implement your own congestion control well&quot;=
. =C2=A0Even with worker support, considerable extension of the APIs would =
be needed to make it work really well there. =C2=A0I&#39;ll also note that =
DataChannels-from-worker support is nowhere near implementation in browsers=
.<br>


<br>
<br>
Another BIG problem as it&#39;s currently defined is that there&#39;s no lo=
wer bound for this limit, so all DataChannel users will need to implement m=
anual chunking even if they use small fixed messages to guarantee spec comp=
liance. =C2=A0Of course they won&#39;t do so... =C2=A0and even if they did,=
 they wouldn&#39;t test it (another big problem). =C2=A0You might say &quot=
;ok, fine, lets set some small lower bound on this value, say 2 or 4 or 16K=
&quot;. =C2=A0That doesn&#39;t really help much either. =C2=A0Many will sen=
d variable-sized messages (because it&#39;s easy), and again won&#39;t test=
 what happens when the messages trip over the spec limit (or the actual bro=
wser implementation limit!) =C2=A0Those with fixed-size messages larger tha=
n the spec lower-bound won&#39;t test the against that; they&#39;ll test ag=
ainst what Firefox and Chrome implement at the moment. =C2=A0So the net res=
ult is they&#39;ll ship applications that can break randomly in the field f=
or no obvious reason (say if IE implements and uses 16K when Chrome used 32=
K and Firefox used 100MB).<br>


<br>
Why hand the application a footgun?<br>
<br>
<br>
Jonas Sicking suggested if the IETF insists on not supporting arbitrary Sen=
d(blob), we&#39;ll need to push in the W3 for a W3 protocol run on top of I=
ETF DataChannels that handles fragmentation and reassembly in order to pres=
erve the JS API for Send(). =C2=A0We can do this, but part of the whole par=
tnership between the IETF and W3 on WebRTC was to try to avoid having the W=
3 define network protocols and keep them in the IETF where they belong.<br>


<br>
Note: abandoning Send(blob) in W3 doesn&#39;t help much, as the comments I =
made above about arbitrary limits and almost-certain lack of testing of mes=
sages violating the negotiated size would still apply. =C2=A0Send(blob) jus=
t makes it easier to trip over the problem (and in fact more likely that th=
e application will test very large sizes).<br>


<br>
<br>
Our options are:<br>
<br>
A) Just accept this complexity and just hope that people write good code or=
 use good libraries. (See above about testing...)<br>
Note: we&#39;d need to set *some* lower bound for the value.<br>
<br>
B) Make the W3 API implementation add a level of protocol on top of the und=
erlying IETF network protocol. This protocol could then deal with fragmenti=
ng messages on the sending side and reassembling them on the receiving side=
.<br>


<br>
C) Convince IETF WG to support arbitrarily sized messages at a protocol lev=
el, at least in the WebRTC context, similar to WebSockets.<div class=3D"im =
HOEnZb"><br>
<br>
<br>
-- <br>
Randell Jesup -- rjesup a t mozilla d o t com<br>
<br></div><div class=3D"HOEnZb"><div class=3D"h5">
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--047d7b675bf070f99404f3a65894--


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Subject: [rtcweb] Proxy browsing (Was: Open data channel issues)
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--047d7b3436d239f47604f3a66fef
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Can you explain more about the specific issue for "U-C 7: Proxy browsing"?
(http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-3.2)

There are existing implementations that do this exact thing, so I don't
know if anything needs to be done on our side to support this.

On Tue, Feb 25, 2014 at 3:44 PM, Michael Tuexen <
Michael.Tuexen@lurchi.franken.de> wrote:

> Dear all,
>
> Magnus asked me to send a list of open issues regarding data channels
> to the list. Here is my current list:
>
>
> * Priority
>   The W3C hasn't defined it yet. Neither for the (S)RTP media nor for the
>   data channels. We agreed on using a non strict policy for the data
> channels
>   (some sort of wighted fair queueing). That is all.
>
> * Protocol
>   It seems not to be clear what needs to be provided when registering a
>   (sub)-protocol at IANA. And the name of the registry is unclear...
>
> * SCTP parameters.
>   There was discussed the issue how to set SCTP parameters, especially
> path.max.retrans
>   and association.max.retrans. Also HB.Interval might be of interest.
>   RFC 4060 recommends path.max.retrans=5, association.max.retrans=10, but
> has multihoming
>   in mind. To avoid the dormant state, path.max.retrans =
> association.max.retrans should be used.
>   I would suggest 10 for this value. Should HEARTBEATs be disabled?
>
> * U-C 7: Proxy browsing
>
> * Alternate CC for SCTP
>   Currently there is only the standard CC. However, in some places
> negotiation of CC is
>   mentioned.
>
> I'm currently going through the backlog of comments regarding the data
> channels
> ID and I'll try to address the issues. If I find other issues, I send an
> update
> to the above list.
>
> Best regards
> Michael
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b3436d239f47604f3a66fef
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<div dir=3D"ltr">Can you explain more about the specific issue for &quot;U-=
C 7: Proxy browsing&quot;? =C2=A0<div>(<a href=3D"http://tools.ietf.org/htm=
l/draft-ietf-rtcweb-data-channel-07#section-3.2">http://tools.ietf.org/html=
/draft-ietf-rtcweb-data-channel-07#section-3.2</a>)</div>

<div><br></div><div>There are existing implementations that do this exact t=
hing, so I don&#39;t know if anything needs to be done on our side to suppo=
rt this.<br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tu=
e, Feb 25, 2014 at 3:44 PM, Michael Tuexen <span dir=3D"ltr">&lt;<a href=3D=
"mailto:Michael.Tuexen@lurchi.franken.de" target=3D"_blank">Michael.Tuexen@=
lurchi.franken.de</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Dear all,<br>
<br>
Magnus asked me to send a list of open issues regarding data channels<br>
to the list. Here is my current list:<br>
<br>
<br>
* Priority<br>
=C2=A0 The W3C hasn&#39;t defined it yet. Neither for the (S)RTP media nor =
for the<br>
=C2=A0 data channels. We agreed on using a non strict policy for the data c=
hannels<br>
=C2=A0 (some sort of wighted fair queueing). That is all.<br>
<br>
* Protocol<br>
=C2=A0 It seems not to be clear what needs to be provided when registering =
a<br>
=C2=A0 (sub)-protocol at IANA. And the name of the registry is unclear...<b=
r>
<br>
* SCTP parameters.<br>
=C2=A0 There was discussed the issue how to set SCTP parameters, especially=
 path.max.retrans<br>
=C2=A0 and association.max.retrans. Also HB.Interval might be of interest.<=
br>
=C2=A0 RFC 4060 recommends path.max.retrans=3D5, association.max.retrans=3D=
10, but has multihoming<br>
=C2=A0 in mind. To avoid the dormant state, path.max.retrans =3D associatio=
n.max.retrans should be used.<br>
=C2=A0 I would suggest 10 for this value. Should HEARTBEATs be disabled?<br=
>
<br>
* U-C 7: Proxy browsing<br>
<br>
* Alternate CC for SCTP<br>
=C2=A0 Currently there is only the standard CC. However, in some places neg=
otiation of CC is<br>
=C2=A0 mentioned.<br>
<br>
I&#39;m currently going through the backlog of comments regarding the data =
channels<br>
ID and I&#39;ll try to address the issues. If I find other issues, I send a=
n update<br>
to the above list.<br>
<br>
Best regards<br>
Michael<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Sun, 2 Mar 2014 13:41:12 -0800
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - PeerConnection BUNDLE policy
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On Mon, Feb 24, 2014 at 9:02 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >> Section 4.1.1. defines, for the PeerConnection constructor, the
> possibility to specify BUNDLE policy.
> >>
> >> However, I am missing the possibility to indicate whether, within a
> BUNDLE group, the same port number shall be used or not.
> >>
> >> Section 5.2.1 does say:
> >>
> >>        "o  The port value is set to the port of the default ICE
> candidate for
> >>        this m= section; if this m= section is not being bundled into
> >>        another m= section, the port value MUST be unique."
> >>
> >> ...but, it doesn't talk about the case when the port IS being bundled
> into another m= section. Normally, in the initial Offer, the port value
> will be
> >> unique also in that case, but if the PeerConnection is created due to
> forking, and it is known that the remote peer supports BUNDLE, then
> >> identical port values can be used already in the initial offer.
> >
> > If a m= section is being bundled into another m= section, then the port
> number must be the same, as indicated in the BUNDLE WG draft.
>
> Before it is known whether the remote peer supports BUNDLE, each m= line
> must have a unique port number.
>

Agreed. Setting aside BUNDLE-only for now, the implementation knows in the
first offer that BUNDLE may not be supported and will use a unique port
number for m= lines in the initial offer.

>
> > I did not include a policy option to force use of the same port; the
> closest thing would be to use a policy of "all", which will mark all lines
> except the
> > first as bundle-only and achieve basically the same result, although it
> would require a second offer to update the bundle-only ports. If this is a
> > problem we could consider adding another policy value.
>
> Not sure I follow. I think the usage of bundle-only is a separate issue.
>
> The use-case I have in mind is where I want to guarantee interoperability,
> so I'd choose the "max-compat" policy.
>
> However, I do need to be able to indicate whether the m= lines should have
> the same port number or not (again, depending on whether the remote peer
> has indicated support of BUNDLE or not).
>

In the typical case, you don't need to set this; the implementation knows
to use unique ports in the initial offer and then can do the appropriate
thing in future offers (i.e., once BUNDLE support by the remote side is
known). In the case where you want to start with shared ports in the
initial offer, because you know through some mechanism that the remote side
supports this, the bundle-policy constraint could allow you to indicate
this, i.e. that you want bundling to start immediately.

I agree that a new value is needed to support this.

--001a11362cd4d392bf04f3a68819
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 24, 2014 at 9:02 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">Hi,<br>
<br>
&gt;&gt; Section 4.1.1. defines, for the PeerConnection constructor, the po=
ssibility to specify BUNDLE policy.<br>
&gt;&gt;<br>
&gt;&gt; However, I am missing the possibility to indicate whether, within =
a BUNDLE group, the same port number shall be used or not.<br>
&gt;&gt;<br>
&gt;&gt; Section 5.2.1 does say:<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;o =C2=A0The port value is set to =
the port of the default ICE candidate for<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 this m=3D section; if this m=3D section=
 is not being bundled into<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 another m=3D section, the port value MU=
ST be unique.&quot;<br>
&gt;&gt;<br>
&gt;&gt; ...but, it doesn&#39;t talk about the case when the port IS being =
bundled into another m=3D section. Normally, in the initial Offer, the port=
 value will be<br>
&gt;&gt; unique also in that case, but if the PeerConnection is created due=
 to forking, and it is known that the remote peer supports BUNDLE, then<br>
&gt;&gt; identical port values can be used already in the initial offer.<br=
>
&gt;<br>
&gt; If a m=3D section is being bundled into another m=3D section, then the=
 port number must be the same, as indicated in the BUNDLE WG draft.=C2=A0<b=
r>
<br>
</div>Before it is known whether the remote peer supports BUNDLE, each m=3D=
 line must have a unique port number.<br></blockquote><div><br></div><div>A=
greed. Setting aside BUNDLE-only for now, the implementation knows in the f=
irst offer that BUNDLE may not be supported and will use a unique port numb=
er for m=3D lines in the initial offer.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D""><br>
&gt; I did not include a policy option to force use of the same port; the c=
losest thing would be to use a policy of &quot;all&quot;, which will mark a=
ll lines except the<br>
&gt; first as bundle-only and achieve basically the same result, although i=
t would require a second offer to update the bundle-only ports. If this is =
a<br>
&gt; problem we could consider adding another policy value.<br>
<br>
</div>Not sure I follow. I think the usage of bundle-only is a separate iss=
ue.<br>
<br>
The use-case I have in mind is where I want to guarantee interoperability, =
so I&#39;d choose the &quot;max-compat&quot; policy.<br>
<br>
However, I do need to be able to indicate whether the m=3D lines should hav=
e the same port number or not (again, depending on whether the remote peer =
has indicated support of BUNDLE or not).<br></blockquote><div><br></div>

<div>In the typical case, you don&#39;t need to set this; the implementatio=
n knows to use unique ports in the initial offer and then can do the approp=
riate thing in future offers (i.e., once BUNDLE support by the remote side =
is known). In the case where you want to start with shared ports in the ini=
tial offer, because you know through some mechanism that the remote side su=
pports this, the bundle-policy constraint could allow you to indicate this,=
 i.e. that you want bundling to start immediately.</div>

<div><br></div><div>I agree that a new value is needed to support this.</di=
v></div></div></div>

--001a11362cd4d392bf04f3a68819--


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From: Justin Uberti <juberti@google.com>
Date: Sun, 2 Mar 2014 13:45:43 -0800
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On Mon, Feb 24, 2014 at 12:01 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >> But, should the application then be able to indicate which media type
> (or, even which media source) shall be considered "first"? The browser
> probably has no clue?
> >
> > I think that Justin has proposed a way to identify tracks that should
> not be bundled.  I recall no objections to this [1].  I'd assume that with
> only one track being marked that way, that track could be the one that gets
> ports and the others not.
> >
> >
> > [1]... though I can't find anything in the API spec, and we did have the
> discussion at an IETF meeting and nowhere else.  Maybe the W3C don't know
> about it yet.
>
> JSEP-06 defines different BUNDLE policies, but they are not on per
> track/m-line level.


We had a discussion in Seattle (IIRC) that the "first" track could be the
first one provided to AddTrack, and I think that is simplest. (This of
course requires the AddTrack proposal to go through.)

--001a11334c38fb909904f3a6988e
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Feb 24, 2014 at 12:01 PM, Christer Holmberg <span dir=3D"lt=
r">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">=
christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<div class=3D""><br>
&gt;&gt; But, should the application then be able to indicate which media t=
ype (or, even which media source) shall be considered &quot;first&quot;? Th=
e browser probably has no clue?<br>
&gt;<br>
&gt; I think that Justin has proposed a way to identify tracks that should =
not be bundled. =C2=A0I recall no objections to this [1]. =C2=A0I&#39;d ass=
ume that with only one track being marked that way, that track could be the=
 one that gets ports and the others not.<br>


&gt;<br>
&gt;<br>
&gt; [1]... though I can&#39;t find anything in the API spec, and we did ha=
ve the discussion at an IETF meeting and nowhere else. =C2=A0Maybe the W3C =
don&#39;t know about it yet.<br>
<br>
</div>JSEP-06 defines different BUNDLE policies, but they are not on per tr=
ack/m-line level.</blockquote><div>=C2=A0</div><div>We had a discussion in =
Seattle (IIRC) that the &quot;first&quot; track could be the first one prov=
ided to AddTrack, and I think that is simplest. (This of course requires th=
e AddTrack proposal to go through.)</div>

</div></div></div>

--001a11334c38fb909904f3a6988e--


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From: Justin Uberti <juberti@google.com>
Date: Sun, 2 Mar 2014 13:46:49 -0800
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Subject: Re: [rtcweb] JSEP-06: max-bundle policy questions
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On Mon, Feb 24, 2014 at 8:54 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> Section 4.1.1 says:
>
>
>
> =E2=80=9Co  "max-bundle": *The application will BUNDLE all of its media s=
treams*
>
> *      on a single transport.*  All streams other than the first will be
>
>       marked as bundle-only.  This policy aims to minimize candidate
>
>       gathering and maximize multiplexing, at the cost of less
>
>       compatibility with legacy endpoints.=E2=80=9D
>
>
>
> *Q1:*
>
>
>
> I think it would be good to clarify that the application will BUNDLE all
> of its media streams **THAT THE APPLICATION IS ABLE TO DE-MULTIPLEX**.
>

I don't follow what you are saying here. How would a WebRTC application end
up with streams it could not demultiplex?

>
>
> *Q2:*
>
>
>
> The text says that all streams other than the first will be marked as
> bundle-only. But, will the application have any clue which media type tha=
t
> first stream will be? Should we e.g. specify that it MUST be audio (if
> audio is enabled)?
>
>
>
> Regards,
>
>
>
> Christer
>

--001a11c3c970ec69ef04f3a69c95
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quo=
te">On Mon, Feb 24, 2014 at 8:54 AM, Christer Holmberg <span dir=3D"ltr">&l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Section 4.1.1 says:<u></u><u></=
u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=E2=80=9Co=C2=A0 &quot;max-bund=
le&quot;: <b>The application will BUNDLE all of its media streams<u></u><u>=
</u></b></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0 on a single transport.</span></b><span lang=3D"EN-US">=C2=A0 All stream=
s other than the first will be<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =
marked as bundle-only.=C2=A0 This policy aims to minimize candidate<u></u><=
u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =
gathering and maximize multiplexing, at the cost of less<u></u><u></u></spa=
n></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =
</span>compatibility with legacy endpoints.=E2=80=9D<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><b>Q1:<u></u><u></u></b></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I think it would be good to cla=
rify that the application will BUNDLE all of its media streams *<b>THAT THE=
 APPLICATION IS ABLE TO DE-MULTIPLEX</b>*.</span></p></div></div></blockquo=
te>

<div><br></div><div>I don&#39;t follow what you are saying here. How would =
a WebRTC application end up with streams it could not demultiplex?=C2=A0</d=
iv><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">

<div lang=3D"FI" link=3D"blue" vlink=3D"purple"><div><p class=3D"MsoNormal"=
><span lang=3D"EN-US">
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US">Q2:<u></u><u></u></span></b>=
</p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">The text says that all streams =
other than the first will be marked as bundle-only. But, will the applicati=
on have any clue which media type that first stream will be? Should we e.g.=
 specify that it MUST be audio (if audio
 is enabled)?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards,<span class=3D"HOEnZb">=
<font color=3D"#888888"><u></u><u></u></font></span></span></p><span class=
=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Christer<u></u><u></u></span></=
p>
</font></span></div>
</div>

</blockquote></div><br></div></div>

--001a11c3c970ec69ef04f3a69c95--


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From: Justin Uberti <juberti@google.com>
Date: Sun, 2 Mar 2014 13:49:00 -0800
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Subject: Re: [rtcweb] JSEP-06: How to create the second offer where only the port numbers are changed?
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On Mon, Feb 24, 2014 at 9:07 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> As specified in BUNDLE, before the remote peer has indicated support of
> BUNDLE, all m- lines (including those within to a bundle group) must have
> unique port numbers.
>
>
>
> Then, when the remote peer has indicated support of BUNDLE, a second offe=
r
> (called the BAS offer) needs to be sent, in which the same port number is
> assigned to all m- lines within a bundle group.
>
>
>
> My question is: how do I create the BAS offer? I don=E2=80=99t want to ch=
ange
> anything =E2=80=93 I just want to get an SDP with the BUNDLE port assigne=
d to all
> m- lines. Is there a flag in createOffer() for that?
>
>
>
The implementation knows that any offers that are created after the BUNDLE
answer should have the same port numbers. So, the app could push the
createOffer button once it receives the answer if it wanted to ensure this
happened immediately. Otherwise, there will be an onnegotiationneeded event
once ICE completes, and the offer sent at that time will have the updated
ports.

--001a11362cd4b7479204f3a6a48a
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quo=
te">On Mon, Feb 24, 2014 at 9:07 AM, Christer Holmberg <span dir=3D"ltr">&l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">





<div lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">As specified in BUNDLE, before =
the remote peer has indicated support of BUNDLE, all m- lines (including th=
ose within to a bundle group) must have unique port numbers.<u></u><u></u><=
/span></p>


<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Then, when the remote peer has =
indicated support of BUNDLE, a second offer (called the BAS offer) needs to=
 be sent, in which the same port number is assigned to all m- lines within =
a bundle group.<u></u><u></u></span></p>


<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">My question is: how do I create=
 the BAS offer? I don=E2=80=99t want to change anything =E2=80=93 I just wa=
nt to get an SDP with the BUNDLE port assigned to all m- lines. Is there a =
flag in createOffer() for that?<u></u><u></u></span></p>


<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0</span></p></div><=
/div></blockquote><div>The implementation knows that any offers that are cr=
eated after the BUNDLE answer should have the same port numbers. So, the ap=
p could push the createOffer button once it receives the answer if it wante=
d to ensure this happened immediately. Otherwise, there will be an onnegoti=
ationneeded event once ICE completes, and the offer sent at that time will =
have the updated ports.</div>

</div></div></div>

--001a11362cd4b7479204f3a6a48a--


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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>,  Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
References: <7594FB04B1934943A5C02806D1A2204B1D1BC516@ESESSMB209.ericsson.se> <CAJuyhsyk4jO_TS=5khD9ROAx7uYU1YAkCV0Xy4QVUkiqXBJJfg@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17826DFFB545@US70UWXCHMBA02.zam.alcatel-lucent.com> <530E4E34.4010707@alum.mit.edu> <E1FE4C082A89A246A11D7F32A95A17826DFFEDAB@US70UWXCHMBA02.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B1D1C023A@ESESSMB209.ericsson.se> <E1FE4C082A89A246A11D7F32A95A17826DFFF2E4@US70UWXCHMBA02.zam.alcatel-lucent.com> <5310B14E.4000809@alum.mit.edu> <E1FE4C082A89A246A11D7F32A95A17826E0011F6@US70UWXCHMBA02.zam.alcatel-lucent.com>
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Subject: Re: [rtcweb] [MMUSIC] draft-ejzak-mmusic-data-channel-sdpneg: external rejection of channel
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I suspect we are not understanding one another.

On 2/28/14 4:46 PM, Makaraju, Maridi Raju (Raju) wrote:
> Hi Paul,
>
>
>>>> Isn't it enough to, in the data channel protocol spec, say that the
>> odd/even
>>>> rule applies unless the stream id is explicitly negotiated using some
>> other
>>>> mechanism?
>>>>
>>> [Raju] Yes, that is sufficient from DCP draft. draft-ejzak-mmusic-data-
>> channel-sdpneg should still need to say that "when both external and DCP are
>> used, even for DCP created streams the stream ids have to be specified and
>> managed by the application (else DCP stack will default to odd/even rule per
>> DTLS role which may conflict with SDP offer/answer rule)".
>>
>> Can we assume there must be a mechanism for using DCEP and still
>> controlling which channels are used?
>
> [Raju] Yes, we can assume that and it is already supported by :
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-6.5
> and
> http://dev.w3.org/2011/webrtc/editor/webrtc.html#widl-RTCDataChannel-id
> I believe it is safe to assume non-browser implementations of
> data channel stack gives an option for application to select
> stream ids independent of DCEP or external negotiation.
>
>
>>
>> It might help if the O/A negotiation also followed the even/odd rule.
> [Raju] May be I am missing something here. Richard's draft already has
> a specific even/odd rule for O/A. But the issue is TLS based rule may
> conflict with the O/A rule as TLS roles may change depending on a=setup.
>
>> But there are difficulties in knowing who is even and who is odd for the
>> first O/A.
>
> [Raju] Not related to first or subsequent O/A, but rather a conflict
> between TLS roles vs. O/A.

OK. I didn't state what I meant clearly, and I forgot Richard had an 
even/odd rule. For O/A there really isn't a need for one, since the only 
cases this will be a problem result in O/A glare, and that is then 
resolved via backoff.

What I meant was it might help if the O/A proposal and DCEP followed the 
*same* rule. I think that is what Richard wanted. But he showed why O/A 
can't use the existing DCEP rule, so *that* would be required to change.

>> If the offerer always creates the channel before sending the offer, then
>> that will prevent the two mechanisms from stepping on one another's
>> stream ids.
> [Raju]
> In some cases, initial offer may not have a data channel stream created.
> I don't know how this can prevent stepping on each other?!
> May be I am missing something here?! In most cases TLS role is
> determined after O/A is completed, but offerer creates a data
> channel before answer, then how does it know even/odd rule?
> So, trying to use TLS role for external negotiation is problematic.

For channels negotiated at the same time the SCTP association is 
negotiated the offerer can use *any* ids, since there can yet be no 
conflict.

Then, once the association is established, the channels declared in the 
SDP need to be "opened" (activated?), independently by both sides. They 
will then be reserved.

After that, if the application follows the DCEP even/odd rule, and 
creates the channel at its end before sending an offer, then that cannot 
conflict with channels dynamically allocated by DCEP. And both sides 
then understand who is even and who is odd, so there will be no 
conflicts by the two sides choosing the same channel.

	Thanks,
	Paul


From nobody Sun Mar  2 14:14:27 2014
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Subject: Re: [rtcweb] Proxy browsing (Was: Open data channel issues)
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On 02 Mar 2014, at 21:34, Justin Uberti <juberti@google.com> wrote:

> Can you explain more about the specific issue for "U-C 7: Proxy =
browsing"? =20
> =
(http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-3.2)=

Magnus said referring to U-C 7:

>>> Yes, this may be something that is possible, but to express it as a =
use
>>> cases intended to be supported might not be the best. We are after =
all
>>> likely talking about circumventing local security policies.

So keep U-C 7 and describe the consequences in the security section or =
remove it.

I responded yo his e-mail and voted for keeping the U-C.

Best regards
Michael
>=20
> There are existing implementations that do this exact thing, so I =
don't know if anything needs to be done on our side to support this.
>=20
> On Tue, Feb 25, 2014 at 3:44 PM, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> Dear all,
>=20
> Magnus asked me to send a list of open issues regarding data channels
> to the list. Here is my current list:
>=20
>=20
> * Priority
>   The W3C hasn't defined it yet. Neither for the (S)RTP media nor for =
the
>   data channels. We agreed on using a non strict policy for the data =
channels
>   (some sort of wighted fair queueing). That is all.
>=20
> * Protocol
>   It seems not to be clear what needs to be provided when registering =
a
>   (sub)-protocol at IANA. And the name of the registry is unclear...
>=20
> * SCTP parameters.
>   There was discussed the issue how to set SCTP parameters, especially =
path.max.retrans
>   and association.max.retrans. Also HB.Interval might be of interest.
>   RFC 4060 recommends path.max.retrans=3D5, =
association.max.retrans=3D10, but has multihoming
>   in mind. To avoid the dormant state, path.max.retrans =3D =
association.max.retrans should be used.
>   I would suggest 10 for this value. Should HEARTBEATs be disabled?
>=20
> * U-C 7: Proxy browsing
>=20
> * Alternate CC for SCTP
>   Currently there is only the standard CC. However, in some places =
negotiation of CC is
>   mentioned.
>=20
> I'm currently going through the backlog of comments regarding the data =
channels
> ID and I'll try to address the issues. If I find other issues, I send =
an update
> to the above list.
>=20
> Best regards
> Michael
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 3/2/2014 4:27 PM, Justin Uberti wrote:
> I'm not sure I understand the actual problem here. We *are* going to 
> handle arbitrary-sized messages once ndata gets done. The max-msg-size 
> stuff is just a solution until ndata is deployed. (To cite your 
> options, this is A followed by C).

So, here's where I think there may be a disconnect (and if I'm wrong, 
great):

ndata solves the monopolization issue between streams, allowing packets 
for different streams to be interleaved on the wire.  It does not (so 
far as I know) relax the usrsctp library's returning EMSGSIZE if the 
sendto() is larger than available buffer space, and the alternative (EOR 
mode) doesn't allow for interleaving of messages when sending at all 
(and I'm not sure it allows interleaving on reception either, but I 
didn't check right now).

Now, this is an implementation/API issue which could be fixed with some 
work, but so far as I know no work has been done on it.  So ndata will 
allow us to expand the max-sending size to min(SCTP_BUFFER_SIZE - N, 
other-sides-max-receive-size).  It does not allow us to expand the size 
to any useful Blob size.

>
> That said, Send(blob) seems like a bit of a footgun to me anyway; I 
> think apps are going to typically avoid it, a) to avoid memory bloat, 
> and b) to get progress indications. The only way out of that situation 
> is to provide stream semantics, which seems like a lot to take on 
> given that WebSockets hasn't gone that route. I also agree that we 
> shouldn't try to go the (B) route that you mention.

Well, Streams would be a good thing once they're done. Progress 
indication could be added at the W3 level without much problem.  The 
memory issue should be resolvable in a number of ways, per my discussion 
with Sicking.  Note that application chunking has a serious memory issue 
today in that the File Writer API hasn't been agreed to; I think there's 
progress towards eventually adopting a version of the Filesystem API 
(with changes), but that will be a while.  Again, Stream should help - 
and note that the semantics for Blob reception allow it to be written to 
disk as it's received and not held entirely in memory when it's handed 
to the application, which is NOT possible today for application chunking.

>
> So I still think manual chunking seems like a reasonable thing to 
> support. I don't quite get the congestion control concerns; just 
> because there is a max chunk size doesn't mean the impl can't buffer 
> multiple chunks in bufferedAmount; the app could let that fill up to a 
> degree to avoid having to poll constantly to prevent underrun.

On a slow link this will work if the browser isn't janky.  On a fast 
link GC pauses and other things may cause the buffer to drain out and go 
idle for significant periods.

   Randell

>
>
> On Sun, Mar 2, 2014 at 1:37 AM, Randell Jesup <randell-ietf@jesup.org 
> <mailto:randell-ietf@jesup.org>> wrote:
>
>     On 2/26/2014 2:50 AM, Magnus Westerlund wrote:
>
>         Thanks Michael,
>
>         WG, please consider these open issues and try to form your own
>         position
>         on them. They are intended to be discussed at the meeting. If
>         you have
>         proposals on how to resolve them or other input you are very
>         welcome to
>         provide that on the list.
>
>
>     One more big issue.  I realize this is very late for pre-meeting
>     discussion; I'd hoped to hash this out earlier but for various
>     reasons (including power outages and my own workload) this didn't
>     happen.
>
>
>     We discussed a way to deal with the issues surrounding maximum
>     message sizes at the last IETF.  Right now we have a proposal in
>     the MMUSIC draft for limiting the maximum message size via the SDP.
>
>     There is a problem with this: it's at odds with the definition of
>     DataChannels in the W3 and with the "duck-typing" of DataChannels
>     to work largely as a superset of WebSockets (outside of channel
>     creation), and the WebAPI folk at Mozilla I talked to don't like
>     the direction we're taking.
>
>     I've been having talks with the WebAPI people at Mozilla, in
>     particular Jonas Sicking, our WebAPI lead, and they strongly
>     dislike encouraging applications to try to implement their
>     large-data/blob transfer protocols; browsers have considerably
>     more tools available to them to avoid memory hits and to make use
>     of off-main-thread resources than the JS apps do.  "Having
>     Send(blob) fail for any size of blob is crazy and non-sensical"
>     was one comment made when I described the impacts of the current plan.
>
>
>     Manual chunking in the application means poorly-implemented
>     congestion control in the app to keep the channel running
>     efficiently (the only feedback available directly is either having
>     the far-end ack at the user level, or trying to estimate sleep
>     times via setTimeout() and bufferedAmount() (which is simply not a
>     great solution), or simply dumping a large amount of smaller
>     transfers into Send() and causing the browser to have to buffer
>     them in memory).  Also GC or other pauses in JS execution may
>     cause hiccups in the transfer and mis-estimation of available
>     bandwidth.  And of course this is being run over a
>     congestion-controlled channel in the first place.
>
>     Unless and until the W3 side makes DataChannels (and by extension,
>     PeerConnection) APIs available from JS workers (and this is
>     implemented), there will be compromises with packet-level
>     protocols in JS.  One of those will be "it's hard to implement
>     your own congestion control well".  Even with worker support,
>     considerable extension of the APIs would be needed to make it work
>     really well there.  I'll also note that DataChannels-from-worker
>     support is nowhere near implementation in browsers.
>
>
>     Another BIG problem as it's currently defined is that there's no
>     lower bound for this limit, so all DataChannel users will need to
>     implement manual chunking even if they use small fixed messages to
>     guarantee spec compliance.  Of course they won't do so...  and
>     even if they did, they wouldn't test it (another big problem).
>      You might say "ok, fine, lets set some small lower bound on this
>     value, say 2 or 4 or 16K".  That doesn't really help much either.
>      Many will send variable-sized messages (because it's easy), and
>     again won't test what happens when the messages trip over the spec
>     limit (or the actual browser implementation limit!)  Those with
>     fixed-size messages larger than the spec lower-bound won't test
>     the against that; they'll test against what Firefox and Chrome
>     implement at the moment.  So the net result is they'll ship
>     applications that can break randomly in the field for no obvious
>     reason (say if IE implements and uses 16K when Chrome used 32K and
>     Firefox used 100MB).
>
>     Why hand the application a footgun?
>
>
>     Jonas Sicking suggested if the IETF insists on not supporting
>     arbitrary Send(blob), we'll need to push in the W3 for a W3
>     protocol run on top of IETF DataChannels that handles
>     fragmentation and reassembly in order to preserve the JS API for
>     Send().  We can do this, but part of the whole partnership between
>     the IETF and W3 on WebRTC was to try to avoid having the W3 define
>     network protocols and keep them in the IETF where they belong.
>
>     Note: abandoning Send(blob) in W3 doesn't help much, as the
>     comments I made above about arbitrary limits and almost-certain
>     lack of testing of messages violating the negotiated size would
>     still apply.  Send(blob) just makes it easier to trip over the
>     problem (and in fact more likely that the application will test
>     very large sizes).
>
>
>     Our options are:
>
>     A) Just accept this complexity and just hope that people write
>     good code or use good libraries. (See above about testing...)
>     Note: we'd need to set *some* lower bound for the value.
>
>     B) Make the W3 API implementation add a level of protocol on top
>     of the underlying IETF network protocol. This protocol could then
>     deal with fragmenting messages on the sending side and
>     reassembling them on the receiving side.
>
>     C) Convince IETF WG to support arbitrarily sized messages at a
>     protocol level, at least in the WebRTC context, similar to
>     WebSockets.
>
>
>
>     -- 
>     Randell Jesup -- rjesup a t mozilla d o t com
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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    <div class="moz-cite-prefix">On 3/2/2014 4:27 PM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">I'm not sure I understand the actual problem here.
        We *are* going to handle arbitrary-sized messages once ndata
        gets done. The max-msg-size stuff is just a solution until ndata
        is deployed. (To cite your options, this is A followed by C).</div>
    </blockquote>
    <br>
    So, here's where I think there may be a disconnect (and if I'm
    wrong, great):<br>
    <br>
    ndata solves the monopolization issue between streams, allowing
    packets for different streams to be interleaved on the wire.&nbsp; It
    does not (so far as I know) relax the usrsctp library's returning
    EMSGSIZE if the sendto() is larger than available buffer space, and
    the alternative (EOR mode) doesn't allow for interleaving of
    messages when sending at all (and I'm not sure it allows
    interleaving on reception either, but I didn't check right now).<br>
    <br>
    Now, this is an implementation/API issue which could be fixed with
    some work, but so far as I know no work has been done on it.&nbsp; So
    ndata will allow us to expand the max-sending size to
    min(SCTP_BUFFER_SIZE - N, other-sides-max-receive-size).&nbsp; It does
    not allow us to expand the size to any useful Blob size.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>
          <br>
        </div>
        <div>That said, Send(blob) seems like a bit of a footgun to me
          anyway; I think apps are going to typically avoid it, a) to
          avoid memory bloat, and b) to get progress indications. The
          only way out of that situation is to provide stream semantics,
          which seems like a lot to take on given that WebSockets hasn't
          gone that route. I also agree that we shouldn't try to go the
          (B) route that you mention.</div>
      </div>
    </blockquote>
    <br>
    Well, Streams would be a good thing once they're done. Progress
    indication could be added at the W3 level without much problem.&nbsp; The
    memory issue should be resolvable in a number of ways, per my
    discussion with Sicking.&nbsp; Note that application chunking has a
    serious memory issue today in that the File Writer API hasn't been
    agreed to; I think there's progress towards eventually adopting a
    version of the Filesystem API (with changes), but that will be a
    while.&nbsp; Again, Stream should help - and note that the semantics for
    Blob reception allow it to be written to disk as it's received and
    not held entirely in memory when it's handed to the application,
    which is NOT possible today for application chunking.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div><br>
        </div>
        <div>So I still think manual chunking seems like a reasonable
          thing to support. I don't quite get the congestion control
          concerns; just because there is a max chunk size doesn't mean
          the impl can't buffer multiple chunks in bufferedAmount; the
          app could let that fill up to a degree to avoid having to poll
          constantly to prevent underrun.</div>
      </div>
    </blockquote>
    <br>
    On a slow link this will work if the browser isn't janky.&nbsp; On a fast
    link GC pauses and other things may cause the buffer to drain out
    and go idle for significant periods.<br>
    <br>
    &nbsp; Randell<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Sun, Mar 2, 2014 at 1:37 AM, Randell
          Jesup <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div class="">On 2/26/2014 2:50 AM, Magnus Westerlund wrote:<br>
              <blockquote class="gmail_quote" style="margin:0 0 0
                .8ex;border-left:1px #ccc solid;padding-left:1ex">
                Thanks Michael,<br>
                <br>
                WG, please consider these open issues and try to form
                your own position<br>
                on them. They are intended to be discussed at the
                meeting. If you have<br>
                proposals on how to resolve them or other input you are
                very welcome to<br>
                provide that on the list.<br>
              </blockquote>
              <br>
            </div>
            One more big issue. &nbsp;I realize this is very late for
            pre-meeting discussion; I'd hoped to hash this out earlier
            but for various reasons (including power outages and my own
            workload) this didn't happen.<br>
            <br>
            <br>
            We discussed a way to deal with the issues surrounding
            maximum message sizes at the last IETF. &nbsp;Right now we have a
            proposal in the MMUSIC draft for limiting the maximum
            message size via the SDP.<br>
            <br>
            There is a problem with this: it's at odds with the
            definition of DataChannels in the W3 and with the
            "duck-typing" of DataChannels to work largely as a superset
            of WebSockets (outside of channel creation), and the WebAPI
            folk at Mozilla I talked to don't like the direction we're
            taking.<br>
            <br>
            I've been having talks with the WebAPI people at Mozilla, in
            particular Jonas Sicking, our WebAPI lead, and they strongly
            dislike encouraging applications to try to implement their
            large-data/blob transfer protocols; browsers have
            considerably more tools available to them to avoid memory
            hits and to make use of off-main-thread resources than the
            JS apps do. &nbsp;"Having Send(blob) fail for any size of blob is
            crazy and non-sensical" was one comment made when I
            described the impacts of the current plan.<br>
            <br>
            <br>
            Manual chunking in the application means poorly-implemented
            congestion control in the app to keep the channel running
            efficiently (the only feedback available directly is either
            having the far-end ack at the user level, or trying to
            estimate sleep times via setTimeout() and bufferedAmount()
            (which is simply not a great solution), or simply dumping a
            large amount of smaller transfers into Send() and causing
            the browser to have to buffer them in memory). &nbsp;Also GC or
            other pauses in JS execution may cause hiccups in the
            transfer and mis-estimation of available bandwidth. &nbsp;And of
            course this is being run over a congestion-controlled
            channel in the first place.<br>
            <br>
            Unless and until the W3 side makes DataChannels (and by
            extension, PeerConnection) APIs available from JS workers
            (and this is implemented), there will be compromises with
            packet-level protocols in JS. &nbsp;One of those will be "it's
            hard to implement your own congestion control well". &nbsp;Even
            with worker support, considerable extension of the APIs
            would be needed to make it work really well there. &nbsp;I'll
            also note that DataChannels-from-worker support is nowhere
            near implementation in browsers.<br>
            <br>
            <br>
            Another BIG problem as it's currently defined is that
            there's no lower bound for this limit, so all DataChannel
            users will need to implement manual chunking even if they
            use small fixed messages to guarantee spec compliance. &nbsp;Of
            course they won't do so... &nbsp;and even if they did, they
            wouldn't test it (another big problem). &nbsp;You might say "ok,
            fine, lets set some small lower bound on this value, say 2
            or 4 or 16K". &nbsp;That doesn't really help much either. &nbsp;Many
            will send variable-sized messages (because it's easy), and
            again won't test what happens when the messages trip over
            the spec limit (or the actual browser implementation limit!)
            &nbsp;Those with fixed-size messages larger than the spec
            lower-bound won't test the against that; they'll test
            against what Firefox and Chrome implement at the moment. &nbsp;So
            the net result is they'll ship applications that can break
            randomly in the field for no obvious reason (say if IE
            implements and uses 16K when Chrome used 32K and Firefox
            used 100MB).<br>
            <br>
            Why hand the application a footgun?<br>
            <br>
            <br>
            Jonas Sicking suggested if the IETF insists on not
            supporting arbitrary Send(blob), we'll need to push in the
            W3 for a W3 protocol run on top of IETF DataChannels that
            handles fragmentation and reassembly in order to preserve
            the JS API for Send(). &nbsp;We can do this, but part of the
            whole partnership between the IETF and W3 on WebRTC was to
            try to avoid having the W3 define network protocols and keep
            them in the IETF where they belong.<br>
            <br>
            Note: abandoning Send(blob) in W3 doesn't help much, as the
            comments I made above about arbitrary limits and
            almost-certain lack of testing of messages violating the
            negotiated size would still apply. &nbsp;Send(blob) just makes it
            easier to trip over the problem (and in fact more likely
            that the application will test very large sizes).<br>
            <br>
            <br>
            Our options are:<br>
            <br>
            A) Just accept this complexity and just hope that people
            write good code or use good libraries. (See above about
            testing...)<br>
            Note: we'd need to set *some* lower bound for the value.<br>
            <br>
            B) Make the W3 API implementation add a level of protocol on
            top of the underlying IETF network protocol. This protocol
            could then deal with fragmenting messages on the sending
            side and reassembling them on the receiving side.<br>
            <br>
            C) Convince IETF WG to support arbitrarily sized messages at
            a protocol level, at least in the WebRTC context, similar to
            WebSockets.
            <div class="im HOEnZb"><br>
              <br>
              <br>
              -- <br>
              Randell Jesup -- rjesup a t mozilla d o t com<br>
              <br>
            </div>
            <div class="HOEnZb">
              <div class="h5">
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org"
                  target="_blank">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
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On 2 March 2014 13:45, Justin Uberti <juberti@google.com> wrote:
> We had a discussion in Seattle (IIRC) that the "first" track could be the
> first one provided to AddTrack, and I think that is simplest. (This of
> course requires the AddTrack proposal to go through.)

That's true.  But then you have to decide which is first :)  We could
assume that the order follows what is in every example ever, but
that's still an assumption.


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On 2 March 2014 18:42, Randell Jesup <randell-ietf@jesup.org> wrote:
> ndata solves the monopolization issue between streams, allowing packets for
> different streams to be interleaved on the wire.  It does not (so far as I
> know) relax the usrsctp library's returning EMSGSIZE if the sendto() is
> larger than available buffer space, and the alternative (EOR mode) doesn't
> allow for interleaving of messages when sending at all (and I'm not sure it
> allows interleaving on reception either, but I didn't check right now).
>
> Now, this is an implementation/API issue which could be fixed with some
> work, but so far as I know no work has been done on it.  So ndata will allow
> us to expand the max-sending size to min(SCTP_BUFFER_SIZE - N,
> other-sides-max-receive-size).  It does not allow us to expand the size to
> any useful Blob size.

Sounds like the complaint would be best addressed by fixing the SCTP
implementations.  I would hope that ndata would allow for transmission
of messages far in excess of what the implementation is willing to
buffer, that seems arbitrarily restrictive.

(A streaming API wounds like a good idea too.)


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Cc: rtcweb@ietf.org, Randall Stewart <rrs@lakerest.net>
Subject: Re: [rtcweb] Open data channel issues
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On 03 Mar 2014, at 02:42, Randell Jesup <randell-ietf@jesup.org> wrote:

> On 3/2/2014 4:27 PM, Justin Uberti wrote:
>> I'm not sure I understand the actual problem here. We *are* going to =
handle arbitrary-sized messages once ndata gets done. The max-msg-size =
stuff is just a solution until ndata is deployed. (To cite your options, =
this is A followed by C).
>=20
> So, here's where I think there may be a disconnect (and if I'm wrong, =
great):
>=20
> ndata solves the monopolization issue between streams, allowing =
packets for different streams to be interleaved on the wire.  It does =
not (so far as I know) relax the usrsctp library's returning=20
Correct.
> EMSGSIZE if the sendto() is larger than available buffer space, and =
the alternative (EOR mode) doesn't allow for interleaving of messages =
when sending at all (and I'm not sure it allows interleaving on =
reception either, but I didn't check right now).
If you want to send messages larger than the socket send buffer, you =
must use the explicit
EOR more. This is independent from using NDATA or not.
If you use NDATA, you can have on each outgoing stream a single message =
(order or unordered)
being sent using explicit EOR mode and the calls can be interleaved. I =
don't think I have
tested this up to know, but this is how it should work.

rrs@: Just to double check: Am I right? Please note that RTCWeb doesn't =
use ordered and
      unordered on the same stream, and you don't need to send more than =
one message with
      explicit EOR on the same stream at the same time. Therefore the =
above simplification...
>=20
> Now, this is an implementation/API issue which could be fixed with =
some work, but so far as I know no work has been done on it.  So ndata =
will allow us to expand the max-sending size to=20
This API work is part of getting NDATA implemented. Randall has =
implemented it, we do have
code, but it needs some testing and cleaning up in the API which =
supports the scheduling...
> min(SCTP_BUFFER_SIZE - N, other-sides-max-receive-size).  It does not =
allow us to expand the size to any useful Blob size.
Please note that you never have a limitation of the receive buffer send =
size, since SCTP
would use partial delivery. So other-sides-max-receive-size is only =
relevant if there are
buffering limits on top of SCTP.

Best regards
Michael
>=20
>>=20
>> That said, Send(blob) seems like a bit of a footgun to me anyway; I =
think apps are going to typically avoid it, a) to avoid memory bloat, =
and b) to get progress indications. The only way out of that situation =
is to provide stream semantics, which seems like a lot to take on given =
that WebSockets hasn't gone that route. I also agree that we shouldn't =
try to go the (B) route that you mention.
>=20
> Well, Streams would be a good thing once they're done. Progress =
indication could be added at the W3 level without much problem.  The =
memory issue should be resolvable in a number of ways, per my discussion =
with Sicking.  Note that application chunking has a serious memory issue =
today in that the File Writer API hasn't been agreed to; I think there's =
progress towards eventually adopting a version of the Filesystem API =
(with changes), but that will be a while.  Again, Stream should help - =
and note that the semantics for Blob reception allow it to be written to =
disk as it's received and not held entirely in memory when it's handed =
to the application, which is NOT possible today for application =
chunking.
>=20
>>=20
>> So I still think manual chunking seems like a reasonable thing to =
support. I don't quite get the congestion control concerns; just because =
there is a max chunk size doesn't mean the impl can't buffer multiple =
chunks in bufferedAmount; the app could let that fill up to a degree to =
avoid having to poll constantly to prevent underrun.
>=20
> On a slow link this will work if the browser isn't janky.  On a fast =
link GC pauses and other things may cause the buffer to drain out and go =
idle for significant periods.
>=20
>   Randell
>=20
>>=20
>>=20
>> On Sun, Mar 2, 2014 at 1:37 AM, Randell Jesup =
<randell-ietf@jesup.org> wrote:
>> On 2/26/2014 2:50 AM, Magnus Westerlund wrote:
>> Thanks Michael,
>>=20
>> WG, please consider these open issues and try to form your own =
position
>> on them. They are intended to be discussed at the meeting. If you =
have
>> proposals on how to resolve them or other input you are very welcome =
to
>> provide that on the list.
>>=20
>> One more big issue.  I realize this is very late for pre-meeting =
discussion; I'd hoped to hash this out earlier but for various reasons =
(including power outages and my own workload) this didn't happen.
>>=20
>>=20
>> We discussed a way to deal with the issues surrounding maximum =
message sizes at the last IETF.  Right now we have a proposal in the =
MMUSIC draft for limiting the maximum message size via the SDP.
>>=20
>> There is a problem with this: it's at odds with the definition of =
DataChannels in the W3 and with the "duck-typing" of DataChannels to =
work largely as a superset of WebSockets (outside of channel creation), =
and the WebAPI folk at Mozilla I talked to don't like the direction =
we're taking.
>>=20
>> I've been having talks with the WebAPI people at Mozilla, in =
particular Jonas Sicking, our WebAPI lead, and they strongly dislike =
encouraging applications to try to implement their large-data/blob =
transfer protocols; browsers have considerably more tools available to =
them to avoid memory hits and to make use of off-main-thread resources =
than the JS apps do.  "Having Send(blob) fail for any size of blob is =
crazy and non-sensical" was one comment made when I described the =
impacts of the current plan.
>>=20
>>=20
>> Manual chunking in the application means poorly-implemented =
congestion control in the app to keep the channel running efficiently =
(the only feedback available directly is either having the far-end ack =
at the user level, or trying to estimate sleep times via setTimeout() =
and bufferedAmount() (which is simply not a great solution), or simply =
dumping a large amount of smaller transfers into Send() and causing the =
browser to have to buffer them in memory).  Also GC or other pauses in =
JS execution may cause hiccups in the transfer and mis-estimation of =
available bandwidth.  And of course this is being run over a =
congestion-controlled channel in the first place.
>>=20
>> Unless and until the W3 side makes DataChannels (and by extension, =
PeerConnection) APIs available from JS workers (and this is =
implemented), there will be compromises with packet-level protocols in =
JS.  One of those will be "it's hard to implement your own congestion =
control well".  Even with worker support, considerable extension of the =
APIs would be needed to make it work really well there.  I'll also note =
that DataChannels-from-worker support is nowhere near implementation in =
browsers.
>>=20
>>=20
>> Another BIG problem as it's currently defined is that there's no =
lower bound for this limit, so all DataChannel users will need to =
implement manual chunking even if they use small fixed messages to =
guarantee spec compliance.  Of course they won't do so...  and even if =
they did, they wouldn't test it (another big problem).  You might say =
"ok, fine, lets set some small lower bound on this value, say 2 or 4 or =
16K".  That doesn't really help much either.  Many will send =
variable-sized messages (because it's easy), and again won't test what =
happens when the messages trip over the spec limit (or the actual =
browser implementation limit!)  Those with fixed-size messages larger =
than the spec lower-bound won't test the against that; they'll test =
against what Firefox and Chrome implement at the moment.  So the net =
result is they'll ship applications that can break randomly in the field =
for no obvious reason (say if IE implements and uses 16K when Chrome =
used 32K and Firefox used 100MB).
>>=20
>> Why hand the application a footgun?
>>=20
>>=20
>> Jonas Sicking suggested if the IETF insists on not supporting =
arbitrary Send(blob), we'll need to push in the W3 for a W3 protocol run =
on top of IETF DataChannels that handles fragmentation and reassembly in =
order to preserve the JS API for Send().  We can do this, but part of =
the whole partnership between the IETF and W3 on WebRTC was to try to =
avoid having the W3 define network protocols and keep them in the IETF =
where they belong.
>>=20
>> Note: abandoning Send(blob) in W3 doesn't help much, as the comments =
I made above about arbitrary limits and almost-certain lack of testing =
of messages violating the negotiated size would still apply.  Send(blob) =
just makes it easier to trip over the problem (and in fact more likely =
that the application will test very large sizes).
>>=20
>>=20
>> Our options are:
>>=20
>> A) Just accept this complexity and just hope that people write good =
code or use good libraries. (See above about testing...)
>> Note: we'd need to set *some* lower bound for the value.
>>=20
>> B) Make the W3 API implementation add a level of protocol on top of =
the underlying IETF network protocol. This protocol could then deal with =
fragmenting messages on the sending side and reassembling them on the =
receiving side.
>>=20
>> C) Convince IETF WG to support arbitrarily sized messages at a =
protocol level, at least in the WebRTC context, similar to WebSockets.
>>=20
>>=20
>>=20
>> --=20
>> Randell Jesup -- rjesup a t mozilla d o t com
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>>=20
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Open data channel issues
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On 03 Mar 2014, at 06:56, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 2 March 2014 18:42, Randell Jesup <randell-ietf@jesup.org> wrote:
>> ndata solves the monopolization issue between streams, allowing =
packets for
>> different streams to be interleaved on the wire.  It does not (so far =
as I
>> know) relax the usrsctp library's returning EMSGSIZE if the sendto() =
is
>> larger than available buffer space, and the alternative (EOR mode) =
doesn't
>> allow for interleaving of messages when sending at all (and I'm not =
sure it
>> allows interleaving on reception either, but I didn't check right =
now).
>>=20
>> Now, this is an implementation/API issue which could be fixed with =
some
>> work, but so far as I know no work has been done on it.  So ndata =
will allow
>> us to expand the max-sending size to min(SCTP_BUFFER_SIZE - N,
>> other-sides-max-receive-size).  It does not allow us to expand the =
size to
>> any useful Blob size.
>=20
> Sounds like the complaint would be best addressed by fixing the SCTP
> implementations.  I would hope that ndata would allow for transmission
> of messages far in excess of what the implementation is willing to
> buffer, that seems arbitrarily restrictive.
As said in my previous mail: Using NDATA will allow to have on each
stream a single ordered or unordered message being sent in explicit
EOR mode. The send calls can be interleaved.

Best regards
Michael
>=20
> (A streaming API wounds like a good idea too.)
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'IESG IESG'" <iesg@ietf.org>
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Subject: Re: [rtcweb] [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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Hi Cullen,

Memory may be an illusion, still I cannot understand how there can be an
ever growing archive in the IETF

/Karl

Allt v=E4l
Hi Cullen,

Not to cause more lost in the threads (I am also) I am addressing both =
your
emails in one, and copy separately to the other guys in the first email. =


Further Inline -->

>From http://www.ietf.org/mail-archive/web/tram/current/msg00275.html=20
Let me also point out draft-deng-tram-isp-turn-00=20
http://www.ietf.org/mail-archive/web/tram/current/msg00214.html from =
China
Mobile that appeared on this mailing list yesterday. It points out the =
need
and willingness from the ISP/NSP side to do something about QoS for =
WebRTC
traffic, that they expect to be large and have to bring to their =
customers
with best QoE. In fact they and some more (a huge European carrier and =
the
cable operators in general - CableLabs) have expressed similar concerns =
to
me *=93This is exactly something we want to work on as the current way =
will
cause severe problem on traffic when rtcweb applications get popular=94* =
is a
direct quote from one of those.

So, in answer to Simon Perreault [simon.perreault at viagenie.ca]=92s =
question
in another thread the 13th:
a) Is this a real problem that is worth fixing?
I think we can be confident that the answer is *YES*. Only these three
ISP/NSP referred to, may represent 50%(?) of the universe=92s IP =
accesses! And
the other will follow these most forward looking carriers.=20


-----Ursprungligt meddelande-----
Fr=E5n: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]=20
Skickat: den 26 februari 2014 01:49
Till: Karl Stahl
Kopia: Alan Johnston; rtcweb@ietf.org; tram@ietf.org; Bernard Aboba; =
David
Singer; Harald Tveit Alvestrand; Marc Robins; Eric Burger; Mary Barnes;
Henning Schulzrinne
=C4mne: Re: [rtcweb] [tram] I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt


> Karl,=20

> I am totally lost on this thread. Could you start a new tmead that
summarize what the issues is, what seems to > be the point of debate, =
and
what your view is on what we should do. I think that would help make
progress.=20

> Thank you,=20

> Cullen

I just did something like that in:
[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff=20
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html=20
Hoping this thread subject amplifies what this SHOULD be about. (An how =
it
will lead to the summary, after all diversion.)

Actually, the outcome of "Interfacing to QoS" will be=20


I challenge anyone that this will be outcome, and will start =
"Interfacing to
QoS" with what IP "Interfacing to QoS" that we have the tiny ADSL modem =
IX78
that implements such CLEAR INTERFACE and the actual QoS methods, for =
both
the Congestion, Default gateway, Surf Pipe=20
Manipulating the RTP extension header etc etc on 6 USD CPU and also =
include
the ADSL part where that quality handling is based on heavy 57 bytes
chopping up
   at this exact=20




-----Ursprungligt meddelande-----
Fr=E5n: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]=20
Skickat: den 26 februari 2014 02:06
Till: Karl Stahl; IESG IESG
Kopia: Magnus Westerlund; Simon Perreault; Ted Hardie; Gonzalo =
Camarillo;
rtcweb@ietf.org; tram@ietf.org; Spencer Dawkins
=C4mne: Re: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter
Clarification regardig Milestone 3: TURN server auto-discovery mechanism =
for
enterprise and ISPs



On Feb 25, 2014, at 7:02 AM, Karl Stahl <karl.stahl@intertex.se> wrote:

> To the TRAM WG and RTCWEB WG and ADs:
> =20
> It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed =
and
encouraged to route quality demanding WebRTC media into their IP pipes =
that
are capable of transporting real-time traffic without quality issues, =
using
TURN servers.
> =20

Reading the charter, the above is *not* at all a clear objective of the =
WG
(note I am not the chair of this WG or the responsible AD).=20

That said, I think you have pointed out this charter is abysmally vague =
- it
does not say what the WG is not going to do. If I decided to do BGP for
routing updates over TURN it would be within the scope of this charter.=20

My advice to the responsible AD is recharter this WG before IETF 90 or =
close
it. I would be glad to help write a charter that is not an infinite =
blank
cheque.=20







From nobody Mon Mar  3 02:31:09 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: JSEP-06: How to create the second offer where only the port numbers are changed?
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: JSEP-06: max-bundle policy questions
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Subject: Re: [rtcweb] JSEP-06: max-bundle policy questions
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From nobody Mon Mar  3 02:39:16 2014
Return-Path: <martin.thomson@gmail.com>
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Date: Mon, 3 Mar 2014 02:39:01 -0800
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From: Martin Thomson <martin.thomson@gmail.com>
To: Michael Tuexen <tuexen@fh-muenster.de>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Open data channel issues
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On 2 March 2014 23:01, Michael Tuexen <tuexen@fh-muenster.de> wrote:
> As said in my previous mail: Using NDATA will allow to have on each
> stream a single ordered or unordered message being sent in explicit
> EOR mode. The send calls can be interleaved.

That's great.  So the protocol can and is being fixed.  I think that
part of Randell's concern comes from the implementations thereof.
Those would seem to be even easier to fix.


From nobody Mon Mar  3 02:44:44 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-06.txt - PeerConnection BUNDLE policy
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On 2014-03-03 07:57, Martin Thomson wrote:=0A=
=0A=
> (A streaming API wounds like a good idea too.)=0A=
=0A=
There is some ongoing work: =0A=
http://lists.w3.org/Archives/Public/public-webapps/2014JanMar/0218.html=0A=
=0A=
>=0A=
> _______________________________________________=0A=
> rtcweb mailing list=0A=
> rtcweb@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=


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On 3 March 2014 02:45, Stefan H=C3=A5kansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
> There is some ongoing work:
> http://lists.w3.org/Archives/Public/public-webapps/2014JanMar/0218.html

I was thinking more specific work applying that to thewebsocketapi.


From nobody Mon Mar  3 03:19:10 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Mar 2014 03:18:42 -0800
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On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup <randell-ietf@jesup.org>wrote:

>  On 3/2/2014 4:27 PM, Justin Uberti wrote:
>
> I'm not sure I understand the actual problem here. We *are* going to
> handle arbitrary-sized messages once ndata gets done. The max-msg-size
> stuff is just a solution until ndata is deployed. (To cite your options,
> this is A followed by C).
>
>
> So, here's where I think there may be a disconnect (and if I'm wrong,
> great):
>
> ndata solves the monopolization issue between streams, allowing packets
> for different streams to be interleaved on the wire.  It does not (so far
> as I know) relax the usrsctp library's returning EMSGSIZE if the sendto()
> is larger than available buffer space, and the alternative (EOR mode)
> doesn't allow for interleaving of messages when sending at all (and I'm not
> sure it allows interleaving on reception either, but I didn't check right
> now).
>
> Now, this is an implementation/API issue which could be fixed with some
> work, but so far as I know no work has been done on it.  So ndata will
> allow us to expand the max-sending size to min(SCTP_BUFFER_SIZE - N,
> other-sides-max-receive-size).  It does not allow us to expand the size to
> any useful Blob size.
>

Based on the discussion of ndata/eor in followup messages, I think this
problem will be addressed in the near future.


>
>
>
>  That said, Send(blob) seems like a bit of a footgun to me anyway; I
> think apps are going to typically avoid it, a) to avoid memory bloat, and
> b) to get progress indications. The only way out of that situation is to
> provide stream semantics, which seems like a lot to take on given that
> WebSockets hasn't gone that route. I also agree that we shouldn't try to go
> the (B) route that you mention.
>
>
> Well, Streams would be a good thing once they're done. Progress indication
> could be added at the W3 level without much problem.  The memory issue
> should be resolvable in a number of ways, per my discussion with Sicking.
> Note that application chunking has a serious memory issue today in that the
> File Writer API hasn't been agreed to; I think there's progress towards
> eventually adopting a version of the Filesystem API (with changes), but
> that will be a while.  Again, Stream should help - and note that the
> semantics for Blob reception allow it to be written to disk as it's
> received and not held entirely in memory when it's handed to the
> application, which is NOT possible today for application chunking.
>

Agree Streams will be useful when done, but that's probably post-1.0 (as
Martin says, we should first let it be applied to WebSockets). However I
don't think app chunking will have any memory problems - the chunks will be
small enough that they can be easily spooled to disk as they come in.

>
>
>  So I still think manual chunking seems like a reasonable thing to
> support. I don't quite get the congestion control concerns; just because
> there is a max chunk size doesn't mean the impl can't buffer multiple
> chunks in bufferedAmount; the app could let that fill up to a degree to
> avoid having to poll constantly to prevent underrun.
>
>
> On a slow link this will work if the browser isn't janky.  On a fast link
> GC pauses and other things may cause the buffer to drain out and go idle
> for significant periods.
>

Since the amount buffered is up to the app, the app can just increase the
buffer size to provide enough data to cover the next second or so of
transfer.

Note that app chunking will clearly be needed in some of the more
interesting cases we want to support, such as where the transfer is not
1:1. For swarm-style downloading, we need to make sure app chunking works
well (and I think it does).

>
>
>   Randell
>
>
>
>
> On Sun, Mar 2, 2014 at 1:37 AM, Randell Jesup <randell-ietf@jesup.org>wrote:
>
>> On 2/26/2014 2:50 AM, Magnus Westerlund wrote:
>>
>>> Thanks Michael,
>>>
>>> WG, please consider these open issues and try to form your own position
>>> on them. They are intended to be discussed at the meeting. If you have
>>> proposals on how to resolve them or other input you are very welcome to
>>> provide that on the list.
>>>
>>
>>  One more big issue.  I realize this is very late for pre-meeting
>> discussion; I'd hoped to hash this out earlier but for various reasons
>> (including power outages and my own workload) this didn't happen.
>>
>>
>> We discussed a way to deal with the issues surrounding maximum message
>> sizes at the last IETF.  Right now we have a proposal in the MMUSIC draft
>> for limiting the maximum message size via the SDP.
>>
>> There is a problem with this: it's at odds with the definition of
>> DataChannels in the W3 and with the "duck-typing" of DataChannels to work
>> largely as a superset of WebSockets (outside of channel creation), and the
>> WebAPI folk at Mozilla I talked to don't like the direction we're taking.
>>
>> I've been having talks with the WebAPI people at Mozilla, in particular
>> Jonas Sicking, our WebAPI lead, and they strongly dislike encouraging
>> applications to try to implement their large-data/blob transfer protocols;
>> browsers have considerably more tools available to them to avoid memory
>> hits and to make use of off-main-thread resources than the JS apps do.
>>  "Having Send(blob) fail for any size of blob is crazy and non-sensical"
>> was one comment made when I described the impacts of the current plan.
>>
>>
>> Manual chunking in the application means poorly-implemented congestion
>> control in the app to keep the channel running efficiently (the only
>> feedback available directly is either having the far-end ack at the user
>> level, or trying to estimate sleep times via setTimeout() and
>> bufferedAmount() (which is simply not a great solution), or simply dumping
>> a large amount of smaller transfers into Send() and causing the browser to
>> have to buffer them in memory).  Also GC or other pauses in JS execution
>> may cause hiccups in the transfer and mis-estimation of available
>> bandwidth.  And of course this is being run over a congestion-controlled
>> channel in the first place.
>>
>> Unless and until the W3 side makes DataChannels (and by extension,
>> PeerConnection) APIs available from JS workers (and this is implemented),
>> there will be compromises with packet-level protocols in JS.  One of those
>> will be "it's hard to implement your own congestion control well".  Even
>> with worker support, considerable extension of the APIs would be needed to
>> make it work really well there.  I'll also note that
>> DataChannels-from-worker support is nowhere near implementation in browsers.
>>
>>
>> Another BIG problem as it's currently defined is that there's no lower
>> bound for this limit, so all DataChannel users will need to implement
>> manual chunking even if they use small fixed messages to guarantee spec
>> compliance.  Of course they won't do so...  and even if they did, they
>> wouldn't test it (another big problem).  You might say "ok, fine, lets set
>> some small lower bound on this value, say 2 or 4 or 16K".  That doesn't
>> really help much either.  Many will send variable-sized messages (because
>> it's easy), and again won't test what happens when the messages trip over
>> the spec limit (or the actual browser implementation limit!)  Those with
>> fixed-size messages larger than the spec lower-bound won't test the against
>> that; they'll test against what Firefox and Chrome implement at the moment.
>>  So the net result is they'll ship applications that can break randomly in
>> the field for no obvious reason (say if IE implements and uses 16K when
>> Chrome used 32K and Firefox used 100MB).
>>
>> Why hand the application a footgun?
>>
>>
>> Jonas Sicking suggested if the IETF insists on not supporting arbitrary
>> Send(blob), we'll need to push in the W3 for a W3 protocol run on top of
>> IETF DataChannels that handles fragmentation and reassembly in order to
>> preserve the JS API for Send().  We can do this, but part of the whole
>> partnership between the IETF and W3 on WebRTC was to try to avoid having
>> the W3 define network protocols and keep them in the IETF where they belong.
>>
>> Note: abandoning Send(blob) in W3 doesn't help much, as the comments I
>> made above about arbitrary limits and almost-certain lack of testing of
>> messages violating the negotiated size would still apply.  Send(blob) just
>> makes it easier to trip over the problem (and in fact more likely that the
>> application will test very large sizes).
>>
>>
>> Our options are:
>>
>> A) Just accept this complexity and just hope that people write good code
>> or use good libraries. (See above about testing...)
>> Note: we'd need to set *some* lower bound for the value.
>>
>> B) Make the W3 API implementation add a level of protocol on top of the
>> underlying IETF network protocol. This protocol could then deal with
>> fragmenting messages on the sending side and reassembling them on the
>> receiving side.
>>
>> C) Convince IETF WG to support arbitrarily sized messages at a protocol
>> level, at least in the WebRTC context, similar to WebSockets.
>>
>>
>>
>> --
>> Randell Jesup -- rjesup a t mozilla d o t com
>>
>>   _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7bacb7386f6ee804f3b1f407
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quo=
te">On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup <span dir=3D"ltr">&lt;<a =
href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@jesup=
.org</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D"">
    <div>On 3/2/2014 4:27 PM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">I&#39;m not sure I understand the actual problem her=
e.
        We *are* going to handle arbitrary-sized messages once ndata
        gets done. The max-msg-size stuff is just a solution until ndata
        is deployed. (To cite your options, this is A followed by C).</div>
    </blockquote>
    <br></div>
    So, here&#39;s where I think there may be a disconnect (and if I&#39;m
    wrong, great):<br>
    <br>
    ndata solves the monopolization issue between streams, allowing
    packets for different streams to be interleaved on the wire.=C2=A0 It
    does not (so far as I know) relax the usrsctp library&#39;s returning
    EMSGSIZE if the sendto() is larger than available buffer space, and
    the alternative (EOR mode) doesn&#39;t allow for interleaving of
    messages when sending at all (and I&#39;m not sure it allows
    interleaving on reception either, but I didn&#39;t check right now).<br=
>
    <br>
    Now, this is an implementation/API issue which could be fixed with
    some work, but so far as I know no work has been done on it.=C2=A0 So
    ndata will allow us to expand the max-sending size to
    min(SCTP_BUFFER_SIZE - N, other-sides-max-receive-size).=C2=A0 It does
    not allow us to expand the size to any useful Blob size.</div></blockqu=
ote><div><br></div><div>Based on the discussion of ndata/eor in followup me=
ssages, I think this problem will be addressed in the near future.</div>

<div>=C2=A0=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0p=
x 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bo=
rder-left-style:solid;padding-left:1ex"><div text=3D"#000000" bgcolor=3D"#F=
FFFFF"><div class=3D"">

<br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>
          <br>
        </div>
        <div>That said, Send(blob) seems like a bit of a footgun to me
          anyway; I think apps are going to typically avoid it, a) to
          avoid memory bloat, and b) to get progress indications. The
          only way out of that situation is to provide stream semantics,
          which seems like a lot to take on given that WebSockets hasn&#39;=
t
          gone that route. I also agree that we shouldn&#39;t try to go the
          (B) route that you mention.</div>
      </div>
    </blockquote>
    <br></div>
    Well, Streams would be a good thing once they&#39;re done. Progress
    indication could be added at the W3 level without much problem.=C2=A0 T=
he
    memory issue should be resolvable in a number of ways, per my
    discussion with Sicking.=C2=A0 Note that application chunking has a
    serious memory issue today in that the File Writer API hasn&#39;t been
    agreed to; I think there&#39;s progress towards eventually adopting a
    version of the Filesystem API (with changes), but that will be a
    while.=C2=A0 Again, Stream should help - and note that the semantics fo=
r
    Blob reception allow it to be written to disk as it&#39;s received and
    not held entirely in memory when it&#39;s handed to the application,
    which is NOT possible today for application chunking.</div></blockquote=
><div><br></div><div>Agree Streams will be useful when done, but that&#39;s=
 probably post-1.0 (as Martin says, we should first let it be applied to We=
bSockets). However I don&#39;t think app chunking will have any memory prob=
lems - the chunks will be small enough that they can be easily spooled to d=
isk as they come in.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D"">
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div><br>
        </div>
        <div>So I still think manual chunking seems like a reasonable
          thing to support. I don&#39;t quite get the congestion control
          concerns; just because there is a max chunk size doesn&#39;t mean
          the impl can&#39;t buffer multiple chunks in bufferedAmount; the
          app could let that fill up to a degree to avoid having to poll
          constantly to prevent underrun.</div>
      </div>
    </blockquote>
    <br></div>
    On a slow link this will work if the browser isn&#39;t janky.=C2=A0 On =
a fast
    link GC pauses and other things may cause the buffer to drain out
    and go idle for significant periods.</div></blockquote><div><br></div><=
div>Since the amount buffered is up to the app, the app can just increase t=
he buffer size to provide enough data to cover the next second or so of tra=
nsfer.=C2=A0</div>

<div><br></div><div><div>Note that app chunking will clearly be needed in s=
ome of the more interesting cases we want to support, such as where the tra=
nsfer is not 1:1. For swarm-style downloading, we need to make sure app chu=
nking works well (and I think it does).</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF"></div></blockquo=
te>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF"><span class=3D""=
><font color=3D"#888888"><br>


    <br>
    =C2=A0 Randell</font></span><div><div class=3D"h5"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
      </div>
      <div class=3D"gmail_extra"><br>
        <br>
        <div class=3D"gmail_quote">On Sun, Mar 2, 2014 at 1:37 AM, Randell
          Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf@jesup.=
org" target=3D"_blank">randell-ietf@jesup.org</a>&gt;</span>
          wrote:<br>
          <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-sty=
le:solid;padding-left:1ex">
            <div>On 2/26/2014 2:50 AM, Magnus Westerlund wrote:<br>
              <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
                Thanks Michael,<br>
                <br>
                WG, please consider these open issues and try to form
                your own position<br>
                on them. They are intended to be discussed at the
                meeting. If you have<br>
                proposals on how to resolve them or other input you are
                very welcome to<br>
                provide that on the list.<br>
              </blockquote>
              <br>
            </div>
            One more big issue. =C2=A0I realize this is very late for
            pre-meeting discussion; I&#39;d hoped to hash this out earlier
            but for various reasons (including power outages and my own
            workload) this didn&#39;t happen.<br>
            <br>
            <br>
            We discussed a way to deal with the issues surrounding
            maximum message sizes at the last IETF. =C2=A0Right now we have=
 a
            proposal in the MMUSIC draft for limiting the maximum
            message size via the SDP.<br>
            <br>
            There is a problem with this: it&#39;s at odds with the
            definition of DataChannels in the W3 and with the
            &quot;duck-typing&quot; of DataChannels to work largely as a su=
perset
            of WebSockets (outside of channel creation), and the WebAPI
            folk at Mozilla I talked to don&#39;t like the direction we&#39=
;re
            taking.<br>
            <br>
            I&#39;ve been having talks with the WebAPI people at Mozilla, i=
n
            particular Jonas Sicking, our WebAPI lead, and they strongly
            dislike encouraging applications to try to implement their
            large-data/blob transfer protocols; browsers have
            considerably more tools available to them to avoid memory
            hits and to make use of off-main-thread resources than the
            JS apps do. =C2=A0&quot;Having Send(blob) fail for any size of =
blob is
            crazy and non-sensical&quot; was one comment made when I
            described the impacts of the current plan.<br>
            <br>
            <br>
            Manual chunking in the application means poorly-implemented
            congestion control in the app to keep the channel running
            efficiently (the only feedback available directly is either
            having the far-end ack at the user level, or trying to
            estimate sleep times via setTimeout() and bufferedAmount()
            (which is simply not a great solution), or simply dumping a
            large amount of smaller transfers into Send() and causing
            the browser to have to buffer them in memory). =C2=A0Also GC or
            other pauses in JS execution may cause hiccups in the
            transfer and mis-estimation of available bandwidth. =C2=A0And o=
f
            course this is being run over a congestion-controlled
            channel in the first place.<br>
            <br>
            Unless and until the W3 side makes DataChannels (and by
            extension, PeerConnection) APIs available from JS workers
            (and this is implemented), there will be compromises with
            packet-level protocols in JS. =C2=A0One of those will be &quot;=
it&#39;s
            hard to implement your own congestion control well&quot;. =C2=
=A0Even
            with worker support, considerable extension of the APIs
            would be needed to make it work really well there. =C2=A0I&#39;=
ll
            also note that DataChannels-from-worker support is nowhere
            near implementation in browsers.<br>
            <br>
            <br>
            Another BIG problem as it&#39;s currently defined is that
            there&#39;s no lower bound for this limit, so all DataChannel
            users will need to implement manual chunking even if they
            use small fixed messages to guarantee spec compliance. =C2=A0Of
            course they won&#39;t do so... =C2=A0and even if they did, they
            wouldn&#39;t test it (another big problem). =C2=A0You might say=
 &quot;ok,
            fine, lets set some small lower bound on this value, say 2
            or 4 or 16K&quot;. =C2=A0That doesn&#39;t really help much eith=
er. =C2=A0Many
            will send variable-sized messages (because it&#39;s easy), and
            again won&#39;t test what happens when the messages trip over
            the spec limit (or the actual browser implementation limit!)
            =C2=A0Those with fixed-size messages larger than the spec
            lower-bound won&#39;t test the against that; they&#39;ll test
            against what Firefox and Chrome implement at the moment. =C2=A0=
So
            the net result is they&#39;ll ship applications that can break
            randomly in the field for no obvious reason (say if IE
            implements and uses 16K when Chrome used 32K and Firefox
            used 100MB).<br>
            <br>
            Why hand the application a footgun?<br>
            <br>
            <br>
            Jonas Sicking suggested if the IETF insists on not
            supporting arbitrary Send(blob), we&#39;ll need to push in the
            W3 for a W3 protocol run on top of IETF DataChannels that
            handles fragmentation and reassembly in order to preserve
            the JS API for Send(). =C2=A0We can do this, but part of the
            whole partnership between the IETF and W3 on WebRTC was to
            try to avoid having the W3 define network protocols and keep
            them in the IETF where they belong.<br>
            <br>
            Note: abandoning Send(blob) in W3 doesn&#39;t help much, as the
            comments I made above about arbitrary limits and
            almost-certain lack of testing of messages violating the
            negotiated size would still apply. =C2=A0Send(blob) just makes =
it
            easier to trip over the problem (and in fact more likely
            that the application will test very large sizes).<br>
            <br>
            <br>
            Our options are:<br>
            <br>
            A) Just accept this complexity and just hope that people
            write good code or use good libraries. (See above about
            testing...)<br>
            Note: we&#39;d need to set *some* lower bound for the value.<br=
>
            <br>
            B) Make the W3 API implementation add a level of protocol on
            top of the underlying IETF network protocol. This protocol
            could then deal with fragmenting messages on the sending
            side and reassembling them on the receiving side.<br>
            <br>
            C) Convince IETF WG to support arbitrarily sized messages at
            a protocol level, at least in the WebRTC context, similar to
            WebSockets.
            <div><br>
              <br>
              <br>
              -- <br>
              Randell Jesup -- rjesup a t mozilla d o t com<br>
              <br>
            </div>
            <div>
              <div>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb=
@ietf.org</a><br>
                <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    <pre cols=3D"72">--=20
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
  </div></div></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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On 3/3/2014 1:53 AM, Michael Tuexen wrote:
> On 03 Mar 2014, at 02:42, Randell Jesup <randell-ietf@jesup.org> wrote:
>
>
>> EMSGSIZE if the sendto() is larger than available buffer space, and the alternative (EOR mode) doesn't allow for interleaving of messages when sending at all (and I'm not sure it allows interleaving on reception either, but I didn't check right now).
> If you want to send messages larger than the socket send buffer, you must use the explicit
> EOR more. This is independent from using NDATA or not.

right.

> If you use NDATA, you can have on each outgoing stream a single message (order or unordered)
> being sent using explicit EOR mode and the calls can be interleaved. I don't think I have
> tested this up to know, but this is how it should work.
>
> rrs@: Just to double check: Am I right? Please note that RTCWeb doesn't use ordered and
>        unordered on the same stream, and you don't need to send more than one message with
>        explicit EOR on the same stream at the same time. Therefore the above simplification...

If this is the case, this should deal with the issue I believe, with one 
caveat for the W3 layer:

If we can only send one at a time, we get an issue where we call 
channel.send(blob1); channel.send(blob2);

the JS interface to send() is synchronous and can't block.  This implies 
that if send(blob1) hasn't completed, the send(blob2) can't start, and 
all following send()s must also be buffered in memory. For a blob, since 
a blob reference is read-only, this is likely ok (need to verify what 
happens if we haven't read the blob into memory - if this isn't the 
case, we have a problem).  For sending an ArrayBuffer (or string), we'll 
have to snapshot the buffer at send() time if it's queued since the 
referenced data may change.  However, this really isn't anything that 
new, since if the SCTP buffers are full you have to do something similar 
anyways, even without any messages using fragmentation.

I presume from your comment (since it wasn't entirely clear) that if I 
have channel.send(blob1); channel.send("foo") that "foo" can't be sent 
until blob1 has been fully transmitted (has hit EOR at the send side), 
even though "foo" doesn't require fragmentation, and so all send()s for 
a channel during a fragmented send() must be copied and queued.

>> Now, this is an implementation/API issue which could be fixed with some work, but so far as I know no work has been done on it.  So ndata will allow us to expand the max-sending size to
> This API work is part of getting NDATA implemented. Randall has implemented it, we do have
> code, but it needs some testing and cleaning up in the API which supports the scheduling...

So the last issue would be backwards compatibility.  If today an app 
must be written to query (via an unspecified W3 property) the max size 
you can reliably send() to the other side, then it must buy into (and 
test!) that functionality now and until it's pretty certain that all 
non-compliant implementations are gone.  This can take a while, even if 
you assume people update promptly, due to things like Firefox "ESR" 
releases (Extended Support for businesses, schools, etc).  Thes are 
valid for most of a year, and if we land ndata support right after one 
is created it could be closer to 1-1.5 years before if even starts to 
reach clients - and it may take even more time for all ESR users to 
approve/test it and update.

The arguments about having all the apps be required to implement their 
own fragmentation, etc for a considerable period adds lots of complexity 
to the apps.  Implementing the PPID chunking as the stopgap (and 
retiring it once ndata is universal) means the API to the application 
remains stable and simple.  Since all DataChannel implementations must 
implement queuing of JS send()s (since the SCTP bufferspace may be 
full), it's easy to support PPID fragmentation if you don't worry about 
memory-use optimization).  ndata not being available yet highlights this 
issue.

>> min(SCTP_BUFFER_SIZE - N, other-sides-max-receive-size).  It does not allow us to expand the size to any useful Blob size.
> Please note that you never have a limitation of the receive buffer send size, since SCTP
> would use partial delivery. So other-sides-max-receive-size is only relevant if there are
> buffering limits on top of SCTP.

I was referring to the SDP value for "minimum value I guarantee receiving".

We would need to specify that if ndata is agreed to (I presume we know 
that) that the application would be told an "infinite" value for 
max-I-can-receive (since we can't use the value from the SDP, as that 
predates the two sides creating the association and negotiating ndata).  
I don't want to renegotiate just to remove the max SCTP receive size.

   Randell

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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On 3/3/2014 6:18 AM, Justin Uberti wrote:
>
>
> On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup <randell-ietf@jesup.org 
> <mailto:randell-ietf@jesup.org>> wrote:
>
>     On 3/2/2014 4:27 PM, Justin Uberti wrote:
>>     I'm not sure I understand the actual problem here. We *are* going
>>     to handle arbitrary-sized messages once ndata gets done. The
>>     max-msg-size stuff is just a solution until ndata is deployed.
>>     (To cite your options, this is A followed by C).
>
>     So, here's where I think there may be a disconnect (and if I'm
>     wrong, great):
>
>     ndata solves the monopolization issue between streams, allowing
>     packets for different streams to be interleaved on the wire.  It
>     does not (so far as I know) relax the usrsctp library's returning
>     EMSGSIZE if the sendto() is larger than available buffer space,
>     and the alternative (EOR mode) doesn't allow for interleaving of
>     messages when sending at all (and I'm not sure it allows
>     interleaving on reception either, but I didn't check right now).
>
>     Now, this is an implementation/API issue which could be fixed with
>     some work, but so far as I know no work has been done on it.  So
>     ndata will allow us to expand the max-sending size to
>     min(SCTP_BUFFER_SIZE - N, other-sides-max-receive-size).  It does
>     not allow us to expand the size to any useful Blob size.
>
>
> Based on the discussion of ndata/eor in followup messages, I think 
> this problem will be addressed in the near future.

Hopefully so, though I hate forcing apps to include a bunch of code for 
backwards compat for a year or two (and many won't, or won't test it, 
etc).  That hurts the feature.  And today no one has ndata.

>
>>     That said, Send(blob) seems like a bit of a footgun to me anyway;
>>     I think apps are going to typically avoid it, a) to avoid memory
>>     bloat, and b) to get progress indications. The only way out of
>>     that situation is to provide stream semantics, which seems like a
>>     lot to take on given that WebSockets hasn't gone that route. I
>>     also agree that we shouldn't try to go the (B) route that you
>>     mention.
>
>     Well, Streams would be a good thing once they're done. Progress
>     indication could be added at the W3 level without much problem. 
>     The memory issue should be resolvable in a number of ways, per my
>     discussion with Sicking.  Note that application chunking has a
>     serious memory issue today in that the File Writer API hasn't been
>     agreed to; I think there's progress towards eventually adopting a
>     version of the Filesystem API (with changes), but that will be a
>     while.  Again, Stream should help - and note that the semantics
>     for Blob reception allow it to be written to disk as it's received
>     and not held entirely in memory when it's handed to the
>     application, which is NOT possible today for application chunking.
>
>
> Agree Streams will be useful when done, but that's probably post-1.0 
> (as Martin says, we should first let it be applied to WebSockets). 
> However I don't think app chunking will have any memory problems - the 
> chunks will be small enough that they can be easily spooled to disk as 
> they come in.

Agree on Streams (post 1.0).  Chunking does have memory problems unless 
you have a File Writer API (we don't) or a Filesystem API (we don't, 
though my understanding is that we believe that with some changes this 
can be adoptable in the future).  You have two choices today for app 
chunking (and I assume using blob slices for sending chunks):

1) create a blob from each received chunk then then create a new blob 
from the concatenation of all the sub-blobs.  This requires a large 
memory hit, though I suppose in theory maybe it could be done without a 
hit (painfully).  Even if the memory hit is avoided there will be a 
large disk IO hit to merge all those files and write a new file from them.

2) hold all the parts in memory, then write the entire thing as a blob 
after the last part is received.  Large memory and disk IO hit on 
reception of final part.

>
>>
>>     So I still think manual chunking seems like a reasonable thing to
>>     support. I don't quite get the congestion control concerns; just
>>     because there is a max chunk size doesn't mean the impl can't
>>     buffer multiple chunks in bufferedAmount; the app could let that
>>     fill up to a degree to avoid having to poll constantly to prevent
>>     underrun.
>
>     On a slow link this will work if the browser isn't janky.  On a
>     fast link GC pauses and other things may cause the buffer to drain
>     out and go idle for significant periods.
>
>
> Since the amount buffered is up to the app, the app can just increase 
> the buffer size to provide enough data to cover the next second or so 
> of transfer.

How does the app increase the size of the internal SCTP buffers? (And 
that buffer is shared by all channels, of course).

> Note that app chunking will clearly be needed in some of the more 
> interesting cases we want to support, such as where the transfer is 
> not 1:1. For swarm-style downloading, we need to make sure app 
> chunking works well (and I think it does).

Agreed, we need good solutions for application chunking (and thus my 
interest in resolving the Filesystem API issues in the W3).  My problem 
is that we don't have those today for the reception side - but this (for 
apps that *need* chunking) is largely a W3 issue.


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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    <div class="moz-cite-prefix">On 3/3/2014 6:18 AM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra"><br>
          <br>
          <div class="gmail_quote">On Sun, Mar 2, 2014 at 6:42 PM,
            Randell Jesup <span dir="ltr">&lt;<a moz-do-not-send="true"
                href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
              <div text="#000000" bgcolor="#FFFFFF">
                <div class="">
                  <div>On 3/2/2014 4:27 PM, Justin Uberti wrote:<br>
                  </div>
                  <blockquote type="cite">
                    <div dir="ltr">I'm not sure I understand the actual
                      problem here. We *are* going to handle
                      arbitrary-sized messages once ndata gets done. The
                      max-msg-size stuff is just a solution until ndata
                      is deployed. (To cite your options, this is A
                      followed by C).</div>
                  </blockquote>
                  <br>
                </div>
                So, here's where I think there may be a disconnect (and
                if I'm wrong, great):<br>
                <br>
                ndata solves the monopolization issue between streams,
                allowing packets for different streams to be interleaved
                on the wire.&nbsp; It does not (so far as I know) relax the
                usrsctp library's returning EMSGSIZE if the sendto() is
                larger than available buffer space, and the alternative
                (EOR mode) doesn't allow for interleaving of messages
                when sending at all (and I'm not sure it allows
                interleaving on reception either, but I didn't check
                right now).<br>
                <br>
                Now, this is an implementation/API issue which could be
                fixed with some work, but so far as I know no work has
                been done on it.&nbsp; So ndata will allow us to expand the
                max-sending size to min(SCTP_BUFFER_SIZE - N,
                other-sides-max-receive-size).&nbsp; It does not allow us to
                expand the size to any useful Blob size.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Based on the discussion of ndata/eor in followup
              messages, I think this problem will be addressed in the
              near future.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Hopefully so, though I hate forcing apps to include a bunch of code
    for backwards compat for a year or two (and many won't, or won't
    test it, etc).&nbsp; That hurts the feature.&nbsp; And today no one has ndata.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>&nbsp;&nbsp;</div>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
              <div text="#000000" bgcolor="#FFFFFF">
                <div class="">
                  <br>
                  <blockquote type="cite">
                    <div dir="ltr">
                      <div> That said, Send(blob) seems like a bit of a
                        footgun to me anyway; I think apps are going to
                        typically avoid it, a) to avoid memory bloat,
                        and b) to get progress indications. The only way
                        out of that situation is to provide stream
                        semantics, which seems like a lot to take on
                        given that WebSockets hasn't gone that route. I
                        also agree that we shouldn't try to go the (B)
                        route that you mention. </div>
                    </div>
                  </blockquote>
                  <br>
                </div>
                Well, Streams would be a good thing once they're done.
                Progress indication could be added at the W3 level
                without much problem.&nbsp; The memory issue should be
                resolvable in a number of ways, per my discussion with
                Sicking.&nbsp; Note that application chunking has a serious
                memory issue today in that the File Writer API hasn't
                been agreed to; I think there's progress towards
                eventually adopting a version of the Filesystem API
                (with changes), but that will be a while.&nbsp; Again, Stream
                should help - and note that the semantics for Blob
                reception allow it to be written to disk as it's
                received and not held entirely in memory when it's
                handed to the application, which is NOT possible today
                for application chunking.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Agree Streams will be useful when done, but that's
              probably post-1.0 (as Martin says, we should first let it
              be applied to WebSockets). However I don't think app
              chunking will have any memory problems - the chunks will
              be small enough that they can be easily spooled to disk as
              they come in. <br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Agree on Streams (post 1.0).&nbsp; Chunking does have memory problems
    unless you have a File Writer API (we don't) or a Filesystem API (we
    don't, though my understanding is that we believe that with some
    changes this can be adoptable in the future).&nbsp; You have two choices
    today for app chunking (and I assume using blob slices for sending
    chunks):<br>
    <br>
    1) create a blob from each received chunk then then create a new
    blob from the concatenation of all the sub-blobs.&nbsp; This requires a
    large memory hit, though I suppose in theory maybe it could be done
    without a hit (painfully).&nbsp; Even if the memory hit is avoided there
    will be a large disk IO hit to merge all those files and write a new
    file from them.<br>
    <br>
    2) hold all the parts in memory, then write the entire thing as a
    blob after the last part is received.&nbsp; Large memory and disk IO hit
    on reception of final part.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
              <div text="#000000" bgcolor="#FFFFFF">
                <div class=""> <br>
                  <blockquote type="cite">
                    <div dir="ltr">
                      <div><br>
                      </div>
                      <div>So I still think manual chunking seems like a
                        reasonable thing to support. I don't quite get
                        the congestion control concerns; just because
                        there is a max chunk size doesn't mean the impl
                        can't buffer multiple chunks in bufferedAmount;
                        the app could let that fill up to a degree to
                        avoid having to poll constantly to prevent
                        underrun.</div>
                    </div>
                  </blockquote>
                  <br>
                </div>
                On a slow link this will work if the browser isn't
                janky.&nbsp; On a fast link GC pauses and other things may
                cause the buffer to drain out and go idle for
                significant periods.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Since the amount buffered is up to the app, the app can
              just increase the buffer size to provide enough data to
              cover the next second or so of transfer. <br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    How does the app increase the size of the internal SCTP buffers?&nbsp;
    (And that buffer is shared by all channels, of course).<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>Note that app chunking will clearly be needed in some
              of the more interesting cases we want to support, such as
              where the transfer is not 1:1. For swarm-style
              downloading, we need to make sure app chunking works well
              (and I think it does).</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Agreed, we need good solutions for application chunking (and thus my
    interest in resolving the Filesystem API issues in the W3).&nbsp; My
    problem is that we don't have those today for the reception side -
    but this (for apps that *need* chunking) is largely a W3 issue.<br>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
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Subject: Re: [rtcweb] Open data channel issues
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On 03 Mar 2014, at 11:25, Randell Jesup <randell-ietf@jesup.org> wrote:

> On 3/3/2014 1:53 AM, Michael Tuexen wrote:
>> On 03 Mar 2014, at 02:42, Randell Jesup <randell-ietf@jesup.org> =
wrote:
>>=20
>>=20
>>> EMSGSIZE if the sendto() is larger than available buffer space, and =
the alternative (EOR mode) doesn't allow for interleaving of messages =
when sending at all (and I'm not sure it allows interleaving on =
reception either, but I didn't check right now).
>> If you want to send messages larger than the socket send buffer, you =
must use the explicit
>> EOR more. This is independent from using NDATA or not.
>=20
> right.
>=20
>> If you use NDATA, you can have on each outgoing stream a single =
message (order or unordered)
>> being sent using explicit EOR mode and the calls can be interleaved. =
I don't think I have
>> tested this up to know, but this is how it should work.
>>=20
>> rrs@: Just to double check: Am I right? Please note that RTCWeb =
doesn't use ordered and
>>       unordered on the same stream, and you don't need to send more =
than one message with
>>       explicit EOR on the same stream at the same time. Therefore the =
above simplification...
>=20
> If this is the case, this should deal with the issue I believe, with =
one caveat for the W3 layer:
>=20
> If we can only send one at a time, we get an issue where we call =
channel.send(blob1); channel.send(blob2);
>=20
> the JS interface to send() is synchronous and can't block.  This =
implies that if send(blob1) hasn't completed, the send(blob2) can't =
start, and all following send()s must also be buffered in memory. For a =
blob, since a blob reference is read-only, this is likely ok (need to =
verify what happens if we haven't read the blob into memory - if this =
isn't the case, we have a problem).  For sending an ArrayBuffer (or =
string), we'll have to snapshot the buffer at send() time if it's queued =
since the referenced data may change.  However, this really isn't =
anything that new, since if the SCTP buffers are full you have to do =
something similar anyways, even without any messages using =
fragmentation.
>=20
> I presume from your comment (since it wasn't entirely clear) that if I =
have channel.send(blob1); channel.send("foo") that "foo" can't be sent =
until blob1 has been fully transmitted (has hit EOR at the send side), =
even though "foo" doesn't require fragmentation, and so all send()s for =
a channel during a fragmented send() must be copied and queued.
Correct. And if you are on an ordered channel, there is no alternative, =
right? On an unordered,
you can't interleave messages on a single stream...
>=20
>>> Now, this is an implementation/API issue which could be fixed with =
some work, but so far as I know no work has been done on it.  So ndata =
will allow us to expand the max-sending size to
>> This API work is part of getting NDATA implemented. Randall has =
implemented it, we do have
>> code, but it needs some testing and cleaning up in the API which =
supports the scheduling...
>=20
> So the last issue would be backwards compatibility.  If today an app =
must be written to query (via an unspecified W3 property) the max size =
you can reliably send() to the other side, then it must buy into (and =
test!) that functionality now and until it's pretty certain that all =
non-compliant implementations are gone.  This can take a while, even if =
you assume people update promptly, due to things like Firefox "ESR" =
releases (Extended Support for businesses, schools, etc).  Thes are =
valid for most of a year, and if we land ndata support right after one =
is created it could be closer to 1-1.5 years before if even starts to =
reach clients - and it may take even more time for all ESR users to =
approve/test it and update.
>=20
> The arguments about having all the apps be required to implement their =
own fragmentation, etc for a considerable period adds lots of complexity =
to the apps.  Implementing the PPID chunking as the stopgap (and =
retiring it once ndata is universal) means the API to the application =
remains stable and simple.  Since all DataChannel implementations must =
implement queuing of JS send()s (since the SCTP bufferspace may be =
full), it's easy to support PPID fragmentation if you don't worry about =
memory-use optimization).  ndata not being available yet highlights this =
issue.
>=20
>>> min(SCTP_BUFFER_SIZE - N, other-sides-max-receive-size).  It does =
not allow us to expand the size to any useful Blob size.
>> Please note that you never have a limitation of the receive buffer =
send size, since SCTP
>> would use partial delivery. So other-sides-max-receive-size is only =
relevant if there are
>> buffering limits on top of SCTP.
>=20
> I was referring to the SDP value for "minimum value I guarantee =
receiving".
Understood. My point was the following:

If NDATA is not yet supported, the sender can choose a message limit to =
avoid
monopolisation. It is a sender side thing.

So any limit the receiver is enforcing and communicating via SDP is for =
different
reasons. So if we agree on support for arbitrary size messages when we =
have
NDATA, then there is ne need to negotiate a message size limit. The =
sender just
enforces whatever he things is appropriate to mitigate monopolisation.
>=20
> We would need to specify that if ndata is agreed to (I presume we know =
that) that the application would be told an "infinite" value for =
max-I-can-receive (since we can't use the value from the SDP, as that =
predates the two sides creating the association and negotiating ndata).  =
I don't want to renegotiate just to remove the max SCTP receive size.
Correct. See my above statement. Why negotiate at all if we agree on the =
receiver being
able to handle arbitrary size messages?

Best regards
Michael
>=20
>  Randell
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
>=20
>=20


From nobody Mon Mar  3 04:22:47 2014
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-07.txt - Comment on section 6 title
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On 12 Feb 2014, at 18:56, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:

>> Hi,
>>=20
>>> I suggest that this just be called "Data Channel", not "WebRTC Data
>> Channel". (Or pick some other neutral name - perhaps SCTP Data =
Channel.)
>>=20
>> Agree. A more generic name would be useful.
>>=20
>> "Simple Data Channel"
>> "Simple SCTP Data Channel"
>> Etc...
>=20
> [Raju] How about "Application Data Channel"?
What about:
"The Usage of SCTP for Data Channels"

Best regards
Michael
>=20
>> (But it would still make sense for the JS API to have a =
WebRTC-specific name.)
> +1
>=20
> -raju
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 23 Feb 2014, at 00:20, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> (Catching up before the meeting)
> Here are some comments on the latest version. There don't necessarily =
relate to recent changes. Rather they come from my perspective trying to =
use this for CLUE, whether using a webrtc endpoint or not.
>=20
> * Section 5:
>=20
> This section says:
>=20
>   Each SCTP user message contains a so called Payload Protocol
>   Identifier (PPID) that is passed to SCTP by its upper layer and sent
>   to its peer.  This value can be used to multiplex multiple protocols
>   over a single SCTP association.  The sender provides for each
>   protocol a specific PPID and the receiver can demultiplex the
>   messages based on the received PPID.  The PPID is used to =
distinguish
>   UTF-8 encoded user data and binary encoded userdata.  The Data
>   Channel Establishment Protocol defined in
>   [I-D.ietf-rtcweb-data-protocol] uses also a specific PPID to be
>   distinguished from user data.
>=20
> The language has proved extremely confusing to several people who =
haven't followed the development and discussion of this from the =
beginning. It is because of the extreme ambiguity in the use of =
"protocol". While I guess we can't rename PPID, the other language could =
be clarified.
>=20
> The problem is that we have SCTP protocol, "WebRTC Data Channel =
Establishment Protocol", and within the DATA_CHANNEL_OPEN message of the =
WebRTC Data Channel Establishment Protocol we have a "Protocol" field.
>=20
> We also have layering of protocols and implementations. So it isn't =
clear above who "sender" and "receiver" above apply to. (Could be code =
that is calling the SCTP stack, or code that is calling the "data =
channel stack", or the browser, or a javascript application on top of =
browser.
>=20
> IIUC, javascript users don't have any visibility to the PPID. But =
others implementing data channels may.
>=20
> How about:
>=20
>   Each SCTP user message contains a so called Payload Protocol
>   Identifier (PPID) that is passed to SCTP by the data channel layer
>   and sent to its peer.  This value is used to multiplex WebRTC Data
>   Channel Establishment Protocol messages (defined in =
[I-D.ietf-rtcweb-
>   data-protocol]) with messages containing user data on a data
>   channel. The PPID is also used to distinguish UTF-8 encoded user
>   data and binary encoded user data.
The above text has changed in the meantime due to comments. It currently =
reads:

<t>Each SCTP user message contains a Payload Protocol Identifier (PPID)
that is passed to SCTP by its upper layer on the sending side and
provided to its upper layer on the receiving side.
The PPID can be used to multiplex/demultiplex multiple upper layers over
a single SCTP association.
In the WebRTP context, the PPID is used to distinguish between
UTF-8 encoded user data,
binary encoded userdata and
the Data Channel Establishment Protocol defined in
<xref target=3D'I-D.ietf-rtcweb-data-protocol'/>.
Please note that the PPID is not accessible via the Javascript API.</t>

Does this address your issues?
>=20
> Also in this section is Figure 2 that shows layering. I'm unclear if =
"WEBRTC DATA" in intended to include rtcweb-data-protocol or not. That =
is defined in a separate draft, is mandatory to support but optional to =
use. So it could go either way. Might be good to clarify:
>=20
>=20
>                 +---------+
>                 |CHANNEL  |
>                 |PROTOCOL |
>                 +---------+
>                 |DATA     |
>                 |CHANNEL  |
>                 +---------+
>                 | SCTP    |
>   +-----------------------+
>   | STUN | SRTP | DTLS    |
>   +-----------------------+
>   |         ICE           |
>   +-----------------------+
>   | UDP1 | UDP2 | ...     |
>   +-----------------------+
Hmm. What about using
              +------+------+------+
              | DCEP | UTF-8|Binary|
              |      | data | data |
              +------+------+------+
              |        SCTP        |
+----------------------------------+
| STUN | SRTP |        DTLS        |
+----------------------------------+
|         ICE                      |
+----------------------------------+
| UDP1 | UDP2 | ...                |
+----------------------------------+

which also makes clear how the PPID is used to multiplex/demultiplex the =
different
upper layers?
>=20
> * Section 6.5
>=20
> This section contains:
>=20
>   Data channels can be opened by using internal or external
>   negotiation.  The details are out of scope of this document.
>=20
>   A simple protocol for internal negotiation is specified in
>   [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>=20
> But internal and external negotiation are not defined in this =
document.
>=20
> I *thought* that internal negotiation was by definition negotiation by =
use of rtcweb-data-protocol. (draft-ejzak-mmusic-data-channel-sdpneg-00 =
thinks so too, but calls it "in-band negotiation".)
We can use in-band and out-of-band negotiation
>=20
> There should be a good definition of these terms, or reference to one. =
And more discussion if there can be other kinds of internal negotiation. =
(If so, how would one be chosen?)
Up to now, there is no other way.... Would you prefer that?
>=20
> * Section 6.6:
>=20
> Say:
>=20
>   All data sent on a Channel in both directions MUST be sent over the
>   underlying stream using the reliability defined when the Channel was
>   opened unless the options are changed, or per-message options are
>   specified by a higher level.
>=20
> Is the recipient to consider it an error if messages are received with =
options different from those defined for the channel?
Please note that the receiver could double the the ordering. The =
receiver could also
detect that messages have been abandoned on ordered streams, but not on =
unordered.
For successfully delivered messages, the receiver can not detect if the =
message was
sent reliably or not. There is no explicit way for the receiver to =
verify the PR SCTP policy
or the corresponding value. So I would vote for ignoring...
>=20
> Also, is it an error if messages are received with a PPID that isn't =
specified in Section 8? (And what about PPID 50?)
>=20
> How are such channel errors to be treated?
Reset the channel? Do we want to specify that?
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 25 Feb 2014, at 10:48, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> Q1:
>>>=20
>>> Section 5 says:
>>>=20
>>> 	"SCTP as specified in [RFC4960] MUST be used in
>>>  	combination with the extension defined in [RFC3758] and provides =
the
>>>  	following interesting features for transporting non-media data
>>>  	between browsers:"
>>>=20
>>> I suggest to say:
>>>=20
>>> 	"The SCTP extension defined in [RFC3758] MUST be supported, and
>>>    	provides the following features:
>>>=20
>> But this implies that the listed features are provided by the =
extension defined in RFC 3758, which is not true. The features are =
provided by the combination of RFC 4960 and RFC 3758.
>=20
> Correct. My mistake.
>=20
> So, my suggestion would then be to remove "as specified in" and =
"interesting".
>=20
> 	"SCTP [RFC4960] MUST be used in combination with the extension =
defined in [RFC3758] and=20
> 	provides the following features for transporting non-media data =
between browsers:"
>=20
> The reference implicitly means "as specified in", in my opinon :)
Removed the 'interesting'.
The document use a 'as specified in' or similar text in several places. =
For consistency
it should be removed in all places. So I kept it for now to be =
consistent.

Best regards
Michael
>=20
> ----------
>=20
>>> Q3:
>>>=20
>>> Section 5 says:
>>>=20
>>> 	"This value can be used to multiplex multiple protocols
>>>  	over a single SCTP association.  The sender provides for each
>>>  	protocol a specific PPID and the receiver can demultiplex the
>>>  	messages based on the received PPID."
>>>=20
>>> I think we discussed this earlier. The PPID value CAN NOT (at least =
not as used by WebRTC) be used to multiplex multiple protocols=20
>>> over a single SCTP association. If you create multiple channels, for =
multiple protocols, the same PPID values may be used for each protocol.
>>>=20
>>> The PPID CAN, however, be used within a data channel, e.g. to =
demultiplex the DCP from channel data.
>>=20
>> I guess the problem is that I consider 'protocol' as something =
running on top of SCTP like DCEP, Binary of String data, you consider =
'protocol'=20
>> as something running on top of Binary of String data and the =
'protocol' being identified in the JS protocol string.
>>=20
>> What about using:
>>=20
>> <t>Each SCTP user message contains a Payload Protocol Identifier =
(PPID) that is passed to SCTP by its upper layer on the sending side and =
provided to its upper layer on the receiving side.
>> The PPID can be used to multiplex/demultiplex multiple upper layers =
over a single SCTP association.
>> In the WebRTP context, the PPID is used to distinguish between
>> UTF-8 encoded user data, binary encoded userdata and the Data Channel =
Establishment Protocol defined in <xref =
target=3D'I-D.ietf-rtcweb-data-protocol'/>.
>> Please note that the PPID is not accessable via the Javascript =
API.</t>
>>=20
>> Does this address your issue?
>=20
> Instead of talking about SCTP associations, could we talk about data =
channels? I.e. the PPID can be used to multiplex/demultiplex multiple =
upper layers within a single data channel?
>=20
> Because, if you have multiple data channels on a single SCTP =
association, you will use the stream IDs  - not the PPIDs - to =
multiplex/demultiplex between the channels.
>=20
> ----------
>=20
>>> Q5:
>>>=20
>>> Sometimes you say "SCTP" and sometimes "SCTP as defined in =
[RFC4960]".
>>>=20
>>> I suggest to e.g. use the reference on the first occurrence, and =
then only use "SCTP".
>>=20
>> I double checked:
>> SCTP as defined in [RFC4960] is used in
>> * In the Introduction (first occurrence) talking about SCTP / UDP
>> * In the Introduction to make sure which specifications are used: RFC =
4960 and RFC 3758
>=20
> In the 2nd paragraph of the Introduction, the text says:
>=20
> 	"SCTP as specified in [RFC4960] with the partial reliability =
extension defined in [RFC3758]"
>=20
> Could you remove "as specified in", and say?
>=20
> 	"SCTP [RFC4960] with the partial reliability extension defined =
in [RFC3758]"
>=20
>=20
>> * In section 5 when used with RFC 2119 language.
>=20
> I still think you could remove "as specified in", and say:
>=20
> 	"SCTP [RFC4960] MUST be used in combination with the extension =
defined..."
>=20
>> * In section 6 to make clear what relates to RFC 4960 and what to the =
NDATA extension.
>>=20
>> All of these are intentional. Which one do you think is not =
necessary?
>=20
> In section 6.6, I think you could remove "specified in", and say:
>=20
> 	"The SCTP base protocol [RFC4960]..."
>=20
>=20
> I am not religious about these things, but again: in my opinion adding =
the reference means "as specified in", and makes the text easier to read =
:)
>=20
> Regards,
>=20
> Christer
>=20
>=20


From nobody Mon Mar  3 05:25:41 2014
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'IESG IESG'" <iesg@ietf.org>
References: <20140214030712.30321.21888.idtracker@ietfa.amsl.com> <CAKhHsXGzA=ZTFGTK7ht9hQbfG70iqKrDtxrZCdQNNMzBYZCk8A@mail.gmail.com> <530604ea.c5bf440a.5cfd.ffffde18SMTPIN_ADDED_BROKEN@mx.google.com> <CAKhHsXGsp6ma6Ko9op+YFRqSGM_Ex-jFo_fjz69rN0SEHfNK2A@mail.gmail.com> <CALDtMrKb3_38Rs0vaGnpEvNvTYz8YUTo89STvLJNXfkfdipDSQ@mail.gmail.com> <93BEDDC39A54294B9E78C7860516FA4724AA4422@AZ-US1EXMB06.global.avaya.com> <B7FA2629-6D48-4569-BB62-56395C3EE4BC@cisco.com> <04ff01cf31b4$8eb73780$ac25a680$@stahl@intertex.se> <AA208926-C949-4580-B20B-DCF172D3C21B@cisco.com> 
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Subject: Re: [rtcweb] [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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Cullen,

Illution
I am checking the IETF archive for links - Memory you know...
I feel I also have an email from Harald to answer




You will not see the below
/Kalle
Allt v=E4l
-----Ursprungligt meddelande-----
Fr=E5n: Karl Stahl [mailto:karl.stahl@intertex.se]=20
Skickat: den 3 mars 2014 10:22
Till: 'Cullen Jennings (fluffy)'; 'IESG IESG'
Kopia: 'Magnus Westerlund'; 'Simon Perreault'; 'Ted Hardie'; 'Gonzalo
Camarillo'; 'rtcweb@ietf.org'; 'tram@ietf.org'; 'Spencer Dawkins'
=C4mne: SV: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter
Clarification regardig Milestone 3: TURN server auto-discovery mechanism =
for
enterprise and ISPs

Hi Cullen,

Memory may be an illusion, still I cannot understand how there can be an
ever growing archive in the IETF

/Karl

Allt v=E4l
Hi Cullen,

Not to cause more lost in the threads (I am also) I am addressing both =
your
emails in one, and copy separately to the other guys in the first email. =


Further Inline -->

>From http://www.ietf.org/mail-archive/web/tram/current/msg00275.html
Let me also point out draft-deng-tram-isp-turn-00
http://www.ietf.org/mail-archive/web/tram/current/msg00214.html from =
China
Mobile that appeared on this mailing list yesterday. It points out the =
need
and willingness from the ISP/NSP side to do something about QoS for =
WebRTC
traffic, that they expect to be large and have to bring to their =
customers
with best QoE. In fact they and some more (a huge European carrier and =
the
cable operators in general - CableLabs) have expressed similar concerns =
to
me *=93This is exactly something we want to work on as the current way =
will
cause severe problem on traffic when rtcweb applications get popular=94* =
is a
direct quote from one of those.

So, in answer to Simon Perreault [simon.perreault at viagenie.ca]=92s =
question
in another thread the 13th:
a) Is this a real problem that is worth fixing?
I think we can be confident that the answer is *YES*. Only these three
ISP/NSP referred to, may represent 50%(?) of the universe=92s IP =
accesses! And
the other will follow these most forward looking carriers.=20


-----Ursprungligt meddelande-----
Fr=E5n: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
Skickat: den 26 februari 2014 01:49
Till: Karl Stahl
Kopia: Alan Johnston; rtcweb@ietf.org; tram@ietf.org; Bernard Aboba; =
David
Singer; Harald Tveit Alvestrand; Marc Robins; Eric Burger; Mary Barnes;
Henning Schulzrinne
=C4mne: Re: [rtcweb] [tram] I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt


> Karl,=20

> I am totally lost on this thread. Could you start a new tmead that
summarize what the issues is, what seems to > be the point of debate, =
and
what your view is on what we should do. I think that would help make
progress.=20

> Thank you,=20

> Cullen

I just did something like that in:
[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff=20
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html=20
Hoping this thread subject amplifies what this SHOULD be about. (An how =
it
will lead to the summary, after all diversion.)

Actually, the outcome of "Interfacing to QoS" will be=20


I challenge anyone that this will be outcome, and will start =
"Interfacing to
QoS" with what IP "Interfacing to QoS" that we have the tiny ADSL modem =
IX78
that implements such CLEAR INTERFACE and the actual QoS methods, for =
both
the Congestion, Default gateway, Surf Pipe=20
Manipulating the RTP extension header etc etc on 6 USD CPU and also =
include
the ADSL part where that quality handling is based on heavy 57 bytes
chopping up
   at this exact=20




-----Ursprungligt meddelande-----
Fr=E5n: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]=20
Skickat: den 26 februari 2014 02:06
Till: Karl Stahl; IESG IESG
Kopia: Magnus Westerlund; Simon Perreault; Ted Hardie; Gonzalo =
Camarillo;
rtcweb@ietf.org; tram@ietf.org; Spencer Dawkins
=C4mne: Re: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter
Clarification regardig Milestone 3: TURN server auto-discovery mechanism =
for
enterprise and ISPs



On Feb 25, 2014, at 7:02 AM, Karl Stahl <karl.stahl@intertex.se> wrote:

> To the TRAM WG and RTCWEB WG and ADs:
> =20
> It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed =
and
encouraged to route quality demanding WebRTC media into their IP pipes =
that
are capable of transporting real-time traffic without quality issues, =
using
TURN servers.
> =20

Reading the charter, the above is *not* at all a clear objective of the =
WG
(note I am not the chair of this WG or the responsible AD).=20

That said, I think you have pointed out this charter is abysmally vague =
- it
does not say what the WG is not going to do. If I decided to do BGP for
routing updates over TURN it would be within the scope of this charter.=20

My advice to the responsible AD is recharter this WG before IETF 90 or =
close
it. I would be glad to help write a charter that is not an infinite =
blank
cheque.=20







From nobody Mon Mar  3 05:29:21 2014
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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Mar 2014 05:28:54 -0800
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On Mon, Mar 3, 2014 at 3:44 AM, Randell Jesup <randell-ietf@jesup.org>wrote:

>  On 3/3/2014 6:18 AM, Justin Uberti wrote:
>
>
>
> On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup <randell-ietf@jesup.org>wrote:
>
>>  On 3/2/2014 4:27 PM, Justin Uberti wrote:
>>
>> I'm not sure I understand the actual problem here. We *are* going to
>> handle arbitrary-sized messages once ndata gets done. The max-msg-size
>> stuff is just a solution until ndata is deployed. (To cite your options,
>> this is A followed by C).
>>
>>
>>  So, here's where I think there may be a disconnect (and if I'm wrong,
>> great):
>>
>> ndata solves the monopolization issue between streams, allowing packets
>> for different streams to be interleaved on the wire.  It does not (so far
>> as I know) relax the usrsctp library's returning EMSGSIZE if the sendto()
>> is larger than available buffer space, and the alternative (EOR mode)
>> doesn't allow for interleaving of messages when sending at all (and I'm not
>> sure it allows interleaving on reception either, but I didn't check right
>> now).
>>
>> Now, this is an implementation/API issue which could be fixed with some
>> work, but so far as I know no work has been done on it.  So ndata will
>> allow us to expand the max-sending size to min(SCTP_BUFFER_SIZE - N,
>> other-sides-max-receive-size).  It does not allow us to expand the size to
>> any useful Blob size.
>>
>
>  Based on the discussion of ndata/eor in followup messages, I think this
> problem will be addressed in the near future.
>
>
> Hopefully so, though I hate forcing apps to include a bunch of code for
> backwards compat for a year or two (and many won't, or won't test it,
> etc).  That hurts the feature.  And today no one has ndata.
>
>
>
>
>>
>>   That said, Send(blob) seems like a bit of a footgun to me anyway; I
>> think apps are going to typically avoid it, a) to avoid memory bloat, and
>> b) to get progress indications. The only way out of that situation is to
>> provide stream semantics, which seems like a lot to take on given that
>> WebSockets hasn't gone that route. I also agree that we shouldn't try to go
>> the (B) route that you mention.
>>
>>
>>  Well, Streams would be a good thing once they're done. Progress
>> indication could be added at the W3 level without much problem.  The memory
>> issue should be resolvable in a number of ways, per my discussion with
>> Sicking.  Note that application chunking has a serious memory issue today
>> in that the File Writer API hasn't been agreed to; I think there's progress
>> towards eventually adopting a version of the Filesystem API (with changes),
>> but that will be a while.  Again, Stream should help - and note that the
>> semantics for Blob reception allow it to be written to disk as it's
>> received and not held entirely in memory when it's handed to the
>> application, which is NOT possible today for application chunking.
>>
>
>  Agree Streams will be useful when done, but that's probably post-1.0 (as
> Martin says, we should first let it be applied to WebSockets). However I
> don't think app chunking will have any memory problems - the chunks will be
> small enough that they can be easily spooled to disk as they come in.
>
>
> Agree on Streams (post 1.0).  Chunking does have memory problems unless
> you have a File Writer API (we don't) or a Filesystem API (we don't, though
> my understanding is that we believe that with some changes this can be
> adoptable in the future).  You have two choices today for app chunking (and
> I assume using blob slices for sending chunks):
>
> 1) create a blob from each received chunk then then create a new blob from
> the concatenation of all the sub-blobs.  This requires a large memory hit,
> though I suppose in theory maybe it could be done without a hit
> (painfully).  Even if the memory hit is avoided there will be a large disk
> IO hit to merge all those files and write a new file from them.
>
> 2) hold all the parts in memory, then write the entire thing as a blob
> after the last part is received.  Large memory and disk IO hit on reception
> of final part.
>
>
>
>>
>>  So I still think manual chunking seems like a reasonable thing to
>> support. I don't quite get the congestion control concerns; just because
>> there is a max chunk size doesn't mean the impl can't buffer multiple
>> chunks in bufferedAmount; the app could let that fill up to a degree to
>> avoid having to poll constantly to prevent underrun.
>>
>>
>>  On a slow link this will work if the browser isn't janky.  On a fast
>> link GC pauses and other things may cause the buffer to drain out and go
>> idle for significant periods.
>>
>
>  Since the amount buffered is up to the app, the app can just increase
> the buffer size to provide enough data to cover the next second or so of
> transfer.
>
>
> How does the app increase the size of the internal SCTP buffers?  (And
> that buffer is shared by all channels, of course).
>

What I meant is the app can make multiple calls to send() at a time, each
with a chunk, which will be buffered in the api layer and increase
bufferedAmount.

>
>
>   Note that app chunking will clearly be needed in some of the more
> interesting cases we want to support, such as where the transfer is not
> 1:1. For swarm-style downloading, we need to make sure app chunking works
> well (and I think it does).
>
>
> Agreed, we need good solutions for application chunking (and thus my
> interest in resolving the Filesystem API issues in the W3).  My problem is
> that we don't have those today for the reception side - but this (for apps
> that *need* chunking) is largely a W3 issue.
>
>
>
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e013a0d420af76e04f3b3c611
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Mar 3, 2014 at 3:44 AM, Randell Jesup <span dir=3D"ltr">&lt=
;<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@j=
esup.org</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D"">
    <div>On 3/3/2014 6:18 AM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_extra"><br>
          <br>
          <div class=3D"gmail_quote">On Sun, Mar 2, 2014 at 6:42 PM,
            Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-i=
etf@jesup.org" target=3D"_blank">randell-ietf@jesup.org</a>&gt;</span>
            wrote:<br>
            <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">
              <div text=3D"#000000" bgcolor=3D"#FFFFFF">
                <div>
                  <div>On 3/2/2014 4:27 PM, Justin Uberti wrote:<br>
                  </div>
                  <blockquote type=3D"cite">
                    <div dir=3D"ltr">I&#39;m not sure I understand the actu=
al
                      problem here. We *are* going to handle
                      arbitrary-sized messages once ndata gets done. The
                      max-msg-size stuff is just a solution until ndata
                      is deployed. (To cite your options, this is A
                      followed by C).</div>
                  </blockquote>
                  <br>
                </div>
                So, here&#39;s where I think there may be a disconnect (and
                if I&#39;m wrong, great):<br>
                <br>
                ndata solves the monopolization issue between streams,
                allowing packets for different streams to be interleaved
                on the wire.=C2=A0 It does not (so far as I know) relax the
                usrsctp library&#39;s returning EMSGSIZE if the sendto() is
                larger than available buffer space, and the alternative
                (EOR mode) doesn&#39;t allow for interleaving of messages
                when sending at all (and I&#39;m not sure it allows
                interleaving on reception either, but I didn&#39;t check
                right now).<br>
                <br>
                Now, this is an implementation/API issue which could be
                fixed with some work, but so far as I know no work has
                been done on it.=C2=A0 So ndata will allow us to expand the
                max-sending size to min(SCTP_BUFFER_SIZE - N,
                other-sides-max-receive-size).=C2=A0 It does not allow us t=
o
                expand the size to any useful Blob size.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Based on the discussion of ndata/eor in followup
              messages, I think this problem will be addressed in the
              near future.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br></div>
    Hopefully so, though I hate forcing apps to include a bunch of code
    for backwards compat for a year or two (and many won&#39;t, or won&#39;=
t
    test it, etc).=C2=A0 That hurts the feature.=C2=A0 And today no one has=
 ndata.<div class=3D""><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_extra">
          <div class=3D"gmail_quote">
            <div>=C2=A0=C2=A0</div>
            <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">
              <div text=3D"#000000" bgcolor=3D"#FFFFFF">
                <div>
                  <br>
                  <blockquote type=3D"cite">
                    <div dir=3D"ltr">
                      <div> That said, Send(blob) seems like a bit of a
                        footgun to me anyway; I think apps are going to
                        typically avoid it, a) to avoid memory bloat,
                        and b) to get progress indications. The only way
                        out of that situation is to provide stream
                        semantics, which seems like a lot to take on
                        given that WebSockets hasn&#39;t gone that route. I
                        also agree that we shouldn&#39;t try to go the (B)
                        route that you mention. </div>
                    </div>
                  </blockquote>
                  <br>
                </div>
                Well, Streams would be a good thing once they&#39;re done.
                Progress indication could be added at the W3 level
                without much problem.=C2=A0 The memory issue should be
                resolvable in a number of ways, per my discussion with
                Sicking.=C2=A0 Note that application chunking has a serious
                memory issue today in that the File Writer API hasn&#39;t
                been agreed to; I think there&#39;s progress towards
                eventually adopting a version of the Filesystem API
                (with changes), but that will be a while.=C2=A0 Again, Stre=
am
                should help - and note that the semantics for Blob
                reception allow it to be written to disk as it&#39;s
                received and not held entirely in memory when it&#39;s
                handed to the application, which is NOT possible today
                for application chunking.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Agree Streams will be useful when done, but that&#39;s
              probably post-1.0 (as Martin says, we should first let it
              be applied to WebSockets). However I don&#39;t think app
              chunking will have any memory problems - the chunks will
              be small enough that they can be easily spooled to disk as
              they come in. <br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br></div>
    Agree on Streams (post 1.0).=C2=A0 Chunking does have memory problems
    unless you have a File Writer API (we don&#39;t) or a Filesystem API (w=
e
    don&#39;t, though my understanding is that we believe that with some
    changes this can be adoptable in the future).=C2=A0 You have two choice=
s
    today for app chunking (and I assume using blob slices for sending
    chunks):<br>
    <br>
    1) create a blob from each received chunk then then create a new
    blob from the concatenation of all the sub-blobs.=C2=A0 This requires a
    large memory hit, though I suppose in theory maybe it could be done
    without a hit (painfully).=C2=A0 Even if the memory hit is avoided ther=
e
    will be a large disk IO hit to merge all those files and write a new
    file from them.<br>
    <br>
    2) hold all the parts in memory, then write the entire thing as a
    blob after the last part is received.=C2=A0 Large memory and disk IO hi=
t
    on reception of final part.<div class=3D""><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_extra">
          <div class=3D"gmail_quote">
            <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">
              <div text=3D"#000000" bgcolor=3D"#FFFFFF">
                <div> <br>
                  <blockquote type=3D"cite">
                    <div dir=3D"ltr">
                      <div><br>
                      </div>
                      <div>So I still think manual chunking seems like a
                        reasonable thing to support. I don&#39;t quite get
                        the congestion control concerns; just because
                        there is a max chunk size doesn&#39;t mean the impl
                        can&#39;t buffer multiple chunks in bufferedAmount;
                        the app could let that fill up to a degree to
                        avoid having to poll constantly to prevent
                        underrun.</div>
                    </div>
                  </blockquote>
                  <br>
                </div>
                On a slow link this will work if the browser isn&#39;t
                janky.=C2=A0 On a fast link GC pauses and other things may
                cause the buffer to drain out and go idle for
                significant periods.</div>
            </blockquote>
            <div><br>
            </div>
            <div>Since the amount buffered is up to the app, the app can
              just increase the buffer size to provide enough data to
              cover the next second or so of transfer. <br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br></div>
    How does the app increase the size of the internal SCTP buffers?=C2=A0
    (And that buffer is shared by all channels, of course).</div></blockquo=
te><div><br></div><div>What I meant is the app can make multiple calls to s=
end() at a time, each with a chunk, which will be buffered in the api layer=
 and increase bufferedAmount.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF"><d=
iv class=3D""><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_extra">
          <div class=3D"gmail_quote">
            <div>Note that app chunking will clearly be needed in some
              of the more interesting cases we want to support, such as
              where the transfer is not 1:1. For swarm-style
              downloading, we need to make sure app chunking works well
              (and I think it does).</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br></div>
    Agreed, we need good solutions for application chunking (and thus my
    interest in resolving the Filesystem API issues in the W3).=C2=A0 My
    problem is that we don&#39;t have those today for the reception side -
    but this (for apps that *need* chunking) is largely a W3 issue.<div cla=
ss=3D""><br>
    <br>
    <br>
    <pre cols=3D"72">--=20
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
  </div></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Mar 2014 05:31:03 -0800
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Subject: Re: [rtcweb] JSEP-06: How to create the second offer where only the port numbers are changed?
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--001a11c3c970c6bb2904f3b3cdb3
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On Mon, Mar 3, 2014 at 2:30 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>  As specified in BUNDLE, before the remote peer has indicated support of
> BUNDLE, all m- lines (including those within to a bundle group) must have
> unique port numbers.
>
> Then, when the remote peer has indicated support of BUNDLE, a second offe=
r
> (called the BAS offer) needs to be sent, in which the same port number is
> assigned to all m- lines within a bundle group.
>
> My question is: how do I create the BAS offer? I don=E2=80=99t want to ch=
ange
> anything =E2=80=93 I just want to get an SDP with the BUNDLE port assigne=
d to all
> m- lines. Is there a flag in createOffer() for that?
>
>  The implementation knows that any offers that are created after the
> BUNDLE answer should have the same port numbers. So, the app could push t=
he
> createOffer button once it receives the answer if it wanted to ensure thi=
s
> happened immediately. Otherwise, there will be an onnegotiationneeded eve=
nt
> once ICE completes, and the offer sent at that time will have the updated
> ports.
>
>
>
> When I call createOffer(), can I be ensure that *only* the port values
> will change?
>
>
>

No. Why would this be necessary?

--001a11c3c970c6bb2904f3b3cdb3
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quo=
te">On Mon, Mar 3, 2014 at 2:30 AM, Christer Holmberg <span dir=3D"ltr">&lt=
;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christ=
er.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,<u></u><u></u></span><=
/p>
<div>
<div>
<div><div><div class=3D"h5">
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">As specified in BUNDLE, before =
the remote peer has indicated support of BUNDLE, all m- lines (including th=
ose within to a bundle group) must have unique port
 numbers.=C2=A0</span><span style=3D"color:#1f497d"><u></u><u></u></span></=
p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Then, when the remote peer has =
indicated support of BUNDLE, a second offer (called the BAS offer) needs to=
 be sent, in which the same port number is assigned
 to all m- lines within a bundle group.</span><span lang=3D"FI" style=3D"co=
lor:#1f497d"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">My question is: how do I create=
 the BAS offer? I don=E2=80=99t want to change anything =E2=80=93 I just wa=
nt to get an SDP with the BUNDLE port assigned to all m- lines. Is there
 a flag in createOffer() for that?</span><span lang=3D"FI" style=3D"color:#=
1f497d"><u></u><u></u></span></p>
</div>
</div>
</blockquote>
</div></div><div><div><div class=3D"h5">
<p class=3D"MsoNormal">The implementation knows that any offers that are cr=
eated after the BUNDLE answer should have the same port numbers. So, the ap=
p could push the createOffer button once it receives the answer if it wante=
d to ensure this happened immediately.
 Otherwise, there will be an onnegotiationneeded event once ICE completes, =
and the offer sent at that time will have the updated ports.<u></u><u></u><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
</div></div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">When I call c=
reateOffer(), can I be ensure that
<b>only</b> the port values will change?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0</span></p><=
/div></div></div></div></div></div></blockquote><div><br></div><div>No. Why=
 would this be necessary?=C2=A0</div>

</div></div></div>

--001a11c3c970c6bb2904f3b3cdb3--


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Date: Mon, 3 Mar 2014 13:32:10 +0000
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Karl Stahl <karl.stahl@intertex.se>, "'Cullen Jennings (fluffy)'" <fluffy@cisco.com>, 'IESG IESG' <iesg@ietf.org>
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Cc: rtcweb@ietf.org, tram@ietf.org
Subject: Re: [rtcweb] [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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Karl,

speaking as RAI AD and TRAM WG chair:

note that our policy is to avoid cross-posting to several mailing lists.
The reason is that doing so may make it more difficult for people who
are subscribed to several lists to follow the discussions. Also,
messages with too many requirements are moderated and need to be
manually approved by the admins of each list.

Therefore, from now on, please choose only *one* mailing list;
preferably one where the discussions you intend to have are within its
scope.

Thanks,

Gonzalo


On 03/03/2014 1:25 PM, Karl Stahl wrote:
> Cullen,
> 
> Illution
> I am checking the IETF archive for links - Memory you know...
> I feel I also have an email from Harald to answer
> 
> 
> 
> 
> You will not see the below
> /Kalle
> Allt väl
> -----Ursprungligt meddelande-----
> Från: Karl Stahl [mailto:karl.stahl@intertex.se] 
> Skickat: den 3 mars 2014 10:22
> Till: 'Cullen Jennings (fluffy)'; 'IESG IESG'
> Kopia: 'Magnus Westerlund'; 'Simon Perreault'; 'Ted Hardie'; 'Gonzalo
> Camarillo'; 'rtcweb@ietf.org'; 'tram@ietf.org'; 'Spencer Dawkins'
> Ämne: SV: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter
> Clarification regardig Milestone 3: TURN server auto-discovery mechanism for
> enterprise and ISPs
> 
> Hi Cullen,
> 
> Memory may be an illusion, still I cannot understand how there can be an
> ever growing archive in the IETF
> 
> /Karl
> 
> Allt väl
> Hi Cullen,
> 
> Not to cause more lost in the threads (I am also) I am addressing both your
> emails in one, and copy separately to the other guys in the first email. 
> 
> Further Inline -->
> 
>>From http://www.ietf.org/mail-archive/web/tram/current/msg00275.html
> Let me also point out draft-deng-tram-isp-turn-00
> http://www.ietf.org/mail-archive/web/tram/current/msg00214.html from China
> Mobile that appeared on this mailing list yesterday. It points out the need
> and willingness from the ISP/NSP side to do something about QoS for WebRTC
> traffic, that they expect to be large and have to bring to their customers
> with best QoE. In fact they and some more (a huge European carrier and the
> cable operators in general - CableLabs) have expressed similar concerns to
> me *"This is exactly something we want to work on as the current way will
> cause severe problem on traffic when rtcweb applications get popular"* is a
> direct quote from one of those.
> 
> So, in answer to Simon Perreault [simon.perreault at viagenie.ca]'s question
> in another thread the 13th:
> a) Is this a real problem that is worth fixing?
> I think we can be confident that the answer is *YES*. Only these three
> ISP/NSP referred to, may represent 50%(?) of the universe's IP accesses! And
> the other will follow these most forward looking carriers. 
> 
> 
> -----Ursprungligt meddelande-----
> Från: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
> Skickat: den 26 februari 2014 01:49
> Till: Karl Stahl
> Kopia: Alan Johnston; rtcweb@ietf.org; tram@ietf.org; Bernard Aboba; David
> Singer; Harald Tveit Alvestrand; Marc Robins; Eric Burger; Mary Barnes;
> Henning Schulzrinne
> Ämne: Re: [rtcweb] [tram] I-D Action:
> draft-thomson-tram-turn-bandwidth-00.txt
> 
> 
>> Karl, 
> 
>> I am totally lost on this thread. Could you start a new tmead that
> summarize what the issues is, what seems to > be the point of debate, and
> what your view is on what we should do. I think that would help make
> progress. 
> 
>> Thank you, 
> 
>> Cullen
> 
> I just did something like that in:
> [tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
> IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff 
> http://www.ietf.org/mail-archive/web/tram/current/msg00331.html 
> Hoping this thread subject amplifies what this SHOULD be about. (An how it
> will lead to the summary, after all diversion.)
> 
> Actually, the outcome of "Interfacing to QoS" will be 
> 
> 
> I challenge anyone that this will be outcome, and will start "Interfacing to
> QoS" with what IP "Interfacing to QoS" that we have the tiny ADSL modem IX78
> that implements such CLEAR INTERFACE and the actual QoS methods, for both
> the Congestion, Default gateway, Surf Pipe 
> Manipulating the RTP extension header etc etc on 6 USD CPU and also include
> the ADSL part where that quality handling is based on heavy 57 bytes
> chopping up
>    at this exact 
> 
> 
> 
> 
> -----Ursprungligt meddelande-----
> Från: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com] 
> Skickat: den 26 februari 2014 02:06
> Till: Karl Stahl; IESG IESG
> Kopia: Magnus Westerlund; Simon Perreault; Ted Hardie; Gonzalo Camarillo;
> rtcweb@ietf.org; tram@ietf.org; Spencer Dawkins
> Ämne: Re: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter
> Clarification regardig Milestone 3: TURN server auto-discovery mechanism for
> enterprise and ISPs
> 
> 
> 
> On Feb 25, 2014, at 7:02 AM, Karl Stahl <karl.stahl@intertex.se> wrote:
> 
>> To the TRAM WG and RTCWEB WG and ADs:
>>  
>> It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed and
> encouraged to route quality demanding WebRTC media into their IP pipes that
> are capable of transporting real-time traffic without quality issues, using
> TURN servers.
>>  
> 
> Reading the charter, the above is *not* at all a clear objective of the WG
> (note I am not the chair of this WG or the responsible AD). 
> 
> That said, I think you have pointed out this charter is abysmally vague - it
> does not say what the WG is not going to do. If I decided to do BGP for
> routing updates over TURN it would be within the scope of this charter. 
> 
> My advice to the responsible AD is recharter this WG before IETF 90 or close
> it. I would be glad to help write a charter that is not an infinite blank
> cheque. 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> tram mailing list
> tram@ietf.org
> https://www.ietf.org/mailman/listinfo/tram
> 


From nobody Mon Mar  3 05:48:45 2014
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To: Justin Uberti <juberti@google.com>
Thread-Topic: JSEP-06: How to create the second offer where only the port numbers are changed?
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Subject: Re: [rtcweb] JSEP-06: How to create the second offer where only the port numbers are changed?
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On 3/3/2014 8:28 AM, Justin Uberti wrote:
> On Mon, Mar 3, 2014 at 3:44 AM, Randell Jesup <randell-ietf@jesup.org 
> <mailto:randell-ietf@jesup.org>> wrote:
>
>     On 3/3/2014 6:18 AM, Justin Uberti wrote:
>>     On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup
>>     <randell-ietf@jesup.org <mailto:randell-ietf@jesup.org>> wrote:
>>>
>>>     So I still think manual chunking seems like a reasonable thing
>>>     to support. I don't quite get the congestion control concerns;
>>>     just because there is a max chunk size doesn't mean the impl
>>>     can't buffer multiple chunks in bufferedAmount; the app could
>>>     let that fill up to a degree to avoid having to poll constantly
>>>     to prevent underrun.
>>
>>         On a slow link this will work if the browser isn't janky.  On
>>         a fast link GC pauses and other things may cause the buffer
>>         to drain out and go idle for significant periods.
>>
>>
>>     Since the amount buffered is up to the app, the app can just
>>     increase the buffer size to provide enough data to cover the next
>>     second or so of transfer.
>
>     How does the app increase the size of the internal SCTP buffers? 
>     (And that buffer is shared by all channels, of course).
>
>
> What I meant is the app can make multiple calls to send() at a time, 
> each with a chunk, which will be buffered in the api layer and 
> increase bufferedAmount.

Aha - that explains "increase the buffer size".  Yes, that will work, 
modulo that the app will have to track the actual bandwidth by either 
monitoring how fast the bufferedAmount drains, or via a JS API to report 
bandwidth ala the bug I opened on JS interaction with congestion 
control.  I still feel this pushes some level of congestion control into 
the JS app (and done like trying to thread needles with oven mitts on - 
ok, mild exaggeration) or the app will simply ignore this issue (likely) 
and get sub-optimal performance, perhaps significantly.  So I think more 
thought would be useful here, especially at the W3/JS level.

If we have the bandwidth reporting, having the app choose a reasonable 
amount of buffering would be much easier/simpler.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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    <div class="moz-cite-prefix">On 3/3/2014 8:28 AM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-3q2TJnVFYLJFJ=mCjdOtOY9-0W1t6=m+_gNC8+Z65b9w@mail.gmail.com"
      type="cite">
      <div dir="ltr">On Mon, Mar 3, 2014 at 3:44 AM, Randell Jesup <span
          dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a>&gt;</span>
        wrote:<br>
        <div class="gmail_extra">
          <div class="gmail_quote">
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div text="#000000" bgcolor="#FFFFFF">
                <div class="">
                  <div>On 3/3/2014 6:18 AM, Justin Uberti wrote:<br>
                  </div>
                  <blockquote type="cite">
                    <div dir="ltr">
                      <div class="gmail_extra">On Sun, Mar 2, 2014 at
                        6:42 PM, Randell Jesup <span dir="ltr">&lt;<a
                            moz-do-not-send="true"
                            href="mailto:randell-ietf@jesup.org"
                            target="_blank">randell-ietf@jesup.org</a>&gt;</span>
                        wrote:<br>
                        <div>
                          <blockquote type="cite">
                            <div dir="ltr">
                              <div><br>
                              </div>
                              <div>So I still think manual chunking
                                seems like a reasonable thing to
                                support. I don't quite get the
                                congestion control concerns; just
                                because there is a max chunk size
                                doesn't mean the impl can't buffer
                                multiple chunks in bufferedAmount; the
                                app could let that fill up to a degree
                                to avoid having to poll constantly to
                                prevent underrun.</div>
                            </div>
                          </blockquote>
                          <br>
                        </div>
                      </div>
                    </div>
                  </blockquote>
                </div>
                <div class="">
                  <blockquote type="cite">
                    <div dir="ltr">
                      <div class="gmail_extra">
                        <div class="gmail_quote">
                          <blockquote class="gmail_quote"
                            style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                            <div text="#000000" bgcolor="#FFFFFF"> On a
                              slow link this will work if the browser
                              isn't janky.&nbsp; On a fast link GC pauses and
                              other things may cause the buffer to drain
                              out and go idle for significant periods.</div>
                          </blockquote>
                          <div><br>
                          </div>
                          <div>Since the amount buffered is up to the
                            app, the app can just increase the buffer
                            size to provide enough data to cover the
                            next second or so of transfer. <br>
                          </div>
                        </div>
                      </div>
                    </div>
                  </blockquote>
                  <br>
                </div>
                How does the app increase the size of the internal SCTP
                buffers?&nbsp; (And that buffer is shared by all channels, of
                course).</div>
            </blockquote>
            <div><br>
            </div>
            <div>What I meant is the app can make multiple calls to
              send() at a time, each with a chunk, which will be
              buffered in the api layer and increase bufferedAmount.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Aha - that explains "increase the buffer size".&nbsp; Yes, that will
    work, modulo that the app will have to track the actual bandwidth by
    either monitoring how fast the bufferedAmount drains, or via a JS
    API to report bandwidth ala the bug I opened on JS interaction with
    congestion control.&nbsp; I still feel this pushes some level of
    congestion control into the JS app (and done like trying to thread
    needles with oven mitts on - ok, mild exaggeration) or the app will
    simply ignore this issue (likely) and get sub-optimal performance,
    perhaps significantly.&nbsp; So I think more thought would be useful
    here, especially at the W3/JS level.<br>
    <br>
    If we have the bandwidth reporting, having the app choose a
    reasonable amount of buffering would be much easier/simpler.<br>
    <pre class="moz-signature" cols="72">-- 
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
  </body>
</html>

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On 3 March 2014 03:44, Randell Jesup <randell-ietf@jesup.org> wrote:
> Hopefully so, though I hate forcing apps to include a bunch of code for
> backwards compat for a year or two (and many won't, or won't test it, etc).
> That hurts the feature.  And today no one has ndata.

Since WebRTC is behind a flag, I don't find this particularly
convincing.  We can implement ndata.

I think that we could even remove our temporary length restriction.


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--001a1136b114beee7e04f3b46297
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But why is it a problem if other things change (e.g. a=3Dremote-candidates =
is
set)?


On Mon, Mar 3, 2014 at 5:48 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>      As specified in BUNDLE, before the remote peer has indicated support
> of BUNDLE, all m- lines (including those within to a bundle group) must
> have unique port numbers.
>
> Then, when the remote peer has indicated support of BUNDLE, a second offe=
r
> (called the BAS offer) needs to be sent, in which the same port number is
> assigned to all m- lines within a bundle group.
>
> My question is: how do I create the BAS offer? I don=E2=80=99t want to ch=
ange
> anything =E2=80=93 I just want to get an SDP with the BUNDLE port assigne=
d to all
> m- lines. Is there a flag in createOffer() for that?
>
>    The implementation knows that any offers that are created after the
> BUNDLE answer should have the same port numbers. So, the app could push t=
he
> createOffer button once it receives the answer if it wanted to ensure thi=
s
> happened immediately. Otherwise, there will be an onnegotiationneeded eve=
nt
> once ICE completes, and the offer sent at that time will have the updated
> ports.
>
> When I call createOffer(), can I be ensure that *only* the port values
> will change?
>
>
>
> No. Why would this be necessary?
>
>
>
> Because the only thing I want to do is to send a new offer, with the same
> port number in each m- line. I don=E2=80=99t want anything else to change=
.
>
>
>
> Regards,
>
>
>
> Christer
>

--001a1136b114beee7e04f3b46297
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<div dir=3D"ltr">But why is it a problem if other things change (e.g. a=3Dr=
emote-candidates is set)?</div><div class=3D"gmail_extra"><br><br><div clas=
s=3D"gmail_quote">On Mon, Mar 3, 2014 at 5:48 AM, Christer Holmberg <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"=
_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">Hi,<u></u><u></u></span></p>
<div>
<div>
<div><div><div class=3D"h5">
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">As specified in BUNDLE, before =
the remote peer has indicated support of BUNDLE, all m- lines (including th=
ose within to a bundle group) must have unique port
 numbers.=C2=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Then, when the remote peer has =
indicated support of BUNDLE, a second offer (called the BAS offer) needs to=
 be sent, in which the same port number is assigned
 to all m- lines within a bundle group.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">My question is: how do I create=
 the BAS offer? I don=E2=80=99t want to change anything =E2=80=93 I just wa=
nt to get an SDP with the BUNDLE port assigned to all m- lines. Is there
 a flag in createOffer() for that?</span><u></u><u></u></p>
</div>
</div>
</blockquote>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">The implementation knows that any offers that are cr=
eated after the BUNDLE answer should have the same port numbers. So, the ap=
p could push the createOffer button once it receives
 the answer if it wanted to ensure this happened immediately. Otherwise, th=
ere will be an onnegotiationneeded event once ICE completes, and the offer =
sent at that time will have the updated ports.<span style=3D"color:#1f497d"=
><u></u><u></u></span></p>


</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">When I call createOffer()=
, can I be ensure that
<b>only</b> the port values will change?</span><span style=3D"color:#1f497d=
"><u></u><u></u></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div><div><div><div class=3D"h5">
<p class=3D"MsoNormal">No. Why would this be necessary?=C2=A0<u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
</div></div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Because the only =
thing I want to do is to send a new offer, with the same port number in eac=
h m- line. I don=E2=80=99t want anything else to change.<u></u><u></u></spa=
n></p>


<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red"><u></u>=C2=A0<u></u></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">Regards,<span class=3D"HOEnZb=
"><font color=3D"#888888"><u></u><u></u></font></span></span></p><span clas=
s=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red"><u></u>=C2=A0<u></u></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">Christer<u></u><u></u></span>=
</p>
</font></span></div>
</div>
</div>
</div>
</div>
</div>

</blockquote></div><br></div>

--001a1136b114beee7e04f3b46297--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: JSEP-06: How to create the second offer where only the port numbers are changed?
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Subject: Re: [rtcweb] JSEP-06: How to create the second offer where only the port numbers are changed?
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References: <31B9253D-E826-4D07-A8A1-1B062B50F163@lurchi.franken.de> <530D9CC5.5080508@ericsson.com> <5312FBBC.5080006@jesup.org> <CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com> <5313EC0D.9030808@jesup.org> <CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com> <53146B2C.8080008@jesup.org> <CAOJ7v-3q2TJnVFYLJFJ=mCjdOtOY9-0W1t6=m+_gNC8+Z65b9w@mail.gmail.com> <531488F0.7090300@jesup.org>
From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Mar 2014 06:37:13 -0800
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Subject: Re: [rtcweb] Open data channel issues
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On Mon, Mar 3, 2014 at 5:51 AM, Randell Jesup <randell-ietf@jesup.org>wrote:

>  On 3/3/2014 8:28 AM, Justin Uberti wrote:
>
> On Mon, Mar 3, 2014 at 3:44 AM, Randell Jesup <randell-ietf@jesup.org>wrote:
>
>>  On 3/3/2014 6:18 AM, Justin Uberti wrote:
>>
>>  On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup <randell-ietf@jesup.org>wrote:
>>
>>
>>  So I still think manual chunking seems like a reasonable thing to
>> support. I don't quite get the congestion control concerns; just because
>> there is a max chunk size doesn't mean the impl can't buffer multiple
>> chunks in bufferedAmount; the app could let that fill up to a degree to
>> avoid having to poll constantly to prevent underrun.
>>
>>
>>       On a slow link this will work if the browser isn't janky.  On a
>>> fast link GC pauses and other things may cause the buffer to drain out and
>>> go idle for significant periods.
>>>
>>
>>  Since the amount buffered is up to the app, the app can just increase
>> the buffer size to provide enough data to cover the next second or so of
>> transfer.
>>
>>
>>  How does the app increase the size of the internal SCTP buffers?  (And
>> that buffer is shared by all channels, of course).
>>
>
>  What I meant is the app can make multiple calls to send() at a time,
> each with a chunk, which will be buffered in the api layer and increase
> bufferedAmount.
>
>
> Aha - that explains "increase the buffer size".  Yes, that will work,
> modulo that the app will have to track the actual bandwidth by either
> monitoring how fast the bufferedAmount drains, or via a JS API to report
> bandwidth ala the bug I opened on JS interaction with congestion control.
> I still feel this pushes some level of congestion control into the JS app
> (and done like trying to thread needles with oven mitts on - ok, mild
> exaggeration) or the app will simply ignore this issue (likely) and get
> sub-optimal performance, perhaps significantly.  So I think more thought
> would be useful here, especially at the W3/JS level.
>
> If we have the bandwidth reporting, having the app choose a reasonable
> amount of buffering would be much easier/simpler.
>

In the simplest case, you could just have a medium-sized buffer of size
(max expected transfer rate * worst-case poll interval); 10 MB would be
plenty for almost all cases.

If you wanted to be smarter, you could start with 1MB buffer, poll every
100 ms, and then adjust the buffer size to be at least ((data sent since
last poll / time since last poll) * worst-case poll interval), where
worst-case poll interval, accounting for GC, is somewhere around 1 second.

I don't think this pushes CC onto apps, it just pushes spooling onto them,
which is the consequence of not having streams yet.


> --
> Randell Jesup -- rjesup a t mozilla d o t com
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Mar 3, 2014 at 5:51 AM, Randell Jesup <span dir=3D"ltr">&lt=
;<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@j=
esup.org</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D"">
    <div>On 3/3/2014 8:28 AM, Justin Uberti
      wrote:<br>
    </div>
    </div><blockquote type=3D"cite">
      <div dir=3D"ltr"><div class=3D"">On Mon, Mar 3, 2014 at 3:44 AM, Rand=
ell Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf@jesup.org" t=
arget=3D"_blank">randell-ietf@jesup.org</a>&gt;</span>
        wrote:<br>
        </div><div class=3D"gmail_extra">
          <div class=3D"gmail_quote">
            <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">
              <div text=3D"#000000" bgcolor=3D"#FFFFFF">
                <div><div class=3D"">
                  <div>On 3/3/2014 6:18 AM, Justin Uberti wrote:<br>
                  </div>
                  </div><blockquote type=3D"cite">
                    <div dir=3D"ltr">
                      <div class=3D"gmail_extra"><div class=3D"">On Sun, Ma=
r 2, 2014 at
                        6:42 PM, Randell Jesup <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@jesup.or=
g</a>&gt;</span>
                        wrote:<br>
                        </div><div class=3D""><div>
                          <blockquote type=3D"cite">
                            <div dir=3D"ltr">
                              <div><br>
                              </div>
                              <div>So I still think manual chunking
                                seems like a reasonable thing to
                                support. I don&#39;t quite get the
                                congestion control concerns; just
                                because there is a max chunk size
                                doesn&#39;t mean the impl can&#39;t buffer
                                multiple chunks in bufferedAmount; the
                                app could let that fill up to a degree
                                to avoid having to poll constantly to
                                prevent underrun.</div>
                            </div>
                          </blockquote>
                          <br>
                        </div>
                      </div></div>
                    </div>
                  </blockquote>
                </div><div class=3D"">
                <div>
                  <blockquote type=3D"cite">
                    <div dir=3D"ltr">
                      <div class=3D"gmail_extra">
                        <div class=3D"gmail_quote">
                          <blockquote class=3D"gmail_quote" style=3D"margin=
:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204)=
;border-left-style:solid;padding-left:1ex">
                            <div text=3D"#000000" bgcolor=3D"#FFFFFF"> On a
                              slow link this will work if the browser
                              isn&#39;t janky.=C2=A0 On a fast link GC paus=
es and
                              other things may cause the buffer to drain
                              out and go idle for significant periods.</div=
>
                          </blockquote>
                          <div><br>
                          </div>
                          <div>Since the amount buffered is up to the
                            app, the app can just increase the buffer
                            size to provide enough data to cover the
                            next second or so of transfer. <br>
                          </div>
                        </div>
                      </div>
                    </div>
                  </blockquote>
                  <br>
                </div>
                How does the app increase the size of the internal SCTP
                buffers?=C2=A0 (And that buffer is shared by all channels, =
of
                course).</div></div>
            </blockquote><div class=3D"">
            <div><br>
            </div>
            <div>What I meant is the app can make multiple calls to
              send() at a time, each with a chunk, which will be
              buffered in the api layer and increase bufferedAmount.</div>
          </div></div>
        </div>
      </div>
    </blockquote>
    <br>
    Aha - that explains &quot;increase the buffer size&quot;.=C2=A0 Yes, th=
at will
    work, modulo that the app will have to track the actual bandwidth by
    either monitoring how fast the bufferedAmount drains, or via a JS
    API to report bandwidth ala the bug I opened on JS interaction with
    congestion control.=C2=A0 I still feel this pushes some level of
    congestion control into the JS app (and done like trying to thread
    needles with oven mitts on - ok, mild exaggeration) or the app will
    simply ignore this issue (likely) and get sub-optimal performance,
    perhaps significantly.=C2=A0 So I think more thought would be useful
    here, especially at the W3/JS level.<br>
    <br>
    If we have the bandwidth reporting, having the app choose a
    reasonable amount of buffering would be much easier/simpler.</div></blo=
ckquote><div><br></div><div>In the simplest case, you could just have a med=
ium-sized buffer of size (max expected transfer rate * worst-case poll inte=
rval); 10 MB would be plenty for almost all cases.</div>

<div><br></div><div>If you wanted to be smarter, you could start with 1MB b=
uffer, poll every 100 ms, and then adjust the buffer size to be at least ((=
data sent since last poll / time since last poll) * worst-case poll interva=
l), where worst-case poll interval, accounting for GC, is somewhere around =
1 second.=C2=A0</div>

<div><br></div><div>I don&#39;t think this pushes CC onto apps, it just pus=
hes spooling onto them, which is the consequence of not having streams yet.=
</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D""><br>
    <pre cols=3D"72">--=20
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
  </div></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Mon, 3 Mar 2014 06:39:02 -0800
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Subject: Re: [rtcweb] JSEP-06: How to create the second offer where only the port numbers are changed?
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Agreed.


On Mon, Mar 3, 2014 at 6:35 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Fair enough, maybe some things can change. And, BUNDLE does not forbid
> changing values, but it does say that the offerer should be careful when
> doing so.
>
>
>
> Section 6.4.3:
>
>
>
> =E2=80=9CThe Offerer MAY modify any SDP parameter in a BAS Offer.
>
>
>
>    NOTE: It is important that the BAS Offer gets accepted by the
>
>    Answerer, so the Offerer needs to consider the necessity to modify
>
>    SDP parameters that could get the Answerer to reject the BAS Offer.
>
>    Removing "m=3D" lines, or reducing the number of codecs, in the BAS
>
>    Offer used for the is considered to have a low risk of being
>
>    rejected.=E2=80=9D
>
>
>
> So, I guess what should be avoided is adding of new m- lines, new codecs,
> or anything which may trigger the answer to reject the offer.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
> *From:* Justin Uberti [mailto:juberti@google.com]
> *Sent:* 03 March 2014 16:13
> *To:* Christer Holmberg
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: JSEP-06: How to create the second offer where only the
> port numbers are changed?
>
>
>
> But why is it a problem if other things change (e.g. a=3Dremote-candidate=
s
> is set)?
>
>
>
> On Mon, Mar 3, 2014 at 5:48 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>  Hi,
>
>      As specified in BUNDLE, before the remote peer has indicated support
> of BUNDLE, all m- lines (including those within to a bundle group) must
> have unique port numbers.
>
> Then, when the remote peer has indicated support of BUNDLE, a second offe=
r
> (called the BAS offer) needs to be sent, in which the same port number is
> assigned to all m- lines within a bundle group.
>
> My question is: how do I create the BAS offer? I don=E2=80=99t want to ch=
ange
> anything =E2=80=93 I just want to get an SDP with the BUNDLE port assigne=
d to all
> m- lines. Is there a flag in createOffer() for that?
>
>    The implementation knows that any offers that are created after the
> BUNDLE answer should have the same port numbers. So, the app could push t=
he
> createOffer button once it receives the answer if it wanted to ensure thi=
s
> happened immediately. Otherwise, there will be an onnegotiationneeded eve=
nt
> once ICE completes, and the offer sent at that time will have the updated
> ports.
>
> When I call createOffer(), can I be ensure that *only* the port values
> will change?
>
>
>
> No. Why would this be necessary?
>
>
>
> Because the only thing I want to do is to send a new offer, with the same
> port number in each m- line. I don=E2=80=99t want anything else to change=
.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>

--001a1136776ae3eb5d04f3b4c02b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Agreed.</div><div class=3D"gmail_extra"><br><br><div class=
=3D"gmail_quote">On Mon, Mar 3, 2014 at 6:35 AM, Christer Holmberg <span di=
r=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_=
blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Fair enough, maybe some t=
hings can change. And, BUNDLE does not forbid changing values, but it does =
say that the offerer should be
 careful when doing so.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Section 6.4.3:<u></u><u><=
/u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=E2=80=9CThe Offerer MAY modify any SDP parameter in a BAS=
 Offer.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0 NOTE: It is important that the BAS Offer gets=
 accepted by the<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0 Answerer, so the Offerer needs to consider th=
e necessity to modify<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0 SDP parameters that could get the Answerer to=
 reject the BAS Offer.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0 Removing &quot;m=3D&quot; lines, or reducing =
the number of codecs, in the BAS<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0 Offer used for the is considered to have a lo=
w risk of being<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=C2=A0=C2=A0 rejected.=E2=80=9D<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">So, I guess what should b=
e avoided is adding of new m- lines, new codecs, or anything which may trig=
ger the answer to reject the
 offer. <u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><a name=3D"144885f0e50fa11c__MailEndCompose"><span s=
tyle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&q=
uot;;color:#1f497d"><u></u>=C2=A0<u></u></span></a></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span=
 lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;"> Justin Uberti [mailto:<a href=3D"mailto:juberti@goo=
gle.com" target=3D"_blank">juberti@google.com</a>]
<br>
<b>Sent:</b> 03 March 2014 16:13<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: JSEP-06: How to create the second offer where only the =
port numbers are changed?<u></u><u></u></span></p><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">But why is it a problem if other things change (e.g.=
 a=3Dremote-candidates is set)?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<div>
<p class=3D"MsoNormal">On Mon, Mar 3, 2014 at 5:48 AM, Christer Holmberg &l=
t;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chris=
ter.holmberg@ericsson.com</a>&gt; wrote:<u></u><u></u></p>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">Hi,</span><u></u><u></u></p>
<div>
<div>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US">As specified in BUNDLE, before =
the remote peer has indicated support of BUNDLE, all m- lines (including th=
ose within to a bundle group) must have unique port
 numbers.=C2=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Then, when the remote peer has =
indicated support of BUNDLE, a second offer (called the BAS offer) needs to=
 be sent, in which the same port number is assigned
 to all m- lines within a bundle group.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">My question is: how do I create=
 the BAS offer? I don=E2=80=99t want to change anything =E2=80=93 I just wa=
nt to get an SDP with the BUNDLE port assigned to all m- lines. Is there
 a flag in createOffer() for that?</span><u></u><u></u></p>
</div>
</div>
</blockquote>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">The implementation knows that any offers that are cr=
eated after the BUNDLE answer should have the same port numbers. So, the ap=
p could push the createOffer button once it receives
 the answer if it wanted to ensure this happened immediately. Otherwise, th=
ere will be an onnegotiationneeded event once ICE completes, and the offer =
sent at that time will have the updated ports.<u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">When I call createOffer()=
, can I be ensure that
<b>only</b> the port values will change?</span><u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">No. Why would this be necessary?=C2=A0<u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">Because the only thing I want=
 to do is to send a new offer, with the same port number in each m- line.
 I don=E2=80=99t want anything else to change.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0</span><u></u><u></u></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">Regards,</span><u></u><u></u>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0</span><span style=3D"c=
olor:#888888"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:red">Christer</span><span style=3D=
"color:#888888"><u></u><u></u></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div></div></div>
</div>

</blockquote></div><br></div>

--001a1136776ae3eb5d04f3b4c02b--


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Date: Mon, 3 Mar 2014 06:41:08 -0800
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From: Martin Thomson <martin.thomson@gmail.com>
To: Randell Jesup <randell-ietf@jesup.org>
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I should have said prefix.

On 3 March 2014 05:53, Martin Thomson <martin.thomson@gmail.com> wrote:
> On 3 March 2014 03:44, Randell Jesup <randell-ietf@jesup.org> wrote:
>> Hopefully so, though I hate forcing apps to include a bunch of code for
>> backwards compat for a year or two (and many won't, or won't test it, etc).
>> That hurts the feature.  And today no one has ndata.
>
> Since WebRTC is behind a flag, I don't find this particularly
> convincing.  We can implement ndata.
>
> I think that we could even remove our temporary length restriction.


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On 26 Feb 2014, at 22:21, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>> * Section 6.5
>>>>=20
>>>> This section contains:
>>>=20
>>>    Data channels can be opened by using internal or external
>>>    negotiation.  The details are out of scope of this document.
>>>=20
>>>    A simple protocol for internal negotiation is specified in
>>>    [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>>=20
>>>> But internal and external negotiation are not defined in this =
document.
>>>>=20
>>>=20
>>>> I *thought* that internal negotiation was by definition negotiation=20=

>>> by use of  > rtcweb-data-protocol.=20
>>> (draft-ejzak-mmusic-data-channel-sdpneg-00
>>> thinks so too,
>>>> but calls it "in-band negotiation".) There should be a good=20
>>> definition of these  > terms, or reference to one. And more =
discussion=20
>>> if there can be other kinds of  > internal negotiation. (If so, how=20=

>>> would one be chosen?)
>>>=20
>>> Well, the text above says that ietf-rtcweb-data-protocol specifies =
an=20
>>> internal negotiation protocol, so it's not that far off.  The split=20=

>>> does allow someone to use an alternative negotiation protocol=20
>>> (internal or external).
>>>=20
>>> How about:
>>>=20
>>>    Data channels can be opened by using negotiation within the SCTP=20=

>>> association or external
>>>    negotiation.  External negotiation is defined as any method which=20=

>>> results in an agreement
>>>    as to the parameters of a channel and the creation thereof.
>>>    The details are out of scope of this document.
>>>=20
>>>    A simple protocol for negotiation within the SCTP association is=20=

>>> specified in
>>>    [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>>=20
>>> Perhaps someone can wordsmith "external negotiation" better.
>>=20
>> ISTM the issue is that Internal vs. External isn't the key =
distinction.=20
>> What is key is that there is one method that MUST be *supported* but =
need not be the only one >used. Other mechanisms may be used. It doesn't =
really matter whether they are conducted >internally (over the SCTP =
association) or externally (anything else).
>=20
> I think the MUST-be-supported-method is a separate question.
>=20
>> E.g., I could use DCEP to establish one channel, and then use my own =
private protocol over that to >negotiate other channels. Would that be =
internal or external negotiation? Does it matter?
>=20
> I guess they would both be in-band methods. Maybe using in-band and =
out-of-band terminology would be more clear - and more aligned with =
existing terminology.
OK, changed to in-band and out-of-band.

Best regards
Michael
>=20
> Regards,
>=20
> Christer
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Mon Mar  3 07:06:34 2014
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: <rtcweb@ietf.org>
References: <31B9253D-E826-4D07-A8A1-1B062B50F163@lurchi.franken.de> <530D9CC5.5080508@ericsson.com> <5312FBBC.5080006@jesup.org> <CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com> <5313EC0D.9030808@jesup.org> <CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com> <53146B2C.8080008@jesup.org> <CAOJ7v-3q2TJnVFYLJFJ=mCjdOtOY9-0W1t6=m+_gNC8+Z65b9w@mail.gmail.com> <531488F0.7090300@jesup.org> <CAOJ7v-15EZjPSjAPotCdnZEu4kgeLzBXJdFnYaccaN1ViBvyjQ@mail.gmail.com>
In-Reply-To: <CAOJ7v-15EZjPSjAPotCdnZEu4kgeLzBXJdFnYaccaN1ViBvyjQ@mail.gmail.com>
Date: Mon, 3 Mar 2014 16:06:22 +0100
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Subject: Re: [rtcweb] Open data channel issues
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=20

=20

Fr=C3=A5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=C3=B6r Justin =
Uberti
Skickat: den 3 mars 2014 15:37
Till: Randell Jesup
Kopia: rtcweb@ietf.org
=C3=84mne: Re: [rtcweb] Open data channel issues

=20

=20

=20

On Mon, Mar 3, 2014 at 5:51 AM, Randell Jesup <randell-ietf@jesup.org> =
wrote:

On 3/3/2014 8:28 AM, Justin Uberti wrote:

On Mon, Mar 3, 2014 at 3:44 AM, Randell Jesup <randell-ietf@jesup.org> =
wrote:

On 3/3/2014 6:18 AM, Justin Uberti wrote:

On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup <randell-ietf@jesup.org> =
wrote:

=20

So I still think manual chunking seems like a reasonable thing to =
support. I don't quite get the congestion control concerns; just because =
there is a max chunk size doesn't mean the impl can't buffer multiple =
chunks in bufferedAmount; the app could let that fill up to a degree to =
avoid having to poll constantly to prevent underrun.

=20

On a slow link this will work if the browser isn't janky.  On a fast =
link GC pauses and other things may cause the buffer to drain out and go =
idle for significant periods.

=20

Since the amount buffered is up to the app, the app can just increase =
the buffer size to provide enough data to cover the next second or so of =
transfer.=20

=20

How does the app increase the size of the internal SCTP buffers?  (And =
that buffer is shared by all channels, of course).

=20

What I meant is the app can make multiple calls to send() at a time, =
each with a chunk, which will be buffered in the api layer and increase =
bufferedAmount.


Aha - that explains "increase the buffer size".  Yes, that will work, =
modulo that the app will have to track the actual bandwidth by either =
monitoring how fast the bufferedAmount drains, or via a JS API to report =
bandwidth ala the bug I opened on JS interaction with congestion =
control.  I still feel this pushes some level of congestion control into =
the JS app (and done like trying to thread needles with oven mitts on - =
ok, mild exaggeration) or the app will simply ignore this issue (likely) =
and get sub-optimal performance, perhaps significantly.  So I think more =
thought would be useful here, especially at the W3/JS level.

If we have the bandwidth reporting, having the app choose a reasonable =
amount of buffering would be much easier/simpler.

=20

In the simplest case, you could just have a medium-sized buffer of size =
(max expected transfer rate * worst-case poll interval); 10 MB would be =
plenty for almost all cases.

=20

If you wanted to be smarter, you could start with 1MB buffer, poll every =
100 ms, and then adjust the buffer size to be at least ((data sent since =
last poll / time since last poll) * worst-case poll interval), where =
worst-case poll interval, accounting for GC, is somewhere around 1 =
second.=20

=20

I don't think this pushes CC onto apps, it just pushes spooling onto =
them, which is the consequence of not having streams yet.

=20





--=20
Randell Jesup -- rjesup a t mozilla d o t com


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

=20


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:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div style=3D'border:none;border-top:solid =
#B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=C3=A5n:</=
span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb =
[mailto:rtcweb-bounces@ietf.org] <b>F=C3=B6r </b>Justin =
Uberti<br><b>Skickat:</b> den 3 mars 2014 15:37<br><b>Till:</b> Randell =
Jesup<br><b>Kopia:</b> rtcweb@ietf.org<br><b>=C3=84mne:</b> Re: [rtcweb] =
Open data channel issues<o:p></o:p></span></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Mon, Mar 3, 2014 at 5:51 AM, Randell Jesup &lt;<a =
href=3D"mailto:randell-ietf@jesup.org" =
target=3D"_blank">randell-ietf@jesup.org</a>&gt; =
wrote:<o:p></o:p></p><div><div><div><p class=3DMsoNormal>On 3/3/2014 =
8:28 AM, Justin Uberti wrote:<o:p></o:p></p></div></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p =
class=3DMsoNormal>On Mon, Mar 3, 2014 at 3:44 AM, Randell Jesup &lt;<a =
href=3D"mailto:randell-ietf@jesup.org" =
target=3D"_blank">randell-ietf@jesup.org</a>&gt; =
wrote:<o:p></o:p></p></div><div><div><blockquote =
style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm =
6.0pt;margin-left:4.8pt;margin-right:0cm'><div><div><div><div><p =
class=3DMsoNormal>On 3/3/2014 6:18 AM, Justin Uberti =
wrote:<o:p></o:p></p></div></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><div><p =
class=3DMsoNormal>On Sun, Mar 2, 2014 at 6:42 PM, Randell Jesup &lt;<a =
href=3D"mailto:randell-ietf@jesup.org" =
target=3D"_blank">randell-ietf@jesup.org</a>&gt; =
wrote:<o:p></o:p></p></div><div><div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>So I still think manual chunking seems like a =
reasonable thing to support. I don't quite get the congestion control =
concerns; just because there is a max chunk size doesn't mean the impl =
can't buffer multiple chunks in bufferedAmount; the app could let that =
fill up to a degree to avoid having to poll constantly to prevent =
underrun.<o:p></o:p></p></div></div></blockquote><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></div></blockquo=
te></div><div><div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><div><blockquote=
 style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm =
0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><div><p =
class=3DMsoNormal>On a slow link this will work if the browser isn't =
janky.&nbsp; On a fast link GC pauses and other things may cause the =
buffer to drain out and go idle for significant =
periods.<o:p></o:p></p></div></blockquote><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Since the amount buffered is up to the app, the app =
can just increase the buffer size to provide enough data to cover the =
next second or so of transfer. =
<o:p></o:p></p></div></div></div></div></blockquote><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>How =
does the app increase the size of the internal SCTP buffers?&nbsp; (And =
that buffer is shared by all channels, of =
course).<o:p></o:p></p></div></div></blockquote><div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>What I meant is the app can make multiple calls to =
send() at a time, each with a chunk, which will be buffered in the api =
layer and increase =
bufferedAmount.<o:p></o:p></p></div></div></div></div></div></blockquote>=
<p class=3DMsoNormal><br>Aha - that explains &quot;increase the buffer =
size&quot;.&nbsp; Yes, that will work, modulo that the app will have to =
track the actual bandwidth by either monitoring how fast the =
bufferedAmount drains, or via a JS API to report bandwidth ala the bug I =
opened on JS interaction with congestion control.&nbsp; I still feel =
this pushes some level of congestion control into the JS app (and done =
like trying to thread needles with oven mitts on - ok, mild =
exaggeration) or the app will simply ignore this issue (likely) and get =
sub-optimal performance, perhaps significantly.&nbsp; So I think more =
thought would be useful here, especially at the W3/JS level.<br><br>If =
we have the bandwidth reporting, having the app choose a reasonable =
amount of buffering would be much =
easier/simpler.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>In the simplest case, you could just have a =
medium-sized buffer of size (max expected transfer rate * worst-case =
poll interval); 10 MB would be plenty for almost all =
cases.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>If you wanted to be smarter, you could start with 1MB =
buffer, poll every 100 ms, and then adjust the buffer size to be at =
least ((data sent since last poll / time since last poll) * worst-case =
poll interval), where worst-case poll interval, accounting for GC, is =
somewhere around 1 second.&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
don't think this pushes CC onto apps, it just pushes spooling onto them, =
which is the consequence of not having streams =
yet.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><blockquote =
style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm =
6.0pt;margin-left:4.8pt;margin-right:0cm'><div><div><p =
class=3DMsoNormal><br><br><o:p></o:p></p><pre>-- =
<o:p></o:p></pre><pre>Randell Jesup -- rjesup a t mozilla d o t =
com<o:p></o:p></pre></div></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><br>______________________________________=
_________<br>rtcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></=
o:p></p></blockquote></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
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From nobody Mon Mar  3 07:25:03 2014
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-data-protocol-03
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On 25 Feb 2014, at 17:49, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Inline
>=20
> On 2/24/14 5:12 AM, Michael Tuexen wrote:
>> On Feb 23, 2014, at 4:07 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> =
wrote:
>>=20
>>> A few comments on this draft:
>>>=20
>>> * Section 5.1:
>>>=20
>>>      DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel =
provides
>>>         a partial reliable in-order bi-directional Communication
>>>         channel.  User messages might not be transmitted or
>>>         retransmitted after a specified life-time given in milli-
>>>         seconds in the Reliability Parameter.  This life-time starts
>>>         when providing the user message to the Javascript engine.
>>>=20
>>> The last sentence above is troubling. This protocol won't always be =
used via Javascript. Can we please have a definition that
>> OK. What about replacing the last sentence with
>>=20
>> This life-time starts when providing the user
>> message to the protocol stack.
>=20
> *which* protocol stack?
>=20
> This is part of the problem. As the documents are currently =
structured, there is a "data channel establishment protocol", but there =
is no "data channel protocol".
>=20
> But in reality there *is* a (thin) data channel protocol. It just =
isn't described explicitly. In a browser it is perhaps just the code =
behind the data channel API. But in other environments it may need to be =
more explicit. It has some obligations. Presumably enforcing this =
"lifetime" parameter is one of them. Enforcing the use of sequenced =
messages between a DCEP open and receipt of ACK is another.
>=20
>>> doesn't depend on that? Is the timing being done by SCTP, or are you =
assuming that a data channel layer on top of SCTP is doing this? Is it =
specific to SCEP, or is it still applicable when the channel has been =
established via external negotiation?
>> It is actually the time when the message is provided to the SCTP =
stack, but I think
>> there might be some code running between the SCTP code and the user =
code (let it be
>> JS or anything else). I'm assuming that there is no substantial =
buffering happening
>> there.
>>>=20
>>> Does use of this option imply that retransmission continues until =
this time limit is reached? Or might it stop after some implementation =
defined number of retransmissions?
>> The only way the retransmissions don't happen until the time limit is =
reached, is that
>> SCTP decides that the association is broken because of excessive =
retransmissions.
>>>=20
>>> The description of the reliability parameter says:
>>>=20
>>>      This field is ignored if a reliable channel is used.
>>>=20
>>> IMO folk wisdom says that some specific value (e.g., 0) should be =
prescribed to be used in such cases. That makes things more =
deterministic and provides more flexibility in extending the protocol in =
the future.
>> What about:
>>=20
>> For reliable channels this field MUST be set to 0 on the sending side
>> and MUST be ignored no the receiving side.
>=20
> WFM
>=20
>>> The name "Protocol" (and "Protocol Length") here is troublesome, =
because it is ambiguous with other sorts of uses. (See my comment about =
this point earlier today wrt draft-ietf-rtcweb-data-channel-07.) Others =
(including draft-ejzak-mmusic-data-channel-sdpneg-00 and websockets) =
call this "subprotocol". Using that would make it a little less =
confusing.
>> In the W3C specification it is also called protocol, but the text =
talks about sub-protocol. Maybe we should do
>> the same here.
>=20
> I'm not following the W3C part.
>=20
> I'm just interested in getting consistency, and avoiding confusion.
> Right now there is a lot of confusion around the use of "protocol".
>=20
>>> Also, ISTM that all of these things are applicable to externally =
negotiated channels, and so ought to be defined by =
draft-ietf-rtcweb-data-channel. (But of course their use in the =
DATA_CHANNEL_OPEN message belongs here.)
>> Reliability is discussed there, but I agree, we need to discuss label =
and protocol there, too.
>=20
> Good!
>=20
>>> * Section 8.4:
>>>=20
>>> ISTM there ought to be a template of the minimum information that =
the specification for a registered (sub)protocol MUST (SHOULD?) include. =
E.g.,
>>>=20
>>> - what Channel Type(s) are valid for this (sub)protocol
>> I assumed all...
>=20
> ISTM that in *most* cases a if a (sub)protocol is designed for use =
with a reliable channel it won't work correctly with an unreliable =
channel.
>=20
> And while things designed to work with an unreliable channel will =
probably continue to work with a reliable channel, they might not have =
the intended properties.
>=20
> That's why its good to have some guidelines about what the =
specification must contain. It prevents people providing something =
sloppy and incomplete that isn't sufficient to use the (sub)protocol.
>=20
>>> - A contact for more information about this protocol
>> You need to provide a reference for the protocol. Isn't that enough?
>=20
> Depends on the form of the reference. Is the reference required to be =
publicly available? Stable?
>=20
> Note the definition of FCFS policy in 5226:
>=20
>      First Come First Served - Assignments are made to anyone on a
>            first come, first served basis.  There is no substantive
>            review of the request, other than to ensure that it is
>            well-formed and doesn't duplicate an existing assignment.
>            However, requests must include a minimal amount of clerical
>            information, such as a point of contact (including an email
>            address) and a brief description of how the value will be
>            used.  Additional information specific to the type of value
>            requested may also need to be provided, as defined by the
>            namespace.  For numbers, the exact value is generally
>            assigned by IANA; with names, specific text strings can
>            usually be requested.
>=20
> Having a sufficient definition of what must be in the registry, and =
what must be in a reference from the registry is *more* important for =
FCFS than it is for some of the more restrictive policies. The policies =
that get careful review can perhaps depend on the reviewers stopping =
something that is incomplete.
>=20
> IMO the registry ought to provide, directly or indirectly, enough =
information to implement/use the (sub)protocol.
I'll bring this up in the RTCWeb meeting.

Best regards
Michael
>=20
> 	Thanks,
> 	Paul
>=20
>> Best regards
>> Michael
>>>=20
>>> 	Thanks,
>>> 	Paul
>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>=20
>>=20
>=20
>=20


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On 3/3/2014 8:53 AM, Martin Thomson wrote:
> On 3 March 2014 03:44, Randell Jesup <randell-ietf@jesup.org> wrote:
>> Hopefully so, though I hate forcing apps to include a bunch of code for
>> backwards compat for a year or two (and many won't, or won't test it, etc).
>> That hurts the feature.  And today no one has ndata.
> Since WebRTC is behind a flag, I don't find this particularly
> convincing.  We can implement ndata.

I don't understand the "behind a flag" comment; an app using 
DataChannels for anything variable-length (unless with low bounds) must 
care until all old implementations are gone.  Since we already have 
DataChannel impls in  ESR releases, there's no avoiding this if we rely 
on that (even worse, since this undefined property doesn't exist yet 
(let alone in ESR), the app has to handle that case as well).  Or add 
browser and version sniffing to all the apps...

> I think that we could even remove our temporary length restriction.

Sure, we can.  This will reduce or eliminate source changes for apps 
that don't care about interleaving, which is a plus.  It will cause 
total stall of all other DataChannels during  transmission of one large 
message, which some applications would be ok with, and others would not, 
and so those that would not would have to somehow deal with it.

So, applications that are ok with blocking other channels with large 
sends could be allowed to send arbitrary-sized messages, and receivers 
would be required to receive them (without chunking). Applications that 
do care about other channels would need to restrict their sending size 
when ndata isn't available at both ends.  We really only need to know if 
ndata is available and expose that fact to the application, not a 
receive size from the receiver.

So this plan would be:
1) We expose ndata being agreed to (currently == false);
2) We turn off the PPID chunking in Firefox (and live with ESR24 and 
current revs of Firefox not being compatible; see 2a)
2a) we make sure the app can detect if the local client has the ndata 
property at all (supports this solution).
3) We make sure Chrome and Firefox both implement both EOR sending and 
EOR reception and unlimited (or virtually) send/receive sizes.

For bonus points, on EOR reception of a blob, spool it to disk in the 
browser (if over some limit perhaps), and on EOR send of a blob, 
optimize it to avoid send-side memory hits and performance issues.

This will break large send/receive between Chrome and current/older 
Firefox - but those are broken today by Chrome not implement PPID 
chunking and Firefox doing so.  Firefox can still fall back on PPID 
chunking if it knows it's talking to another Firefox until we decide 
that this no longer matters.  I could be wrong, but I think Chrome 
doesn't support EOR sending currently; I know Firefox doesn't since we 
were using PPID chunking, so we'd need to add that.

Once ndata is supported, all will work well, and applications that 
didn't care about interleaving don't have to change, and those that do 
will trigger on ndata and switch (and they can test ndata==true mode 
reasonably well by forcing it in the app, since send(large) will work).

-- 
Randell Jesup -- rjesup a t mozilla d o t com


From nobody Mon Mar  3 07:39:01 2014
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On 26 Feb 2014, at 21:02, Randell Jesup <randell-ietf@jesup.org> wrote:

> On 2/26/2014 2:50 PM, Paul Kyzivat wrote:
>> Michael,
>>=20
>> There have been no replies to my comments:
>>=20
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11518.html
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11519.html
>>=20
>> The first of those is about channels, the 2nd about DCEP. One of my =
issues is that the partitioning between those two drafts is awkward. At =
the least some things from DCEP should be moved to the channel draft. It =
might be better to just merge those two drafts, or do a serious =
refactoring.
>=20
> There seem to have been a number of responses to your second posting =
at least.
>=20
>> How about:
>=20
>   Each SCTP user message contains a so called Payload Protocol
>   Identifier (PPID) that is passed to SCTP by the data channel layer
>   and sent to its peer.  This value is used to multiplex WebRTC Data
>   Channel Establishment Protocol messages (defined in =
[I-D.ietf-rtcweb-
>   data-protocol]) with messages containing user data on a data
>   channel. The PPID is also used to distinguish UTF-8 encoded user
>   data and binary encoded user data.
>=20
> That seems reasonable and clearer.
>=20
>=20
>> * Section 6.5
>>=20
>> This section contains:
>=20
>   Data channels can be opened by using internal or external
>   negotiation.  The details are out of scope of this document.
>=20
>   A simple protocol for internal negotiation is specified in
>   [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>=20
>> But internal and external negotiation are not defined in this =
document.
>>=20
>=20
> > I *thought* that internal negotiation was by definition negotiation =
by use of
> > rtcweb-data-protocol. (draft-ejzak-mmusic-data-channel-sdpneg-00 =
thinks so too,
> > but calls it "in-band negotiation".) There should be a good =
definition of these
> > terms, or reference to one. And more discussion if there can be =
other kinds of
> > internal negotiation. (If so, how would one be chosen?)
>=20
> Well, the text above says that ietf-rtcweb-data-protocol specifies an =
internal negotiation protocol,
> so it's not that far off.  The split does allow someone to use an =
alternative negotiation protocol
> (internal or external).
>=20
> How about:
>=20
>   Data channels can be opened by using negotiation within the SCTP =
association or external
>   negotiation.  External negotiation is defined as any method which =
results in an agreement
>   as to the parameters of a channel and the creation thereof.
>   The details are out of scope of this document.
>=20
>   A simple protocol for negotiation within the SCTP association is =
specified in
>   [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>=20
>=20
> Perhaps someone can wordsmith "external negotiation" better.
Taken, but with in-band and out-of-band...
>=20
>=20
>> * Section 6.6:
>>=20
>> Say:
>=20
>   All data sent on a Channel in both directions MUST be sent over the
>   underlying stream using the reliability defined when the Channel was
>   opened unless the options are changed, or per-message options are
>   specified by a higher level.
>=20
> > Is the recipient to consider it an error if messages are received =
with options different from those
> > defined for the channel? Also, is it an error if messages are =
received with a PPID that isn't specified
> > in Section 8? (And what about PPID 50?)
>=20
>> How are such channel errors to be treated?
>=20
>=20
> I'd propose that if messages are received with options that don't =
match the initial definition, that fact should be ignored and the =
message processed.
> If messages with an unknown PID are received, those messages should be =
ignored.
I'll bring that up at the WebRTC session.

Best regards
Michael
>=20
> In both cases, logging of such an event to the application (or onerror =
calls in WebRTC W3 JS layers) is allowed but not required. Note that the =
onerror callback is currently under-defined, so this might change.
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 3 March 2014 07:26, Randell Jesup <randell-ietf@jesup.org> wrote:
> I don't understand the "behind a flag" comment; an app using DataChannels
> for anything variable-length (unless with low bounds) must care until all
> old implementations are gone.

My point was that these people who you want to protect have already
renounced rights to protection by using the prefixed API.  Don't use
that as a basis for argument.  We should concentrate on building the
functionality we want to have.


From nobody Mon Mar  3 09:54:59 2014
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Subject: Re: [rtcweb] Open data channel issues
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On Mon, Mar 3, 2014 at 7:26 AM, Randell Jesup <randell-ietf@jesup.org>wrote:

> On 3/3/2014 8:53 AM, Martin Thomson wrote:
>
>> On 3 March 2014 03:44, Randell Jesup <randell-ietf@jesup.org> wrote:
>>
>>> Hopefully so, though I hate forcing apps to include a bunch of code for
>>> backwards compat for a year or two (and many won't, or won't test it,
>>> etc).
>>> That hurts the feature.  And today no one has ndata.
>>>
>> Since WebRTC is behind a flag, I don't find this particularly
>> convincing.  We can implement ndata.
>>
>
> I don't understand the "behind a flag" comment; an app using DataChannels
> for anything variable-length (unless with low bounds) must care until all
> old implementations are gone.  Since we already have DataChannel impls in
>  ESR releases, there's no avoiding this if we rely on that (even worse,
> since this undefined property doesn't exist yet (let alone in ESR), the app
> has to handle that case as well).  Or add browser and version sniffing to
> all the apps...
>
>
>  I think that we could even remove our temporary length restriction.
>>
>
> Sure, we can.  This will reduce or eliminate source changes for apps that
> don't care about interleaving, which is a plus.  It will cause total stall
> of all other DataChannels during  transmission of one large message, which
> some applications would be ok with, and others would not, and so those that
> would not would have to somehow deal with it.
>
> So, applications that are ok with blocking other channels with large sends
> could be allowed to send arbitrary-sized messages, and receivers would be
> required to receive them (without chunking). Applications that do care
> about other channels would need to restrict their sending size when ndata
> isn't available at both ends.  We really only need to know if ndata is
> available and expose that fact to the application, not a receive size from
> the receiver.
>
> So this plan would be:
> 1) We expose ndata being agreed to (currently == false);
> 2) We turn off the PPID chunking in Firefox (and live with ESR24 and
> current revs of Firefox not being compatible; see 2a)
> 2a) we make sure the app can detect if the local client has the ndata
> property at all (supports this solution).
> 3) We make sure Chrome and Firefox both implement both EOR sending and EOR
> reception and unlimited (or virtually) send/receive sizes.
>
> For bonus points, on EOR reception of a blob, spool it to disk in the
> browser (if over some limit perhaps), and on EOR send of a blob, optimize
> it to avoid send-side memory hits and performance issues.
>
> This will break large send/receive between Chrome and current/older
> Firefox - but those are broken today by Chrome not implement PPID chunking
> and Firefox doing so.  Firefox can still fall back on PPID chunking if it
> knows it's talking to another Firefox until we decide that this no longer
> matters.  I could be wrong, but I think Chrome doesn't support EOR sending
> currently; I know Firefox doesn't since we were using PPID chunking, so
> we'd need to add that.
>
> Once ndata is supported, all will work well, and applications that didn't
> care about interleaving don't have to change, and those that do will
> trigger on ndata and switch (and they can test ndata==true mode reasonably
> well by forcing it in the app, since send(large) will work).


While I agree with the final outcome of #3, I don't think going down the
1/2/2a path makes sense, since it leads to an additional intermediate state
over the current situation. i.e. we end up with 3 states:
a) today: max safe send is 16 KB
b) soon: ndata exists, but large sends lock out other sends
c) final: ndata exists, and large sends work OK

I would rather maintain a) or similar for now (using the max-message-size
param) and jump to c) when we have the right stuff in SCTP.

>
>
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--bcaec51d2eb8d7aa7704f3b77ba6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Mar 3, 2014 at 7:26 AM, Randell Jesup <span dir=3D"ltr">&lt=
;<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@j=
esup.org</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>On 3/3/2014 8:53 AM, Martin Thomson wro=
te:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 3 March 2014 03:44, Randell Jesup &lt;<a href=3D"mailto:randell-ietf@jes=
up.org" target=3D"_blank">randell-ietf@jesup.org</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hopefully so, though I hate forcing apps to include a bunch of code for<br>
backwards compat for a year or two (and many won&#39;t, or won&#39;t test i=
t, etc).<br>
That hurts the feature. =C2=A0And today no one has ndata.<br>
</blockquote>
Since WebRTC is behind a flag, I don&#39;t find this particularly<br>
convincing. =C2=A0We can implement ndata.<br>
</blockquote>
<br></div>
I don&#39;t understand the &quot;behind a flag&quot; comment; an app using =
DataChannels for anything variable-length (unless with low bounds) must car=
e until all old implementations are gone. =C2=A0Since we already have DataC=
hannel impls in =C2=A0ESR releases, there&#39;s no avoiding this if we rely=
 on that (even worse, since this undefined property doesn&#39;t exist yet (=
let alone in ESR), the app has to handle that case as well). =C2=A0Or add b=
rowser and version sniffing to all the apps...<div>


<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I think that we could even remove our temporary length restriction.<br>
</blockquote>
<br></div>
Sure, we can. =C2=A0This will reduce or eliminate source changes for apps t=
hat don&#39;t care about interleaving, which is a plus. =C2=A0It will cause=
 total stall of all other DataChannels during =C2=A0transmission of one lar=
ge message, which some applications would be ok with, and others would not,=
 and so those that would not would have to somehow deal with it.<br>



<br>
So, applications that are ok with blocking other channels with large sends =
could be allowed to send arbitrary-sized messages, and receivers would be r=
equired to receive them (without chunking). Applications that do care about=
 other channels would need to restrict their sending size when ndata isn&#3=
9;t available at both ends. =C2=A0We really only need to know if ndata is a=
vailable and expose that fact to the application, not a receive size from t=
he receiver.<br>



<br>
So this plan would be:<br>
1) We expose ndata being agreed to (currently =3D=3D false);<br>
2) We turn off the PPID chunking in Firefox (and live with ESR24 and curren=
t revs of Firefox not being compatible; see 2a)<br>
2a) we make sure the app can detect if the local client has the ndata prope=
rty at all (supports this solution).<br>
3) We make sure Chrome and Firefox both implement both EOR sending and EOR =
reception and unlimited (or virtually) send/receive sizes.<br>
<br>
For bonus points, on EOR reception of a blob, spool it to disk in the brows=
er (if over some limit perhaps), and on EOR send of a blob, optimize it to =
avoid send-side memory hits and performance issues.<br>
<br>
This will break large send/receive between Chrome and current/older Firefox=
 - but those are broken today by Chrome not implement PPID chunking and Fir=
efox doing so. =C2=A0Firefox can still fall back on PPID chunking if it kno=
ws it&#39;s talking to another Firefox until we decide that this no longer =
matters. =C2=A0I could be wrong, but I think Chrome doesn&#39;t support EOR=
 sending currently; I know Firefox doesn&#39;t since we were using PPID chu=
nking, so we&#39;d need to add that.<br>



<br>
Once ndata is supported, all will work well, and applications that didn&#39=
;t care about interleaving don&#39;t have to change, and those that do will=
 trigger on ndata and switch (and they can test ndata=3D=3Dtrue mode reason=
ably well by forcing it in the app, since send(large) will work).</blockquo=
te>


<div><br></div><div>While I agree with the final outcome of #3, I don&#39;t=
 think going down the 1/2/2a path makes sense, since it leads to an additio=
nal intermediate state over the current situation. i.e. we end up with 3 st=
ates:</div>


<div>a) today: max safe send is 16 KB</div><div>b) soon: ndata exists, but =
large sends lock out other sends</div><div>c) final: ndata exists, and larg=
e sends work OK</div><div><br></div><div>I would rather maintain a) or simi=
lar for now (using the max-message-size param) and jump to c) when we have =
the right stuff in SCTP.</div>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><br>
<br>
-- <br>
Randell Jesup -- rjesup a t mozilla d o t com<br>
<br></div><div><div>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--bcaec51d2eb8d7aa7704f3b77ba6--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>, Randell Jesup <randell-ietf@jesup.org>
Thread-Topic: [rtcweb] Open data channel issues
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--_000_E1FE4C082A89A246A11D7F32A95A17826E006069US70UWXCHMBA02z_--


From nobody Mon Mar  3 14:45:49 2014
Return-Path: <Michael.Tuexen@lurchi.franken.de>
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Date: Mon, 3 Mar 2014 22:45:36 +0000
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References: <31B9253D-E826-4D07-A8A1-1B062B50F163@lurchi.franken.de> <530D9CC5.5080508@ericsson.com> <5312FBBC.5080006@jesup.org> <CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com> <5313EC0D.9030808@jesup.org> <CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com> <53146B2C.8080008@jesup.org> <CABkgnnV1BdAMvFqGq8xiL0Zu28aZBCAdPwZB8whpMeXrCD7AHQ@mail.gmail.com> <53149F0C.6070001@jesup.org> <CAOJ7v-0xSq2xvF_5Do-R4SG6yAVNdqELwRvt_VBU69ir6nHhGQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17826E006069@US70UWXCHMBA02.zam.alcatel-lucent.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Open data channel issues
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On 03 Mar 2014, at 22:19, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:

> Sorry to jump in from middle of nowhere in this interesting =
conversation. I couldn=92t resist providing this feedback and I hope it =
is useful.
> =20
> First, on =93EOR solving monopolization issue=94:
> I could be wrong, but I am not sure how EOR based approach avoids SCTP =
link monopolization?! Per my understanding EOR is a local send () =
utility rather than receive side utility and it is implemented on top of =
base SCTP RFC (so no indication of special EOR in SCTP headers other =
than base SCTP RFC B/E fragment indication bits). Once a send () with =
EOR set to false is called, on a stream id, the SCTP link is monopolized =
until send () on same stream id with EOR set to true is called; so same =
limitation exist.  EOR allows sending side to start sending chunks =
without waiting for complete data. EOR has no impact on receiving side, =
the SCTP stack waits for last chunk (which is the result of sending side =
setting EOR=3Dtrue) and delivers entire message to app.
You are right, if you consider RFC 4960. But using
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
overcomes this limitation. You can have on each stream a message being =
sent in explicit EOR mode.
Using an interleaving scheduler, they get interleaved on the wire. So =
you need the NDATA extension...
> =20
> New PC approach:
> If an application is sensitive to SCTP link monopolisation then there =
is another option it can consider =96 use of separate new PeerConnection =
(PC) for the data channel(s) which may monopolize the link.
> Depending on how an application is designed, using a new PC could be =
simpler than having an application level chunking protocol =
(fragmentation and reassembly).  If app needs such =93heavy=94 data =
channel infrequently then it can create new PC as needed; if it is =
needed frequently then a dedicated PC can be allocated and keep it open =
all the time.
> =20
> One disadvantage with application level chunking protocol is both the =
peers need to use it; so it may not work between clients of different =
types or same clients with different versions. New PC based approach =
obviously does not have this issue as no chunking is involved.
> =20
> At first, new PC approach may appear heavy weight wrt # of PCs in use =
(which I don=92t think an issue on browsers), ICE overhead for new PC =
etc. But in comparision to application level chunking new PC approach =
may not be that bad.
I think the large message support with NDATA is sufficient to do =
whatever RTCWeb wants...
> =20
> Please see additional comments inline.
> =20
> -Raju
> =20
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Justin =
Uberti
> Sent: Monday, March 03, 2014 11:54 AM
> To: Randell Jesup
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Open data channel issues
> =20
> =20
> =20
>=20
> On Mon, Mar 3, 2014 at 7:26 AM, Randell Jesup <randell-ietf@jesup.org> =
wrote:
> On 3/3/2014 8:53 AM, Martin Thomson wrote:
> On 3 March 2014 03:44, Randell Jesup <randell-ietf@jesup.org> wrote:
> Hopefully so, though I hate forcing apps to include a bunch of code =
for
> backwards compat for a year or two (and many won't, or won't test it, =
etc).
> That hurts the feature.  And today no one has ndata.
> Since WebRTC is behind a flag, I don't find this particularly
> convincing.  We can implement ndata.
> =20
> I don't understand the "behind a flag" comment; an app using =
DataChannels for anything variable-length (unless with low bounds) must =
care until all old implementations are gone.  Since we already have =
DataChannel impls in  ESR releases, there's no avoiding this if we rely =
on that (even worse, since this undefined property doesn't exist yet =
(let alone in ESR), the app has to handle that case as well).  Or add =
browser and version sniffing to all the apps...
> =20
>=20
> I think that we could even remove our temporary length restriction.
> =20
> Sure, we can.  This will reduce or eliminate source changes for apps =
that don't care about interleaving, which is a plus.  It will cause =
total stall of all other DataChannels during  transmission of one large =
message, which some applications would be ok with, and others would not, =
and so those that would not would have to somehow deal with it.
>=20
> So, applications that are ok with blocking other channels with large =
sends could be allowed to send arbitrary-sized messages, and receivers =
would be required to receive them (without chunking). Applications that =
do care about other channels would need to restrict their sending size =
when ndata isn't available at both ends.  We really only need to know if =
ndata is available and expose that fact to the application, not a =
receive size from the receiver.
>=20
> So this plan would be:
> 1) We expose ndata being agreed to (currently =3D=3D false);
> 2) We turn off the PPID chunking in Firefox (and live with ESR24 and =
current revs of Firefox not being compatible; see 2a)
> 2a) we make sure the app can detect if the local client has the ndata =
property at all (supports this solution).
> 3) We make sure Chrome and Firefox both implement both EOR sending and =
EOR reception and unlimited (or virtually) send/receive sizes.
> [Raju] This requires PeerConnection API to indicate EOR during send(). =
Right? Which may be useful in some cases where app wants to send partial =
data as it is available/ready. But as explained above, in my opinion EOR =
won=92t solve SCTP link monopolization and unncessarily introduces =
overhead at application.
At the API to the SCTP stack you need EOR support. But this API is not =
exposed to the JS user.
The JS user can just call send(bigmessage) and it works...
>=20
> For bonus points, on EOR reception of a blob, spool it to disk in the =
browser (if over some limit perhaps), and on EOR send of a blob, =
optimize it to avoid send-side memory hits and performance issues.
>=20
> This will break large send/receive between Chrome and current/older =
Firefox - but those are broken today by Chrome not implement PPID =
chunking and Firefox doing so.  Firefox can still fall back on PPID =
chunking if it knows it's talking to another Firefox until we decide =
that this no longer matters.  I could be wrong, but I think Chrome =
doesn't support EOR sending currently; I know Firefox doesn't since we =
were using PPID chunking, so we'd need to add that.
>=20
> Once ndata is supported, all will work well, and applications that =
didn't care about interleaving don't have to change, and those that do =
will trigger on ndata and switch (and they can test ndata=3D=3Dtrue mode =
reasonably well by forcing it in the app, since send(large) will work).
> =20
> While I agree with the final outcome of #3, I don't think going down =
the 1/2/2a path makes
> sense, since it leads to an additional intermediate state over the =
current situation. i.e. we end up with 3 states:
> a) today: max safe send is 16 KB
> b) soon: ndata exists, but large sends lock out other sends
> c) final: ndata exists, and large sends work OK
> =20
> I would rather maintain a) or similar for now (using the =
max-message-size param) and jump to c) when we have the right stuff in =
SCTP.
> [Raju] With 16KB safe size won=92t applications have to have their own =
chunking protocol? as per my understanding EOR won=92t solve the =
monopolization issue, so it adds no value.
> =20
> To avoid applications coming up with their own chunking protocol, an =
attractive alternative would be to use the PPID chunking that=92s =
already mentioned abovr and implemented by Firefox. Then the issue of =
interworking pushed down to browser from application area. But this =
obviously requires browsers on both the ends to support this PPID based =
chunking.
... correct. And at the last RTCWeb session it was decided to avoid this =
intermediate step and
just use NDATA...

Best regards
Michael
> =20
> -Raju
> =20
>=20
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 3/3/2014 12:54 PM, Justin Uberti wrote:
>
>
>
> On Mon, Mar 3, 2014 at 7:26 AM, Randell Jesup <randell-ietf@jesup.org 
> <mailto:randell-ietf@jesup.org>> wrote:
>
>     On 3/3/2014 8:53 AM, Martin Thomson wrote:
>
>         On 3 March 2014 03:44, Randell Jesup <randell-ietf@jesup.org
>         <mailto:randell-ietf@jesup.org>> wrote:
>
>             Hopefully so, though I hate forcing apps to include a
>             bunch of code for
>             backwards compat for a year or two (and many won't, or
>             won't test it, etc).
>             That hurts the feature.  And today no one has ndata.
>
>         Since WebRTC is behind a flag, I don't find this particularly
>         convincing.  We can implement ndata.
>
>
>     I don't understand the "behind a flag" comment; an app using
>     DataChannels for anything variable-length (unless with low bounds)
>     must care until all old implementations are gone.  Since we
>     already have DataChannel impls in  ESR releases, there's no
>     avoiding this if we rely on that (even worse, since this undefined
>     property doesn't exist yet (let alone in ESR), the app has to
>     handle that case as well).  Or add browser and version sniffing to
>     all the apps...
>
>
>         I think that we could even remove our temporary length
>         restriction.
>
>
>     Sure, we can.  This will reduce or eliminate source changes for
>     apps that don't care about interleaving, which is a plus.  It will
>     cause total stall of all other DataChannels during  transmission
>     of one large message, which some applications would be ok with,
>     and others would not, and so those that would not would have to
>     somehow deal with it.
>
>     So, applications that are ok with blocking other channels with
>     large sends could be allowed to send arbitrary-sized messages, and
>     receivers would be required to receive them (without chunking).
>     Applications that do care about other channels would need to
>     restrict their sending size when ndata isn't available at both
>     ends.  We really only need to know if ndata is available and
>     expose that fact to the application, not a receive size from the
>     receiver.
>
>     So this plan would be:
>     1) We expose ndata being agreed to (currently == false);
>     2) We turn off the PPID chunking in Firefox (and live with ESR24
>     and current revs of Firefox not being compatible; see 2a)
>     2a) we make sure the app can detect if the local client has the
>     ndata property at all (supports this solution).
>     3) We make sure Chrome and Firefox both implement both EOR sending
>     and EOR reception and unlimited (or virtually) send/receive sizes.
>
>     For bonus points, on EOR reception of a blob, spool it to disk in
>     the browser (if over some limit perhaps), and on EOR send of a
>     blob, optimize it to avoid send-side memory hits and performance
>     issues.
>
>     This will break large send/receive between Chrome and
>     current/older Firefox - but those are broken today by Chrome not
>     implement PPID chunking and Firefox doing so.  Firefox can still
>     fall back on PPID chunking if it knows it's talking to another
>     Firefox until we decide that this no longer matters.  I could be
>     wrong, but I think Chrome doesn't support EOR sending currently; I
>     know Firefox doesn't since we were using PPID chunking, so we'd
>     need to add that.
>
>     Once ndata is supported, all will work well, and applications that
>     didn't care about interleaving don't have to change, and those
>     that do will trigger on ndata and switch (and they can test
>     ndata==true mode reasonably well by forcing it in the app, since
>     send(large) will work).
>
>
> While I agree with the final outcome of #3, I don't think going down 
> the 1/2/2a path makes sense, since it leads to an additional 
> intermediate state over the current situation. i.e. we end up with 3 
> states:
> a) today: max safe send is 16 KB
> b) soon: ndata exists, but large sends lock out other sends
> c) final: ndata exists, and large sends work OK
>
> I would rather maintain a) or similar for now (using the 
> max-message-size param) and jump to c) when we have the right stuff in 
> SCTP.

So, is there any significant advantage to using a max-message-size param 
(with >16K) as an interim step?  You get 3 similar steps (though perhaps 
with a smaller lockout).

To take your suggestion:
a) applications assume 16KB if ndata wasn't negotiated (per the current 
suggestion)
b) implement ndata and EOR support in the browsers, and drop the 16KB 
limit when ndata is negotiated.
All this needs is "is ndata supported by this association" exposed to 
the app; no sdp changes needed.  (One less dependency!)

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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    <div class="moz-cite-prefix">On 3/3/2014 12:54 PM, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-0xSq2xvF_5Do-R4SG6yAVNdqELwRvt_VBU69ir6nHhGQ@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra"><br>
          <br>
          <div class="gmail_quote">On Mon, Mar 3, 2014 at 7:26 AM,
            Randell Jesup <span dir="ltr">&lt;<a moz-do-not-send="true"
                href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div>On 3/3/2014 8:53 AM, Martin Thomson wrote:<br>
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  On 3 March 2014 03:44, Randell Jesup &lt;<a
                    moz-do-not-send="true"
                    href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a>&gt;
                  wrote:<br>
                  <blockquote class="gmail_quote" style="margin:0 0 0
                    .8ex;border-left:1px #ccc solid;padding-left:1ex">
                    Hopefully so, though I hate forcing apps to include
                    a bunch of code for<br>
                    backwards compat for a year or two (and many won't,
                    or won't test it, etc).<br>
                    That hurts the feature. &nbsp;And today no one has ndata.<br>
                  </blockquote>
                  Since WebRTC is behind a flag, I don't find this
                  particularly<br>
                  convincing. &nbsp;We can implement ndata.<br>
                </blockquote>
                <br>
              </div>
              I don't understand the "behind a flag" comment; an app
              using DataChannels for anything variable-length (unless
              with low bounds) must care until all old implementations
              are gone. &nbsp;Since we already have DataChannel impls in &nbsp;ESR
              releases, there's no avoiding this if we rely on that
              (even worse, since this undefined property doesn't exist
              yet (let alone in ESR), the app has to handle that case as
              well). &nbsp;Or add browser and version sniffing to all the
              apps...
              <div>
                <br>
                <br>
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  I think that we could even remove our temporary length
                  restriction.<br>
                </blockquote>
                <br>
              </div>
              Sure, we can. &nbsp;This will reduce or eliminate source
              changes for apps that don't care about interleaving, which
              is a plus. &nbsp;It will cause total stall of all other
              DataChannels during &nbsp;transmission of one large message,
              which some applications would be ok with, and others would
              not, and so those that would not would have to somehow
              deal with it.<br>
              <br>
              So, applications that are ok with blocking other channels
              with large sends could be allowed to send arbitrary-sized
              messages, and receivers would be required to receive them
              (without chunking). Applications that do care about other
              channels would need to restrict their sending size when
              ndata isn't available at both ends. &nbsp;We really only need
              to know if ndata is available and expose that fact to the
              application, not a receive size from the receiver.<br>
              <br>
              So this plan would be:<br>
              1) We expose ndata being agreed to (currently == false);<br>
              2) We turn off the PPID chunking in Firefox (and live with
              ESR24 and current revs of Firefox not being compatible;
              see 2a)<br>
              2a) we make sure the app can detect if the local client
              has the ndata property at all (supports this solution).<br>
              3) We make sure Chrome and Firefox both implement both EOR
              sending and EOR reception and unlimited (or virtually)
              send/receive sizes.<br>
              <br>
              For bonus points, on EOR reception of a blob, spool it to
              disk in the browser (if over some limit perhaps), and on
              EOR send of a blob, optimize it to avoid send-side memory
              hits and performance issues.<br>
              <br>
              This will break large send/receive between Chrome and
              current/older Firefox - but those are broken today by
              Chrome not implement PPID chunking and Firefox doing so.
              &nbsp;Firefox can still fall back on PPID chunking if it knows
              it's talking to another Firefox until we decide that this
              no longer matters. &nbsp;I could be wrong, but I think Chrome
              doesn't support EOR sending currently; I know Firefox
              doesn't since we were using PPID chunking, so we'd need to
              add that.<br>
              <br>
              Once ndata is supported, all will work well, and
              applications that didn't care about interleaving don't
              have to change, and those that do will trigger on ndata
              and switch (and they can test ndata==true mode reasonably
              well by forcing it in the app, since send(large) will
              work).</blockquote>
            <div><br>
            </div>
            <div>While I agree with the final outcome of #3, I don't
              think going down the 1/2/2a path makes sense, since it
              leads to an additional intermediate state over the current
              situation. i.e. we end up with 3 states:</div>
            <div>a) today: max safe send is 16 KB</div>
            <div>b) soon: ndata exists, but large sends lock out other
              sends</div>
            <div>c) final: ndata exists, and large sends work OK</div>
            <div><br>
            </div>
            <div>I would rather maintain a) or similar for now (using
              the max-message-size param) and jump to c) when we have
              the right stuff in SCTP.<br>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    So, is there any significant advantage to using a max-message-size
    param (with &gt;16K) as an interim step?&nbsp; You get 3 similar steps
    (though perhaps with a smaller lockout).<br>
    <br>
    To take your suggestion:<br>
    a) applications assume 16KB if ndata wasn't negotiated (per the
    current suggestion)<br>
    b) implement ndata and EOR support in the browsers, and drop the
    16KB limit when ndata is negotiated.<br>
    All this needs is "is ndata supported by this association" exposed
    to the app; no sdp changes needed.&nbsp; (One less dependency!)<br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Randell Jesup -- rjesup a t mozilla d o t com
</pre>
  </body>
</html>

--------------050703090904060301060305--


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Date: Tue, 4 Mar 2014 07:00:18 +0000
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To: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Open data channel issues
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On 4 March 2014 03:38, Randell Jesup <randell-ietf@jesup.org> wrote:
> To take your suggestion:
> a) applications assume 16KB if ndata wasn't negotiated (per the current
> suggestion)
> b) implement ndata and EOR support in the browsers, and drop the 16KB limit
> when ndata is negotiated.
> All this needs is "is ndata supported by this association" exposed to the
> app; no sdp changes needed.  (One less dependency!)


Or, to avoid vestigal crap in the API in the end state, when you
install the 16K limitation, add an indicator that applications can
check.  Once you remove the cap, remove the indicator.

if (dataChannel.isNotVeryGood) {
  chunkMessage(message);
} else {
  dataChannel.send(message);
}


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On 3/4/2014 2:00 AM, Martin Thomson wrote:
> On 4 March 2014 03:38, Randell Jesup <randell-ietf@jesup.org> wrote:
>> To take your suggestion:
>> a) applications assume 16KB if ndata wasn't negotiated (per the current
>> suggestion)
>> b) implement ndata and EOR support in the browsers, and drop the 16KB limit
>> when ndata is negotiated.
>> All this needs is "is ndata supported by this association" exposed to the
>> app; no sdp changes needed.  (One less dependency!)
>
> Or, to avoid vestigal crap in the API in the end state, when you
> install the 16K limitation, add an indicator that applications can
> check.  Once you remove the cap, remove the indicator.
>
> if (dataChannel.isNotVeryGood) {
>    chunkMessage(message);
> } else {
>    dataChannel.send(message);
> }

Yup, that works for me.  dataChannel.requiresFragmentation would be the 
wordy version or mustFragmentAt16K for self-documentation.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


From nobody Mon Mar  3 23:55:34 2014
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On 04 Mar 2014, at 07:30, Randell Jesup <randell-ietf@jesup.org> wrote:

> On 3/4/2014 2:00 AM, Martin Thomson wrote:
>> On 4 March 2014 03:38, Randell Jesup <randell-ietf@jesup.org> wrote:
>>> To take your suggestion:
>>> a) applications assume 16KB if ndata wasn't negotiated (per the =
current
>>> suggestion)
>>> b) implement ndata and EOR support in the browsers, and drop the =
16KB limit
>>> when ndata is negotiated.
>>> All this needs is "is ndata supported by this association" exposed =
to the
>>> app; no sdp changes needed.  (One less dependency!)
>>=20
>> Or, to avoid vestigal crap in the API in the end state, when you
>> install the 16K limitation, add an indicator that applications can
>> check.  Once you remove the cap, remove the indicator.
>>=20
>> if (dataChannel.isNotVeryGood) {
>>   chunkMessage(message);
>> } else {
>>   dataChannel.send(message);
>> }
>=20
> Yup, that works for me.  dataChannel.requiresFragmentation would be =
the wordy version or mustFragmentAt16K for self-documentation.
Or you return a number which gives you the limit if you don't want to =
specify the
16K limit...

Best regards
Michael
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Tue Mar  4 01:02:11 2014
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On 3/4/2014 2:55 AM, Michael Tuexen wrote:
> On 04 Mar 2014, at 07:30, Randell Jesup <randell-ietf@jesup.org> wrote:
>
>> On 3/4/2014 2:00 AM, Martin Thomson wrote:
>>> On 4 March 2014 03:38, Randell Jesup <randell-ietf@jesup.org> wrote:
>>>> To take your suggestion:
>>>> a) applications assume 16KB if ndata wasn't negotiated (per the current
>>>> suggestion)
>>>> b) implement ndata and EOR support in the browsers, and drop the 16KB limit
>>>> when ndata is negotiated.
>>>> All this needs is "is ndata supported by this association" exposed to the
>>>> app; no sdp changes needed.  (One less dependency!)
>>> Or, to avoid vestigal crap in the API in the end state, when you
>>> install the 16K limitation, add an indicator that applications can
>>> check.  Once you remove the cap, remove the indicator.
>>>
>>> if (dataChannel.isNotVeryGood) {
>>>    chunkMessage(message);
>>> } else {
>>>    dataChannel.send(message);
>>> }


Wait, the reason why we had it be 16K as a default was that current 
impls don't have the property.  So datachannel.isntFixed (ok, better 
names can be found - but that's W3 anyways.


-- 
Randell Jesup -- rjesup a t mozilla d o t com


From nobody Tue Mar  4 01:09:51 2014
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References: <31B9253D-E826-4D07-A8A1-1B062B50F163@lurchi.franken.de> <530D9CC5.5080508@ericsson.com> <5312FBBC.5080006@jesup.org> <CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com> <5313EC0D.9030808@jesup.org> <CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com> <53146B2C.8080008@jesup.org> <CABkgnnV1BdAMvFqGq8xiL0Zu28aZBCAdPwZB8whpMeXrCD7AHQ@mail.gmail.com> <53149F0C.6070001@jesup.org> <CAOJ7v-0xSq2xvF_5Do-R4SG6yAVNdqELwRvt_VBU69ir6nHhGQ@mail.gmail.com> <53154AA8.4000202@jesup.org> <CABkgnnVWESeVgrPjYnxuhB5w1QTjbXNW+zBLfYRwUL059Q0_hw@mail.gmail.com> <53158103.4000809@jesup.org> <1D3890D9-AF14-4648-820D-7586134D56DA@lurchi.franken.de> <53159637.8050909@jesup.org>
Date: Tue, 4 Mar 2014 09:09:36 +0000
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From: Martin Thomson <martin.thomson@gmail.com>
To: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Open data channel issues
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On 4 March 2014 09:00, Randell Jesup <randell-ietf@jesup.org> wrote:
> Wait, the reason why we had it be 16K as a default was that current impls
> don't have the property.  So datachannel.isntFixed (ok, better names can be
> found - but that's W3 anyways.

Some current implementations don't have a limit either.

We could spend lots of our precious cycles on fixing shit that's both
broken and unchangeable.  Or, we could try to design the thing that we
actually want.


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Date: Tue, 04 Mar 2014 11:03:04 +0000
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [rtcweb] *MUST* DCEP?
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Ted,

(I think this was open issue #8)

You closed discussion on this, saying that this should be specified in 
some rtcweb "system" document. IIUC, then I disagree.

This is important to others besides rtcweb. We want to use this 
mechanism, both with and without components that are implementing 
rtcweb. Certainly for non-browsers.

This ultimately comes down to what I can expect after having negotiated 
(in SDP) the establishment of an SCTP association. When doing so, I use 
a=sctpmap to negotiate the how the association will be used. If I intend 
to use data channels and DCEP then I want an assurance that the other 
end does too. (That is the purpose of the SDP negotiation.)

AFAIK, current when I specify 'a=sctpmap:webrtc-datachannel' (or 
'a=sctp-datachannel' depending on where I look), that means the other 
side supports data channels *and* guarantees support for DCEP. There is 
no means to indicate support for data channels *without* DCEP.

I don't see any urgent need to have data channels without DCEP, as long 
as *use* of DCEP is optional.

IMO things would be less confusing if the two documents were merged into 
one, that defines the *Data Channel Protocol* (that runs over SCTP). 
This protocol runs over pairs of SCTP streams, has a small number of 
messages (open, ack, string-payload, binary-payload, close), and encodes 
them in SCTP using SCTP messages with particular payloads and SCTP 
stream reset. The open and ack messages would be optional.

	Thanks,
	Paul


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On 4 March 2014 11:03, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
> This is important to others besides rtcweb. We want to use this mechanism,
> both with and without components that are implementing rtcweb. Certainly for
> non-browsers.

if (need(DCEP)) {
  addStatementToMyDoc("MUST", DCEP)
}


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From: Ted Hardie <ted.ietf@gmail.com>
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--001a11c3c196b643ec04f3c60b4c
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On Tue, Mar 4, 2014 at 11:03 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Ted,
>
> (I think this was open issue #8)
>
> You closed discussion on this, saying that this should be specified in
> some rtcweb "system" document. IIUC, then I disagree.
>
> This is important to others besides rtcweb. We want to use this mechanism,
> both with and without components that are implementing rtcweb. Certainly
> for non-browsers.
>
>
Help me understand why it wouldn't then be in their systems documents?

Ted




> This ultimately comes down to what I can expect after having negotiated
> (in SDP) the establishment of an SCTP association. When doing so, I use
> a=sctpmap to negotiate the how the association will be used. If I intend to
> use data channels and DCEP then I want an assurance that the other end does
> too. (That is the purpose of the SDP negotiation.)
>
> AFAIK, current when I specify 'a=sctpmap:webrtc-datachannel' (or
> 'a=sctp-datachannel' depending on where I look), that means the other side
> supports data channels *and* guarantees support for DCEP. There is no means
> to indicate support for data channels *without* DCEP.
>
> I don't see any urgent need to have data channels without DCEP, as long as
> *use* of DCEP is optional.
>
> IMO things would be less confusing if the two documents were merged into
> one, that defines the *Data Channel Protocol* (that runs over SCTP). This
> protocol runs over pairs of SCTP streams, has a small number of messages
> (open, ack, string-payload, binary-payload, close), and encodes them in
> SCTP using SCTP messages with particular payloads and SCTP stream reset.
> The open and ack messages would be optional.
>
>         Thanks,
>         Paul
>

--001a11c3c196b643ec04f3c60b4c
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Tue, Mar 4, 2014 at 11:03 AM=
, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@alum.mit.ed=
u" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;</span> wrote:<br><div cl=
ass=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Ted,<br>
<br>
(I think this was open issue #8)<br>
<br>
You closed discussion on this, saying that this should be specified in some=
 rtcweb &quot;system&quot; document. IIUC, then I disagree.<br>
<br>
This is important to others besides rtcweb. We want to use this mechanism, =
both with and without components that are implementing rtcweb. Certainly fo=
r non-browsers.<br>
<br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-family=
:georgia,serif">Help me understand why it wouldn&#39;t then be in their sys=
tems documents?<br><br>Ted</div><br><br>=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">

This ultimately comes down to what I can expect after having negotiated (in=
 SDP) the establishment of an SCTP association. When doing so, I use a=3Dsc=
tpmap to negotiate the how the association will be used. If I intend to use=
 data channels and DCEP then I want an assurance that the other end does to=
o. (That is the purpose of the SDP negotiation.)<br>

<br>
AFAIK, current when I specify &#39;a=3Dsctpmap:webrtc-datachannel&#39; (or =
&#39;a=3Dsctp-datachannel&#39; depending on where I look), that means the o=
ther side supports data channels *and* guarantees support for DCEP. There i=
s no means to indicate support for data channels *without* DCEP.<br>

<br>
I don&#39;t see any urgent need to have data channels without DCEP, as long=
 as *use* of DCEP is optional.<br>
<br>
IMO things would be less confusing if the two documents were merged into on=
e, that defines the *Data Channel Protocol* (that runs over SCTP). This pro=
tocol runs over pairs of SCTP streams, has a small number of messages (open=
, ack, string-payload, binary-payload, close), and encodes them in SCTP usi=
ng SCTP messages with particular payloads and SCTP stream reset. The open a=
nd ack messages would be optional.<br>

<br>
=A0 =A0 =A0 =A0 Thanks,<br>
=A0 =A0 =A0 =A0 Paul<br>
</blockquote></div><br></div></div>

--001a11c3c196b643ec04f3c60b4c--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Open data channel issues
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References: <31B9253D-E826-4D07-A8A1-1B062B50F163@lurchi.franken.de> <530D9CC5.5080508@ericsson.com> <5312FBBC.5080006@jesup.org> <CAOJ7v-2GHt37u8raWDKquNFLCjSv-ptP0YGojPwuLv02da_m1Q@mail.gmail.com> <5313EC0D.9030808@jesup.org> <CAOJ7v-3L5_V2wuwVLKpz61VHey7yx3LnG_xrPLb5W+6DOLiEJw@mail.gmail.com> <53146B2C.8080008@jesup.org> <CABkgnnV1BdAMvFqGq8xiL0Zu28aZBCAdPwZB8whpMeXrCD7AHQ@mail.gmail.com> <53149F0C.6070001@jesup.org> <CAOJ7v-0xSq2xvF_5Do-R4SG6yAVNdqELwRvt_VBU69ir6nHhGQ@mail.gmail.com> <E1FE4C082A89A246A11D7F32A95A17826E006069@US70UWXCHMBA02.zam.alcatel-lucent.com> <7387F0E6-2DE0-4372-8559-B9DE9B9F365F@lurchi.franken.de>
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Please see inline comments.

> -----Original Message-----
> From: Michael Tuexen [mailto:Michael.Tuexen@lurchi.franken.de]
> Sent: Monday, March 03, 2014 4:46 PM
> To: Makaraju, Maridi Raju (Raju)
> Cc: Justin Uberti; Randell Jesup; rtcweb@ietf.org
> Subject: Re: [rtcweb] Open data channel issues
>=20
> On 03 Mar 2014, at 22:19, Makaraju, Maridi Raju (Raju)
> <Raju.Makaraju@alcatel-lucent.com> wrote:
>=20
> > Sorry to jump in from middle of nowhere in this interesting conversatio=
n.
> I couldn't resist providing this feedback and I hope it is useful.
> >
> > First, on "EOR solving monopolization issue":
> > I could be wrong, but I am not sure how EOR based approach avoids SCTP
> link monopolization?! Per my understanding EOR is a local send () utility
> rather than receive side utility and it is implemented on top of base SCT=
P
> RFC (so no indication of special EOR in SCTP headers other than base SCTP
> RFC B/E fragment indication bits). Once a send () with EOR set to false i=
s
> called, on a stream id, the SCTP link is monopolized until send () on sam=
e
> stream id with EOR set to true is called; so same limitation exist.  EOR
> allows sending side to start sending chunks without waiting for complete
> data. EOR has no impact on receiving side, the SCTP stack waits for last
> chunk (which is the result of sending side setting EOR=3Dtrue) and delive=
rs
> entire message to app.
> You are right, if you consider RFC 4960. But using
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
> overcomes this limitation. You can have on each stream a message being se=
nt
> in explicit EOR mode.
> Using an interleaving scheduler, they get interleaved on the wire. So you
> need the NDATA extension...

[Raju]=20
Thanks for clarifying. Earlier it wasn't clear to me as EOR was referred to=
 be EOR+NDATA. I was reading EOR as just EOR support without NDATA.
Yes, NDATA support is a must to avoid monopolization.
[/Raju]

> >
> > New PC approach:
> > If an application is sensitive to SCTP link monopolisation then there i=
s
> another option it can consider - use of separate new PeerConnection (PC) =
for
> the data channel(s) which may monopolize the link.
> > Depending on how an application is designed, using a new PC could be
> simpler than having an application level chunking protocol (fragmentation
> and reassembly).  If app needs such "heavy" data channel infrequently the=
n
> it can create new PC as needed; if it is needed frequently then a dedicat=
ed
> PC can be allocated and keep it open all the time.
> >
> > One disadvantage with application level chunking protocol is both the
> peers need to use it; so it may not work between clients of different typ=
es
> or same clients with different versions. New PC based approach obviously
> does not have this issue as no chunking is involved.
> >
> > At first, new PC approach may appear heavy weight wrt # of PCs in use
> (which I don't think an issue on browsers), ICE overhead for new PC etc. =
But
> in comparision to application level chunking new PC approach may not be t=
hat
> bad.
> I think the large message support with NDATA is sufficient to do whatever
> RTCWeb wants...

[Raju]=20
Yes, I agree that NDATA is the final solution. But in the interim, applicat=
ion chunking per 16K size requires some chunking protocol at app.
Do we know the timeframe for NDATA support from browsers?
Until NDATA is available, to avoid monopolization one can use new PC as an =
alternative to application level chunking (which also require some kind of =
flow/congestion control).
Appreictae any comments on this approach.
[/Raju]

Best Regards
Raju


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Data Presentation

Issue #1=20

- Will say MUST use NDATA and ignore intermediate states till we have =
that
- Will keep the signaling of receiver seize limits

Issue #2
- add general statement on congestion control and specific for TCP =
windows based congestion control=20


Issue #3
- say nothing about this in our specification and leave it to the SCTP =
specifications
- will propose the path max retrain =3D association max retrans in TSVWG
- no text involving this in rtcweb specs=20

Issue #4=20
- will remove negation and discussion of this. It is a point for the WG =
developing  SCTP.
- Ted to send text on how to craft such a statement=20


Issue #5=20
- part of the issue here is there is a protocol on top of SCTP=20
- change to text say =93reset channel=94


Issue #6
-  will use the the "IANA registry for WebSocket Subprotocol names "


Issue #7=20
- skipped due to time=20


Issue #8
- The transports documents should make the DCEP  be MTI=20
- The data channel draft will not require implementation of DCEP


Issue #9=20
- this is an example of something you can do with tool set we provide =
and may or may not be a bad idea=20
- do nothing for now and be sad later=20


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To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Open data channel issues
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> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: Tuesday, March 04, 2014 3:01 AM
> On 3/4/2014 2:55 AM, Michael Tuexen wrote:
> > On 04 Mar 2014, at 07:30, Randell Jesup <randell-ietf@jesup.org> wrote:
> >
> >> On 3/4/2014 2:00 AM, Martin Thomson wrote:
> >>> On 4 March 2014 03:38, Randell Jesup <randell-ietf@jesup.org> wrote:
> >>>> To take your suggestion:
> >>>> a) applications assume 16KB if ndata wasn't negotiated (per the curr=
ent
> >>>> suggestion)
> >>>> b) implement ndata and EOR support in the browsers, and drop the 16K=
B
> limit
> >>>> when ndata is negotiated.
> >>>> All this needs is "is ndata supported by this association" exposed t=
o
> the
> >>>> app; no sdp changes needed.  (One less dependency!)
> >>> Or, to avoid vestigal crap in the API in the end state, when you
> >>> install the 16K limitation, add an indicator that applications can
> >>> check.  Once you remove the cap, remove the indicator.
> >>>
> >>> if (dataChannel.isNotVeryGood) {
> >>>    chunkMessage(message);
> >>> } else {
> >>>    dataChannel.send(message);
> >>> }
>=20
>=20
> Wait, the reason why we had it be 16K as a default was that current
> impls don't have the property.  So datachannel.isntFixed (ok, better
> names can be found - but that's W3 anyways.

[Raju]=20
NDATA requires support from both the ends. Is browser going to update datac=
hannel.isntFixed/ndata after SCTP is up? Per my reading of PeerConnection o=
nopen() won't be called before SCTP is up (however may be called before DCE=
P, if used, is done). So, app knows NDATA support before sending first mess=
age.

Thanks
Raju


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While a=3Dsctpmap assumes use of DCEP, indication of external negotiation (=
instead of DCEP) also requires subprotocol and it's parameters to be convey=
ed. This has to be done per data channel stream.
Are you looking at more than what is documented in [I-D.ejzak-mmusic-data-c=
hannel-sdpneg]?
Or are you looking for an indication of this (for all streams) directly in =
a=3Dsctpmap?

How about item 3 of the following proposal?
http://www.ietf.org/mail-archive/web/mmusic/current/msg13155.html

-Raju


From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ted Hardie
Sent: Tuesday, March 04, 2014 5:17 AM
To: Paul Kyzivat
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] *MUST* DCEP?

On Tue, Mar 4, 2014 at 11:03 AM, Paul Kyzivat <pkyzivat@alum.mit.edu<mailto=
:pkyzivat@alum.mit.edu>> wrote:
Ted,

(I think this was open issue #8)

You closed discussion on this, saying that this should be specified in some=
 rtcweb "system" document. IIUC, then I disagree.

This is important to others besides rtcweb. We want to use this mechanism, =
both with and without components that are implementing rtcweb. Certainly fo=
r non-browsers.

Help me understand why it wouldn't then be in their systems documents?

Ted



This ultimately comes down to what I can expect after having negotiated (in=
 SDP) the establishment of an SCTP association. When doing so, I use a=3Dsc=
tpmap to negotiate the how the association will be used. If I intend to use=
 data channels and DCEP then I want an assurance that the other end does to=
o. (That is the purpose of the SDP negotiation.)

AFAIK, current when I specify 'a=3Dsctpmap:webrtc-datachannel' (or 'a=3Dsct=
p-datachannel' depending on where I look), that means the other side suppor=
ts data channels *and* guarantees support for DCEP. There is no means to in=
dicate support for data channels *without* DCEP.

I don't see any urgent need to have data channels without DCEP, as long as =
*use* of DCEP is optional.

IMO things would be less confusing if the two documents were merged into on=
e, that defines the *Data Channel Protocol* (that runs over SCTP). This pro=
tocol runs over pairs of SCTP streams, has a small number of messages (open=
, ack, string-payload, binary-payload, close), and encodes them in SCTP usi=
ng SCTP messages with particular payloads and SCTP stream reset. The open a=
nd ack messages would be optional.

        Thanks,
        Paul


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">While a=3Dsctpmap assumes=
 use of DCEP, indication of external negotiation (instead of DCEP) also req=
uires subprotocol and it&#8217;s parameters to be conveyed. This
 has to be done per data channel stream.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Are you looking at more t=
han what is documented in [I-D.ejzak-mmusic-data-channel-sdpneg]?<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Or are you looking for an=
 indication of this (for all streams) directly in a=3Dsctpmap?
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">How about item 3 of the f=
ollowing proposal?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><a href=3D"http://www.iet=
f.org/mail-archive/web/mmusic/current/msg13155.html">http://www.ietf.org/ma=
il-archive/web/mmusic/current/msg13155.html</a><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">-Raju<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb [=
mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Ted Hardie<br>
<b>Sent:</b> Tuesday, March 04, 2014 5:17 AM<br>
<b>To:</b> Paul Kyzivat<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] *MUST* DCEP?<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On Tue, Mar 4, 2014 at 11:03 AM, Paul Kyzivat &lt;<a=
 href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.=
edu</a>&gt; wrote:<o:p></o:p></p>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Ted,<br>
<br>
(I think this was open issue #8)<br>
<br>
You closed discussion on this, saying that this should be specified in some=
 rtcweb &quot;system&quot; document. IIUC, then I disagree.<br>
<br>
This is important to others besides rtcweb. We want to use this mechanism, =
both with and without components that are implementing rtcweb. Certainly fo=
r non-browsers.<o:p></o:p></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Georgia&quot;,&quot=
;serif&quot;">Help me understand why it wouldn't then be in their systems d=
ocuments?<br>
<br>
Ted<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal">This ultimately comes down to what I can expect afte=
r having negotiated (in SDP) the establishment of an SCTP association. When=
 doing so, I use a=3Dsctpmap to negotiate the how the association will be u=
sed. If I intend to use data channels
 and DCEP then I want an assurance that the other end does too. (That is th=
e purpose of the SDP negotiation.)<br>
<br>
AFAIK, current when I specify 'a=3Dsctpmap:webrtc-datachannel' (or 'a=3Dsct=
p-datachannel' depending on where I look), that means the other side suppor=
ts data channels *and* guarantees support for DCEP. There is no means to in=
dicate support for data channels *without*
 DCEP.<br>
<br>
I don't see any urgent need to have data channels without DCEP, as long as =
*use* of DCEP is optional.<br>
<br>
IMO things would be less confusing if the two documents were merged into on=
e, that defines the *Data Channel Protocol* (that runs over SCTP). This pro=
tocol runs over pairs of SCTP streams, has a small number of messages (open=
, ack, string-payload, binary-payload,
 close), and encodes them in SCTP using SCTP messages with particular paylo=
ads and SCTP stream reset. The open and ack messages would be optional.<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; Thanks,<br>
&nbsp; &nbsp; &nbsp; &nbsp; Paul<o:p></o:p></p>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</body>
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From nobody Tue Mar  4 06:53:17 2014
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Date: Tue, 04 Mar 2014 14:50:56 +0000
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] *MUST* DCEP?
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IIUC rtcweb is predicated on the assumption that there is a single 
entity orchestrating the "system" including both of the endpoints of the 
sctp association. You therefore assume that you can a priori assume that 
both ends conform to your "systems document".

A sip endpoint doesn't have that. A major purpose of the signaling and 
the O/A exchange is to find out what we agree to. We won't know in 
advance if the other endpoint follows the CLUE "systems document".

ISTM that you are defining things so that we must use some mechanism to 
negotiate an application-specific agreement between the two ends about 
use of data channels *before* using any data channels.

With the ejzak draft we can use the O/A to negotiate a specific clue 
subprotocol, and then have a reasonable expectation that we can follow 
our own documentation of that protocol to go on.

Using DCEP would be more straightforward, especially for browser based 
endpoints. But if we can't even depend on support of DCEP when we have a 
data channel SCTP association, then DCEP is less useful.

	Thanks,
	Paul

On 3/4/14 11:17 AM, Ted Hardie wrote:
> On Tue, Mar 4, 2014 at 11:03 AM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     Ted,
>
>     (I think this was open issue #8)
>
>     You closed discussion on this, saying that this should be specified
>     in some rtcweb "system" document. IIUC, then I disagree.
>
>     This is important to others besides rtcweb. We want to use this
>     mechanism, both with and without components that are implementing
>     rtcweb. Certainly for non-browsers.
>
>
> Help me understand why it wouldn't then be in their systems documents?
>
> Ted
>
>
>     This ultimately comes down to what I can expect after having
>     negotiated (in SDP) the establishment of an SCTP association. When
>     doing so, I use a=sctpmap to negotiate the how the association will
>     be used. If I intend to use data channels and DCEP then I want an
>     assurance that the other end does too. (That is the purpose of the
>     SDP negotiation.)
>
>     AFAIK, current when I specify 'a=sctpmap:webrtc-datachannel' (or
>     'a=sctp-datachannel' depending on where I look), that means the
>     other side supports data channels *and* guarantees support for DCEP.
>     There is no means to indicate support for data channels *without* DCEP.
>
>     I don't see any urgent need to have data channels without DCEP, as
>     long as *use* of DCEP is optional.
>
>     IMO things would be less confusing if the two documents were merged
>     into one, that defines the *Data Channel Protocol* (that runs over
>     SCTP). This protocol runs over pairs of SCTP streams, has a small
>     number of messages (open, ack, string-payload, binary-payload,
>     close), and encodes them in SCTP using SCTP messages with particular
>     payloads and SCTP stream reset. The open and ack messages would be
>     optional.
>
>              Thanks,
>              Paul
>
>


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Subject: Re: [rtcweb] *MUST* DCEP?
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On 3/4/14 12:08 PM, Makaraju, Maridi Raju (Raju) wrote:
> While a=sctpmap assumes use of DCEP, indication of external negotiation
> (instead of DCEP) also requires subprotocol and it’s parameters to be
> conveyed. This has to be done per data channel stream.
>
> Are you looking at more than what is documented in
> [I-D.ejzak-mmusic-data-channel-sdpneg]?
>
> Or are you looking for an indication of this (for all streams) directly
> in a=sctpmap?

I'm ok as things currently stand, though the documents are confusing.
The 'a=sctpmap:nnn websocket-datachannel' means that DCEP is supported.

My concern was a proposal I think I heard that would make support for 
DCEP to be optional. In that case, IMO it is important that the O/A 
exchange establish whether DCEP is supported or not.

	Thanks,
	Paul

> How about item 3 of the following proposal?
>
> http://www.ietf.org/mail-archive/web/mmusic/current/msg13155.html
>
> -Raju
>
> *From:*rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Ted Hardie
> *Sent:* Tuesday, March 04, 2014 5:17 AM
> *To:* Paul Kyzivat
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] *MUST* DCEP?
>
> On Tue, Mar 4, 2014 at 11:03 AM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     Ted,
>
>     (I think this was open issue #8)
>
>     You closed discussion on this, saying that this should be specified
>     in some rtcweb "system" document. IIUC, then I disagree.
>
>     This is important to others besides rtcweb. We want to use this
>     mechanism, both with and without components that are implementing
>     rtcweb. Certainly for non-browsers.
>
> Help me understand why it wouldn't then be in their systems documents?
>
> Ted
>
>
>
>     This ultimately comes down to what I can expect after having
>     negotiated (in SDP) the establishment of an SCTP association. When
>     doing so, I use a=sctpmap to negotiate the how the association will
>     be used. If I intend to use data channels and DCEP then I want an
>     assurance that the other end does too. (That is the purpose of the
>     SDP negotiation.)
>
>     AFAIK, current when I specify 'a=sctpmap:webrtc-datachannel' (or
>     'a=sctp-datachannel' depending on where I look), that means the
>     other side supports data channels *and* guarantees support for DCEP.
>     There is no means to indicate support for data channels *without* DCEP.
>
>     I don't see any urgent need to have data channels without DCEP, as
>     long as *use* of DCEP is optional.
>
>     IMO things would be less confusing if the two documents were merged
>     into one, that defines the *Data Channel Protocol* (that runs over
>     SCTP). This protocol runs over pairs of SCTP streams, has a small
>     number of messages (open, ack, string-payload, binary-payload,
>     close), and encodes them in SCTP using SCTP messages with particular
>     payloads and SCTP stream reset. The open and ack messages would be
>     optional.
>
>              Thanks,
>              Paul
>


From nobody Tue Mar  4 07:40:31 2014
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Subject: Re: [rtcweb] *MUST* DCEP?
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, March 04, 2014 8:59 AM
> On 3/4/14 12:08 PM, Makaraju, Maridi Raju (Raju) wrote:
> > While a=3Dsctpmap assumes use of DCEP, indication of external negotiati=
on
> > (instead of DCEP) also requires subprotocol and it's parameters to be
> > conveyed. This has to be done per data channel stream.
> >
> > Are you looking at more than what is documented in
> > [I-D.ejzak-mmusic-data-channel-sdpneg]?
> >
> > Or are you looking for an indication of this (for all streams) directly
> > in a=3Dsctpmap?
>=20
> I'm ok as things currently stand, though the documents are confusing.
> The 'a=3Dsctpmap:nnn websocket-datachannel' means that DCEP is supported.
>=20
> My concern was a proposal I think I heard that would make support for
> DCEP to be optional. In that case, IMO it is important that the O/A
> exchange establish whether DCEP is supported or not.

[Raju]
That's the reason why I proposed to add 'protocol' parameter to a=3Ddcmap.
Having this parameter at a=3Dsctmap level may not make that much sense due =
to:
sctpmap is for entire association and protocol defaults to DCEP. To make us=
e of external negotiation, which is per data channel stream, one would need=
 additional a=3D line(s) (e.g. ejzak's draft) to convey subprotocol info. I=
MO adding data channel protocol option to a=3Ddcamp, which is per subprotoc=
ol, makes sense.
[/Raju]=20

Best Regards
Raju


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Subject: Re: [rtcweb] Notes from Data discussion today - reset channel?
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On 3/4/14 11:32 AM, Cullen Jennings wrote:

> Issue #5
> - part of the issue here is there is a protocol on top of SCTP
> - change to text say “reset channel”

I understand what an SCTP stream reset is.

I can guess what "reset channel" means, but AFAIK it isn't defined anywhere.

I guess it may mean sending an Outgoing SSN Reset Request, or an 
Incoming SSN Reset Request, or both.

(If there was a notion of a Data Channel Protocol, then it could have a 
channel reset request, that mapped to the appropriate corresponding SCTP 
actions.)

	Thanks,
	Paul


From nobody Tue Mar  4 07:50:36 2014
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On 04 Mar 2014, at 15:44, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 3/4/14 11:32 AM, Cullen Jennings wrote:
>=20
>> Issue #5
>> - part of the issue here is there is a protocol on top of SCTP
>> - change to text say =93reset channel=94
>=20
> I understand what an SCTP stream reset is.
>=20
> I can guess what "reset channel" means, but AFAIK it isn't defined =
anywhere.
More precise language: Close the channel. This is described in
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-6.7

Best regards
Michael
>=20
> I guess it may mean sending an Outgoing SSN Reset Request, or an =
Incoming SSN Reset Request, or both.
>=20
> (If there was a notion of a Data Channel Protocol, then it could have =
a channel reset request, that mapped to the appropriate corresponding =
SCTP actions.)
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Notes from Data discussion today - DCEP MTI?
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On 3/4/14 11:32 AM, Cullen Jennings wrote:

> Issue #8
> - The transports documents should make the DCEP  be MTI
> - The data channel draft will not require implementation of DCEP

It isn't hugely important to me which document the MTI is in.
(I do care for clarity, but not as far as substance.)

What *is* important to me is what the definition of "webrtc-datachannel" 
is, in the registry defined by draft-ietf-mmusic-sctp-sdp-06. (I can't 
find anything that does the definition.)

If it says DCEP is MTI then I'm ok. If it doesn't then I think there is 
a problem.

	Thanks,
	Paul


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Subject: Re: [rtcweb] Notes from Data discussion today - reset channel?
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On 3/4/14 3:50 PM, Michael Tuexen wrote:
> On 04 Mar 2014, at 15:44, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> On 3/4/14 11:32 AM, Cullen Jennings wrote:
>>
>>> Issue #5
>>> - part of the issue here is there is a protocol on top of SCTP
>>> - change to text say “reset channel”
>>
>> I understand what an SCTP stream reset is.
>>
>> I can guess what "reset channel" means, but AFAIK it isn't defined anywhere.
> More precise language: Close the channel. This is described in
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-6.7
>
> Best regards
> Michael

OK. Close enough I guess. (Close - Reset)

	Sorry,
	Paul


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Subject: Re: [rtcweb] *MUST* DCEP?
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On 3/4/14 3:39 PM, Makaraju, Maridi Raju (Raju) wrote:
>
>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, March 04, 2014 8:59 AM
>> On 3/4/14 12:08 PM, Makaraju, Maridi Raju (Raju) wrote:
>>> While a=sctpmap assumes use of DCEP, indication of external negotiation
>>> (instead of DCEP) also requires subprotocol and it's parameters to be
>>> conveyed. This has to be done per data channel stream.
>>>
>>> Are you looking at more than what is documented in
>>> [I-D.ejzak-mmusic-data-channel-sdpneg]?
>>>
>>> Or are you looking for an indication of this (for all streams) directly
>>> in a=sctpmap?
>>
>> I'm ok as things currently stand, though the documents are confusing.
>> The 'a=sctpmap:nnn websocket-datachannel' means that DCEP is supported.
>>
>> My concern was a proposal I think I heard that would make support for
>> DCEP to be optional. In that case, IMO it is important that the O/A
>> exchange establish whether DCEP is supported or not.
>
> [Raju]
> That's the reason why I proposed to add 'protocol' parameter to a=dcmap.
> Having this parameter at a=sctmap level may not make that much sense due to:
> sctpmap is for entire association and protocol defaults to DCEP. To make use of external negotiation, which is per data channel stream, one would need additional a= line(s) (e.g. ejzak's draft) to convey subprotocol info. IMO adding data channel protocol option to a=dcamp, which is per subprotocol, makes sense.
> [/Raju]

I have no problem with a=dcmap.

It is the 'app' parameter of the sctpmap attribute that I am talking about.

If I want to use DCEP, the dcmap attribute is irrelevant to me. What is 
it that tells me that DCEP is supported. The only datum available that 
could imply that is the 'app' parameter of sctpmap.

Right now, if app is webrtc-datachannel, that implies DCEP is MTI. That 
serves the purpose.

But as of today there was a suggestion that you could have cases where 
there is an SCTP association that supports Data Channels, but not DCEP. 
I don't know how the two would be distinguishable. Possibly by a 
different 'app' value.

A related question is what is the applicability of a=dcmap. Surely it 
doesn't apply for all uses of SCTP - those that don't use data channels 
at all. Do you propose to couple this to particular 'app' values? Or is 
support for this negotiated independently of a=sctpmap?

	Thanks,
	Paul


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On 2014-02-25 17:07, Michael Tuexen wrote:
> 
> On Feb 24, 2014, at 5:32 PM, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> 
>> Hi,
>> (as individual)
> Hi Magnus,
> 
> thank you very much for the detailed review. Please see my comments in-line.
> 
> Best regards
> Michael
>>
>> I have reviewed the WebRTC Data Channel Establishment protocol
>> (draft-ietf-rtcweb-data-protocol-03) and have some comments:
>>
>> 1. Section 4:
>> Shouldn't this section discuss the priority field?
> I added in the list of consistent properties:
> 
> <t>the priority of the data Channel.</t>
> 
> and in the text below that enumeration:
> 
> ??????

Yes, text for this needs to be figured out.

>>
>> 2. Section 4:
>>
>> The method
>>   used to determine which side uses odd or even is based on the
>>   underlying DTLS connection role when used in WebRTC, with the side
>>   acting as the DTLS client using even stream identifiers.
>>
>> Isn't this unnecessary using the vague word of WebRTC instead of simply
>> pointing to the DTLS roles of the established data channel?

> The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and therefore
> you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
> or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
> In that case DTLS is not used and you can not refer to the DTLS role.
> That is why the restriction is used.

Ok, if that concern then you still should be able to write a normative
specification under the condition that it is SCTP over DTLS. If not how
do you determine that? Are suggesting just to hand way or point to a
higher signaling layer.

>>
>> 3. Section 5.1:
>>      DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel provides
>>         a partial reliable in-order bi-directional Communication
>>         channel.  User messages might not be transmitted or
>>         retransmitted after a specified life-time given in milli-
>>         seconds in the Reliability Parameter.  This life-time starts
>>         when providing the user message to the Javascript engine.
>>
>> I think the use of "Javascript engine" in the last sentence is unclear.
>> Can we provide a implementation neutral definition, like "This life-time
>> starts when requesting transmission of the user message."?

> This comment was already made on the list and I changed the text to
> 
> This life-time starts when providing the user message to the protocol stack.
> 
> OK?

Yes.

>>
>> 4. Section 5.1:
>>   Label: Variable Length (sequence of characters)
>>      The name of the channel.  This may be an empty string.
>>
>>   Protocol: Variable Length (sequence of characters)
>>      The protocol for the channel.  If this is an empty string the
>>      protocol us unspecified.  If it is an non-empty string, it
>>      specifies an IANA-registered protocol (see Section 8.4).
>>
>> Both of these fields are strings, shouldn't a particular encoding be
>> specified here? Like UTF-8. Secondly, what values are allowed, the full
>> set of Unicode?

> You are right. Any need for restrictions? We only need to be able to
> transform it to a DomString.
> So I changed it to:
> 
> <t hangText='Label: Variable Length (sequence of characters)'>
> <vspace blankLines='0'/>
> The name of the channel as a UTF-8 encoded string.
> This may be an empty string.</t>
> 
> <t hangText='Protocol: Variable Length (sequence of characters)'>
> <vspace blankLines='0'/>
> The protocol for the channel as a UTF-8 encoded string.
> If this is an empty string the protocol us unspecified.
> If it is an non-empty string, it specifies an IANA-registered protocol
> (see <xref target='iana_protocol'/>).</t>

I guess this is slightly overtaken by event. It have to be aligned with
what the websocket sub-protocol identifier.

>>
>> 5. Section 6:
>> All Data Channel Establishment Protocol messages MUST be sent
>>   requesting ordered delivery and using reliable transmission.
>>
>> I wonder of the use of requesting ordered delivery and using reliable
>> transmission, from an SCTP stream perspective, wouldn't using in both
>> places be appropriate? Or is how object which has been requested to be
>> transmitted unordered interact in SCTP with the ordered ones?

> I'm sorry, I don't understand what you are asking...

Sorry, it really is a language issue. The above sentence first states
"sent 'requesting' ordered" and later "and 'using' reliable".

This inconsistency is what I reacted to.

> 
>> 6. Section 7:
>>
>> I think this section can be beefed up a bit. First make clear that the
>> Data Channel's required usage of DTLS ensures that the message integrity
>> and possible source authentication as well as confidentiality. Then
>> going over any security risks with a malicous peer using this protocol.
>> Can a malicous side screw up the peer using this protocol? What are the
>> implications?

> Just to double check: Aren't the referenced documents the ones which
> discuss all security stuff in one place?

The security document do discuss system level important aspects. But,
from my perspective, details that are very specific to this protocol
should be discussed in this document.

I think this is highly relevant in this document as there are others
that are interested in using it.

>>
>> 7. Section 8.2:
>>
>> This registry reserves 0xff without any motivation, please add one.
> I added:
> 
> The value 0xff has been reserved for future extensibility.
> 

Ok

>>
>> 8. Section 8.3:
>> If I define a channel behavior that allows mixed ordered and unordered,
>> how would I register this?

> You can't. For SCTP properties like ordered/unordered delivery are
> on a per message base. In RTCWeb it was decided to make per channel
> base. So you can't have a data channel supporting a mixtures of
> ordered and unordered messages.

Ok, fine.

>>
>> 9. Section 8.4:
>>
>> I have a suggestion for this registries name, instead of "Protocol
>>   Registry" I propose  "DCEP Protocol Identification Registry"

> The problem I have with this definition is the following:
> Your name suggests that the use of this entity is restricted to
> DCEP. But isn't the protocol a property of a data channel, no matter
> whether negotiated by used DCEP or not?

As we now are going to point at the websocket one this is a mote point.

>>
>> 10. Section 8.4:
>>
>> There is missing any specification of what type of strings I may
>> register. Also, no length limitation, although the protocol allows at
>> maximum of 255 octets of encoded characters.

> I don't understand the limitation to 255 octets. The length field
> is 16 bit, so you can have up to 65535 octets. Therefore I changed:
> <t>A name for the protocol;</t>
> to
> <t>A name for the protocol which length is smaller than 65536 when using
> an UTF-8 encoding;</t>

Overtaken by events, limitations will be the ones of the Websocket
registry.

>>
>> 11. This comments concerns the relation to the Data Channel
>> specification. So my interpretation is that the WebRTC Data channel
>> draft has become both the base for this specification as well as the one
>> that specifies that the DCEP shall be implemented and supported. For
>> WebRTC I don't think this matters much, but if someone likes to re-use
>> the basic Data Channel specification, this structure makes it more
>> difficult and have no need for the bi-directional negotiated parameter
>> settings that the DCEP provides. In that case one have to sub-set the

> I think the data channel draft discusses the data channels independent
> how they are opened. This covers data parameters, user data transmission
> and closing of data channels. The data channel protocol draft discusses
> an in-band protocol for setting up data channels.

I think this was discussed today. And make it clear that DCEP
requirement shouldn't be from the Data Channel spec.

>> data channel specification. I wonder if it would be better to move some
>> normative statements around, so that the chain of implementation
>> requirement goes draft-ietf-rtcweb-transports ->
>> draft-ietf-rtcweb-data-protocol -> draft-ietf-rtcweb-data-channel thus
>> providing a cleaner and more modular build up of the protocols.
> I think you can implement draft-ietf-rtcweb-data-channel without
> draft-ietf-rtcweb-data-protocol, but not vice versa.
> 
> So you don't like the statement:
> 
>    A simple protocol for internal negotiation is specified in
>    [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
> 
> from draft-ietf-rtcweb-data-channel. What would you suggest?

I think the high level requirement on DCEP might be in -rtcweb-transports.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Tue Mar  4 09:41:53 2014
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From nobody Tue Mar  4 09:50:58 2014
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Changed milestone "Send Use Cases document
(draft-ietf-rtcweb-use-cases-and- requirements) to IESG for
publication as Informational", resolved as "Done".

URL: http://datatracker.ietf.org/wg/rtcweb/charter/


From nobody Tue Mar  4 09:56:58 2014
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On 2/25/14 5:07 PM, Michael Tuexen wrote:

>> 2. Section 4:
>>
>> The method
>>    used to determine which side uses odd or even is based on the
>>    underlying DTLS connection role when used in WebRTC, with the side
>>    acting as the DTLS client using even stream identifiers.
>>
>> Isn't this unnecessary using the vague word of WebRTC instead of simply
>> pointing to the DTLS roles of the established data channel?
> The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and therefore
> you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
> or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
> In that case DTLS is not used and you can not refer to the DTLS role.
> That is why the restriction is used.

So data channels could work over SCTP/IP or SCTP/UDP/IP, but in fact 
can't solely because the choice of even/odd role is dependent on DTLS 
connection role?

Couldn't you find a way to choose the even/odd role based solely on the 
SCTP layer and the SDP? Then data channels could be used over those 
other stacks.

	Thanks,
	Paul


From nobody Tue Mar  4 10:28:38 2014
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On 04 Mar 2014, at 17:54, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 2/25/14 5:07 PM, Michael Tuexen wrote:
>=20
>>> 2. Section 4:
>>>=20
>>> The method
>>>   used to determine which side uses odd or even is based on the
>>>   underlying DTLS connection role when used in WebRTC, with the side
>>>   acting as the DTLS client using even stream identifiers.
>>>=20
>>> Isn't this unnecessary using the vague word of WebRTC instead of =
simply
>>> pointing to the DTLS roles of the established data channel?
>> The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and =
therefore
>> you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
>> or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
>> In that case DTLS is not used and you can not refer to the DTLS role.
>> That is why the restriction is used.
>=20
> So data channels could work over SCTP/IP or SCTP/UDP/IP, but in fact =
can't solely because the choice of even/odd role is dependent on DTLS =
connection role?
When not running over DTLS, you have not the security features provided =
by DTLS, of course.
>=20
> Couldn't you find a way to choose the even/odd role based solely on =
the SCTP layer and the SDP? Then data channels could be used over those =
other stacks.
When running over UDP or IP you could define a client and server role =
and use a similar rule.

Best regards
Michael
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)
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On 2014-03-01 16:47, Harald Alvestrand wrote:
> On 02/25/2014 11:23 AM, Magnus Westerlund wrote:
>> On 2014-02-19 19:11, Harald Alvestrand wrote:
>>> On 02/19/2014 09:16 AM, Magnus Westerlund wrote:
>>>> On 2014-02-18 23:58, Harald Alvestrand wrote:
>>>>> I may be a little simple-minded, but if we have a recommendation that
>>>>> receivers MUST be able to receive all packetizations of G.711 and OPUS
>>>>> up to 200 ms per packet, and that receivers should signal this by
>>>>> sending "a=maxptime:200" in their SDP, what situations exist where
>>>>> interoperability is not going to be achieved?
>>>> Interoperability is going to be achieve in the direction towards this
>>>> endpoint. The question is if we achieve interoperability in the
>>>> direction from that endpoint.
>>> So, given that there is nothing in the G.711 specification about which
>>> sample sizes a G.711 receiver has to support - the right approach seems
>>> to be to amend the G.711 MIME registration with this information.
>> Yes, it would for the future. But considering the wide-spread deployment
>> of this payload format I don't see that having any short term effect on
>> the interoperability.
>>
>> Also, the recommendations are actually context dependent. Thus, it is
>> not obvious that a single recommendation is suitable.
>>
>>> This is not a WebRTC issue; it is an absence of specification for the
>>> non-WebRTC devices that use G.711. The RTCWEB specifications can
>>> reasonably be expected to point to existing information about this
>>> issue; it is not reasonable (in my mind) that the RTCWEB WG decide what
>>> these values should be.
>>>
>> I would argue for this consideration based on that this is done in the
>> WebRTC context, not any other context.
> 
> For which value of "this" is that true?
> 
> All other contexts that use G.711 send packets with a certain ptime
> (which determines the number of samples per packet).
> 
> It seems that all other contexts have managed to survive without a
> specification of which ptime it makes sense to send, except for an upper
> bound (maxptime).
> 
> When this situation occurs, usually one of two things is happening:
> 
> - There's an agreement not captured in the standard, which all
> participants follow.
> - The choice doesn't matter for interoperability.
> 
> If the second is the case, I think the RTCWEB specification needs to say
> nothing.
> If the first is the case, the agreement not captured in the standard
> needs to be captured in a standard. But this is not an RTCWEB standard.
> 

I appear to be in the rough on this. I don't expect any real issues with
this as long as implementations do sane things. (I was tempted to write
"the right thing (TM)".

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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> >> 2. Section 4:
> >>
> >> The method
> >>    used to determine which side uses odd or even is based on the
> >>    underlying DTLS connection role when used in WebRTC, with the side
> >>    acting as the DTLS client using even stream identifiers.
> >>
> >> Isn't this unnecessary using the vague word of WebRTC instead of simpl=
y
> >> pointing to the DTLS roles of the established data channel?
> > The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and therefor=
e
> > you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
> > or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
> > In that case DTLS is not used and you can not refer to the DTLS role.
> > That is why the restriction is used.
>=20
> So data channels could work over SCTP/IP or SCTP/UDP/IP, but in fact
> can't solely because the choice of even/odd role is dependent on DTLS
> connection role?
>=20
> Couldn't you find a way to choose the even/odd role based solely on the
> SCTP layer and the SDP? Then data channels could be used over those
> other stacks.
[Raju]=20
+1
a=3Dsetup could be used to determine this, but the final determination is d=
one only at SDP answer (which is same for DTLS).
For DTLS/SCTP case since both DTLS and SCTP layers use the same a=3Dsetup, =
both DTLS and SCTP client/servers roles match.
So, DCEP changing text to indicate role determination be done per SCTP clie=
nt/server role instead of DTLS role may just work and is compatible with ex=
isting text.
[/Raju]

Best Regards
Raju


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> >>> Are you looking at more than what is documented in
> >>> [I-D.ejzak-mmusic-data-channel-sdpneg]?
> >>>
> >>> Or are you looking for an indication of this (for all streams) direct=
ly
> >>> in a=3Dsctpmap?
> >>
> >> I'm ok as things currently stand, though the documents are confusing.
> >> The 'a=3Dsctpmap:nnn websocket-datachannel' means that DCEP is support=
ed.
> >>
> >> My concern was a proposal I think I heard that would make support for
> >> DCEP to be optional. In that case, IMO it is important that the O/A
> >> exchange establish whether DCEP is supported or not.
> >
> > [Raju]
> > That's the reason why I proposed to add 'protocol' parameter to a=3Ddcm=
ap.
> > Having this parameter at a=3Dsctmap level may not make that much sense =
due
> to:
> > sctpmap is for entire association and protocol defaults to DCEP. To mak=
e
> use of external negotiation, which is per data channel stream, one would
> need additional a=3D line(s) (e.g. ejzak's draft) to convey subprotocol i=
nfo.
> IMO adding data channel protocol option to a=3Ddcamp, which is per
> subprotocol, makes sense.
> > [/Raju]
>=20
> I have no problem with a=3Ddcmap.
>=20
> It is the 'app' parameter of the sctpmap attribute that I am talking abou=
t.
>=20
> If I want to use DCEP, the dcmap attribute is irrelevant to me. What is
> it that tells me that DCEP is supported. The only datum available that
> could imply that is the 'app' parameter of sctpmap.
>=20
> Right now, if app is webrtc-datachannel, that implies DCEP is MTI. That
> serves the purpose.
>=20
> But as of today there was a suggestion that you could have cases where
> there is an SCTP association that supports Data Channels, but not DCEP.
> I don't know how the two would be distinguishable. Possibly by a
> different 'app' value.

[Raju]=20
Yes, one option is to define a new 'app'.=20
Another option is to use 'webrtc-datachannel', but then convey your own
external data channel protocol via SDP and use it for all the streams.
When you say "not DCEP", you mean it is "external" or "a different flavor
of internal DCEP"?=20

[/Raju]

>=20
> A related question is what is the applicability of a=3Ddcmap. Surely it
> doesn't apply for all uses of SCTP - those that don't use data channels
> at all. Do you propose to couple this to particular 'app' values? Or is
> support for this negotiated independently of a=3Dsctpmap?
[Raju]=20
Correct. dcmap applies to SCTP data channels only. dcmap's 'protocol' param=
eter
allows you to override webrtc default DCEP per data channel stream.
When you are using dcmap it allows per data channel "interal(DCEP)" or=20
"external(SDP based)" determination using 'protocol' parameter.
If there is a need we can allow the 'protocol' value to be extended (per IA=
NA, may be?!
):
"internal-DCEP"
"external-SDP-based" (ejzak's draft)
"new-enhanced-internal-DCEP"
"pre-defined-stream-id-use-arrangement"

[/Raju]

Best Regards
Raju


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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 5 Mar 2014 09:27:15 +0000
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Subject: [rtcweb] Issue with new Trickle ICE text
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--047d7bb04dc0bb468e04f3d8a262
Content-Type: text/plain; charset=ISO-8859-1

S 3.4.2 of JSEP now reads:

    JSEP applications typically inform the browser to begin ICE gathering
    via the information supplied to setLocalDescription, as this is where
    the app specifies the number of media streams to gather candidates
    for.  However, to accelerate cases where the browser knows the number
    of media streams to use ahead of time, the application MAY ask the
    browser to gather a pool of potential ICE candidates to help ensure
    rapid media setup.  When setLocalDescription is eventually called,
    and the browser goes to gather the needed ICE candidates, it can
    start by checking if any candidates are available in the pool.

The issue here is that this basically mandates trickle ICE. If the
pool is smaller than the number of independent ICE streams needed,
then there is no way for you to "start gathering" (since this text
says it happens with sLD) and therefore you cannot supply all of the
candidates in the offer.

If that's not the intention to require (and I don't believe we agreed
to this semantic), then this needs to be rewritten.

-Ekr

--047d7bb04dc0bb468e04f3d8a262
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr">S 3.4.2 of JSEP now reads:<div><br></div><div><div>=A0 =A0=
 JSEP applications typically inform the browser to begin ICE gathering<span=
 class=3D"" style=3D"white-space:pre">	</span></div><div>=A0 =A0 via the in=
formation supplied to setLocalDescription, as this is where<span class=3D""=
 style=3D"white-space:pre">	</span></div>

<div>=A0 =A0 the app specifies the number of media streams to gather candid=
ates<span class=3D"" style=3D"white-space:pre">	</span></div><div>=A0 =A0 f=
or. =A0However, to accelerate cases where the browser knows the number<span=
 class=3D"" style=3D"white-space:pre">	</span></div>

<div>=A0 =A0 of media streams to use ahead of time, the application MAY ask=
 the<span class=3D"" style=3D"white-space:pre">	</span></div><div>=A0 =A0 b=
rowser to gather a pool of potential ICE candidates to help ensure<span cla=
ss=3D"" style=3D"white-space:pre">	</span></div>

<div>=A0 =A0 rapid media setup. =A0When setLocalDescription is eventually c=
alled,<span class=3D"" style=3D"white-space:pre">	</span></div><div>=A0 =A0=
 and the browser goes to gather the needed ICE candidates, it can<span clas=
s=3D"" style=3D"white-space:pre">	</span></div>

<div>=A0 =A0 start by checking if any candidates are available in the pool.=
</div><div><br></div><div>The issue here is that this basically mandates tr=
ickle ICE. If the</div><div>pool is smaller than the number of independent =
ICE streams needed,</div>

<div>then there is no way for you to &quot;start gathering&quot; (since thi=
s text</div><div>says it happens with sLD) and therefore you cannot supply =
all of the</div><div>candidates in the offer.</div><div><br></div><div>

If that&#39;s not the intention to require (and I don&#39;t believe we agre=
ed</div><div>to this semantic), then this needs to be rewritten.</div></div=
><div><br></div><div>-Ekr</div><div><br></div></div>

--047d7bb04dc0bb468e04f3d8a262--


From nobody Wed Mar  5 02:04:39 2014
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I haven't thought about this in detail yet, but I wanted to ensure
that this issue is tracked.  I think that our chairs might be able to
create an issue in some tracker.  Can I request that they do that for
this issue?


From nobody Wed Mar  5 02:47:43 2014
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On 3/4/14 6:28 PM, Michael Tuexen wrote:
> On 04 Mar 2014, at 17:54, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> On 2/25/14 5:07 PM, Michael Tuexen wrote:
>>
>>>> 2. Section 4:
>>>>
>>>> The method
>>>>    used to determine which side uses odd or even is based on the
>>>>    underlying DTLS connection role when used in WebRTC, with the side
>>>>    acting as the DTLS client using even stream identifiers.
>>>>
>>>> Isn't this unnecessary using the vague word of WebRTC instead of simply
>>>> pointing to the DTLS roles of the established data channel?
>>> The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and therefore
>>> you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
>>> or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
>>> In that case DTLS is not used and you can not refer to the DTLS role.
>>> That is why the restriction is used.
>>
>> So data channels could work over SCTP/IP or SCTP/UDP/IP, but in fact can't solely because the choice of even/odd role is dependent on DTLS connection role?
> When not running over DTLS, you have not the security features provided by DTLS, of course.

Of course.

One might argue that in the new world we shouldn't even define proto 
stacks that don't include security. But for the moment we are defining 
those protos. I see no reason why data channels should work over sctp, 
regardless of what is below that.

>> Couldn't you find a way to choose the even/odd role based solely on the SCTP layer and the SDP? Then data channels could be used over those other stacks.
> When running over UDP or IP you could define a client and server role and use a similar rule.

Defining the rule differently for each proto stack is *possible* but 
unpleasant. IMO it would be far better to have a rule that works without 
regard for what is under sctp.

	Thanks,
	Paul

> Best regards
> Michael
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>


From nobody Wed Mar  5 04:32:57 2014
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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Subject: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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I am opposed to mandating that Client-to-Mixer Audio Level RTP header is RE=
QUIRED to be encrypted. It means that we can not really build conferencing =
system where the mixers do not have access to decrypt the media.

I would like to propose that the JS application can control  if the header =
is encrypted or not. =20

I am aware of the paper that suggests these audio levels can reveal the con=
tents of the conversation. There is some element of truth to this. There is=
 also some element of truth to the point that if this was true, speech reco=
gnition system would work better than they do. Encrypting this or not is a =
risk that the application using the header extension needs to evaluate, and=
 WebRTC system needs to provide the applications with a control surfaces th=
at allows them to use this in the way the applications desires.

I will note that an normal application that used this header could decide i=
f it was encrypted or not, what is different in RTCWeb is that we are build=
ing a platform and my view is that this platform should allow the applicati=
ons build on the platform to decide the tradeoffs of risk for encrypting th=
is or not.=20







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On 2014-03-05 10:04, Martin Thomson wrote:
> I haven't thought about this in detail yet, but I wanted to ensure
> that this issue is tracked.  I think that our chairs might be able to
> create an issue in some tracker.  Can I request that they do that for
> this issue?
> 

I will currently not stuff anything into our tracker.

However, I will give a bit of context from what the rtp-usage says about
this.

On the receiver side, we have a clear requirement on handling multiple
CNAMEs in Section 11.

   A WebRTC endpoint MUST support receiving multiple MediaStreamTracks,
   where each of different MediaStreamTracks (and their sets of
   associated packet streams) uses different CNAMEs.  However,
   MediaStreamTracks that are received with different CNAMEs have no
   defined synchronisation.

On the sending side the text says the following:

   The same MediaStreamTrack can also be included in multiple
   MediaStreams, thus multiple sets of MediaStreams can implicitly need
   to use the same synchronisation base.  To ensure that this works in
   all cases, and don't forces a endpoint to change synchronisation base
   and CNAME in the middle of a ongoing delivery of any packet streams,
   which would cause media disruption; all MediaStreamTracks and their
   associated SSRCs originating from the same endpoint MUST be sent
   using the same CNAME within one RTCPeerConnection as well as across
   all RTCPeerConnections part of the same communication session
   context, which for a browser are a single origin.


Martin, you talked about linking in this context. I wonder if there
really are an issue with linking as this is all in the same
communication context. Can you please make clear your concerns?

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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On 5 March 2014 12:44, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> Martin, you talked about linking in this context. I wonder if there
> really are an issue with linking as this is all in the same
> communication context. Can you please make clear your concerns?

I'm mostly concerned about being able to communicate with multiple
people from the same page without revealing a linkage between sessions
based on cert or CNAME.  It may be that we need API hooks to control
this.

As I said, I haven't thought it through fully.


From nobody Wed Mar  5 05:59:16 2014
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On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy) <fluffy@cisco.com> =
wrote:

>=20
> I am opposed to mandating that Client-to-Mixer Audio Level RTP header =
is REQUIRED to be encrypted. It means that we can not really build =
conferencing system where the mixers do not have access to decrypt the =
media.

I have to ask:
Does that matter enough to weaken user security for everyone else?=20


>=20
> I would like to propose that the JS application can control  if the =
header is encrypted or not. =20

I am deeply uncomfortable about that. I am not a good enough =
cryptographer to know, but my instinct
is that exposing the fact that (for example) most of the top bits of the =
ulaw data are zeros in this packet
must be a risk.

I=92m pretty confident that one could detect the pitch of the speaker =
and probably fingerprint that down
to individuals or languages with enough data.

>=20
> I am aware of the paper that suggests these audio levels can reveal =
the contents of the conversation. There is some element of truth to =
this. There is also some element of truth to the point that if this was =
true, speech recognition system would work better than they do. =
Encrypting this or not is a risk that the application using the header =
extension needs to evaluate, and WebRTC system needs to provide the =
applications with a control surfaces that allows them to use this in the =
way the applications desires.

I am also uncomfortable about the number of little ad-hoc control =
surfaces that we seem to be mandating
without an overall design.

My take is that we should push this out to 2.0 when hopefully we will =
have a designed control surface in place.

>=20
> I will note that an normal application that used this header could =
decide if it was encrypted or not, what is different in RTCWeb is that =
we are building a platform and my view is that this platform should =
allow the applications build on the platform to decide the tradeoffs of =
risk for encrypting this or not.=20

That isn=92t the line we have taken in any other encryption discussion.

Tim.


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: tim panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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Well if people don=92t want to allow the application to choose to encrypt t=
his header or not, I will be arguing for MUST NOT encrypt the RTP headers -=
 all of them.=20

A recurring theme at IETF is that people trying to protect E2E privacy, do =
things that result in the real world having the only way they can build app=
lications is to build middle boxes that pretend to to be an end to the thin=
gs on both sides. This means that the middle boxes end up with 100% ability=
 to change everything and we loose all the protection we hoped to get with =
E2E security.=20


On Mar 5, 2014, at 1:59 PM, tim panton <tim@phonefromhere.com> wrote:

>=20
> On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy) <fluffy@cisco.com> wrot=
e:
>=20
>>=20
>> I am opposed to mandating that Client-to-Mixer Audio Level RTP header is=
 REQUIRED to be encrypted. It means that we can not really build conferenci=
ng system where the mixers do not have access to decrypt the media.
>=20
> I have to ask:
> Does that matter enough to weaken user security for everyone else?=20
>=20
>=20
>>=20
>> I would like to propose that the JS application can control  if the head=
er is encrypted or not. =20
>=20
> I am deeply uncomfortable about that. I am not a good enough cryptographe=
r to know, but my instinct
> is that exposing the fact that (for example) most of the top bits of the =
ulaw data are zeros in this packet
> must be a risk.
>=20
> I=92m pretty confident that one could detect the pitch of the speaker and=
 probably fingerprint that down
> to individuals or languages with enough data.
>=20
>>=20
>> I am aware of the paper that suggests these audio levels can reveal the =
contents of the conversation. There is some element of truth to this. There=
 is also some element of truth to the point that if this was true, speech r=
ecognition system would work better than they do. Encrypting this or not is=
 a risk that the application using the header extension needs to evaluate, =
and WebRTC system needs to provide the applications with a control surfaces=
 that allows them to use this in the way the applications desires.
>=20
> I am also uncomfortable about the number of little ad-hoc control surface=
s that we seem to be mandating
> without an overall design.
>=20
> My take is that we should push this out to 2.0 when hopefully we will hav=
e a designed control surface in place.
>=20
>>=20
>> I will note that an normal application that used this header could decid=
e if it was encrypted or not, what is different in RTCWeb is that we are bu=
ilding a platform and my view is that this platform should allow the applic=
ations build on the platform to decide the tradeoffs of risk for encrypting=
 this or not.=20
>=20
> That isn=92t the line we have taken in any other encryption discussion.
>=20
> Tim.
>=20


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Date: Wed, 5 Mar 2014 14:25:42 +0000
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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--047d7b33cddce9149404f3dccc44
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So there are three things here:
1) MUST the implementation offer encrypted header extensions? (i.e.
mandatory-to-implement)
2) MUST the implementation use encrypted header extensions? (i.e.
mandatory-to-use)
3) MUST the implementation expose an API to control this? (i.e. no SDP
munging needed)

I think we want yes for #1, no for #2, and #3 is potentially interesting
but out of scope for 1.0.
That gives encrypted headers for audio on by default, but remote parties
can negotiate this off using RFC 6904 mechanisms.



On Wed, Mar 5, 2014 at 2:05 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> Well if people don=E2=80=99t want to allow the application to choose to e=
ncrypt
> this header or not, I will be arguing for MUST NOT encrypt the RTP header=
s
> - all of them.
>
> A recurring theme at IETF is that people trying to protect E2E privacy, d=
o
> things that result in the real world having the only way they can build
> applications is to build middle boxes that pretend to to be an end to the
> things on both sides. This means that the middle boxes end up with 100%
> ability to change everything and we loose all the protection we hoped to
> get with E2E security.
>
>
> On Mar 5, 2014, at 1:59 PM, tim panton <tim@phonefromhere.com> wrote:
>
> >
> > On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy) <fluffy@cisco.com>
> wrote:
> >
> >>
> >> I am opposed to mandating that Client-to-Mixer Audio Level RTP header
> is REQUIRED to be encrypted. It means that we can not really build
> conferencing system where the mixers do not have access to decrypt the
> media.
> >
> > I have to ask:
> > Does that matter enough to weaken user security for everyone else?
> >
> >
> >>
> >> I would like to propose that the JS application can control  if the
> header is encrypted or not.
> >
> > I am deeply uncomfortable about that. I am not a good enough
> cryptographer to know, but my instinct
> > is that exposing the fact that (for example) most of the top bits of th=
e
> ulaw data are zeros in this packet
> > must be a risk.
> >
> > I=E2=80=99m pretty confident that one could detect the pitch of the spe=
aker and
> probably fingerprint that down
> > to individuals or languages with enough data.
> >
> >>
> >> I am aware of the paper that suggests these audio levels can reveal th=
e
> contents of the conversation. There is some element of truth to this. The=
re
> is also some element of truth to the point that if this was true, speech
> recognition system would work better than they do. Encrypting this or not
> is a risk that the application using the header extension needs to
> evaluate, and WebRTC system needs to provide the applications with a
> control surfaces that allows them to use this in the way the applications
> desires.
> >
> > I am also uncomfortable about the number of little ad-hoc control
> surfaces that we seem to be mandating
> > without an overall design.
> >
> > My take is that we should push this out to 2.0 when hopefully we will
> have a designed control surface in place.
> >
> >>
> >> I will note that an normal application that used this header could
> decide if it was encrypted or not, what is different in RTCWeb is that we
> are building a platform and my view is that this platform should allow th=
e
> applications build on the platform to decide the tradeoffs of risk for
> encrypting this or not.
> >
> > That isn=E2=80=99t the line we have taken in any other encryption discu=
ssion.
> >
> > Tim.
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b33cddce9149404f3dccc44
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<div dir=3D"ltr">So there are three things here:<div>1) MUST the implementa=
tion offer encrypted header extensions? (i.e. mandatory-to-implement)</div>=
<div>2) MUST the implementation use encrypted header extensions? (i.e. mand=
atory-to-use)</div>

<div>3) MUST the implementation expose an API to control this? (i.e. no SDP=
 munging needed)<br></div><div><br></div><div>I think we want yes for #1, n=
o for #2, and #3 is potentially interesting but out of scope for 1.0.</div>

<div>That gives encrypted headers for audio on by default, but remote parti=
es can negotiate this off using RFC 6904 mechanisms.</div><div><br></div></=
div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, M=
ar 5, 2014 at 2:05 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</=
span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
Well if people don=E2=80=99t want to allow the application to choose to enc=
rypt this header or not, I will be arguing for MUST NOT encrypt the RTP hea=
ders - all of them.<br>
<br>
A recurring theme at IETF is that people trying to protect E2E privacy, do =
things that result in the real world having the only way they can build app=
lications is to build middle boxes that pretend to to be an end to the thin=
gs on both sides. This means that the middle boxes end up with 100% ability=
 to change everything and we loose all the protection we hoped to get with =
E2E security.<br>


<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On Mar 5, 2014, at 1:59 PM, tim panton &lt;<a href=3D"mailto:tim@phonefromh=
ere.com">tim@phonefromhere.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt; On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy) &lt;<a href=3D"mailt=
o:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I am opposed to mandating that Client-to-Mixer Audio Level RTP hea=
der is REQUIRED to be encrypted. It means that we can not really build conf=
erencing system where the mixers do not have access to decrypt the media.<b=
r>


&gt;<br>
&gt; I have to ask:<br>
&gt; Does that matter enough to weaken user security for everyone else?<br>
&gt;<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I would like to propose that the JS application can control =C2=A0=
if the header is encrypted or not.<br>
&gt;<br>
&gt; I am deeply uncomfortable about that. I am not a good enough cryptogra=
pher to know, but my instinct<br>
&gt; is that exposing the fact that (for example) most of the top bits of t=
he ulaw data are zeros in this packet<br>
&gt; must be a risk.<br>
&gt;<br>
&gt; I=E2=80=99m pretty confident that one could detect the pitch of the sp=
eaker and probably fingerprint that down<br>
&gt; to individuals or languages with enough data.<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I am aware of the paper that suggests these audio levels can revea=
l the contents of the conversation. There is some element of truth to this.=
 There is also some element of truth to the point that if this was true, sp=
eech recognition system would work better than they do. Encrypting this or =
not is a risk that the application using the header extension needs to eval=
uate, and WebRTC system needs to provide the applications with a control su=
rfaces that allows them to use this in the way the applications desires.<br=
>


&gt;<br>
&gt; I am also uncomfortable about the number of little ad-hoc control surf=
aces that we seem to be mandating<br>
&gt; without an overall design.<br>
&gt;<br>
&gt; My take is that we should push this out to 2.0 when hopefully we will =
have a designed control surface in place.<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I will note that an normal application that used this header could=
 decide if it was encrypted or not, what is different in RTCWeb is that we =
are building a platform and my view is that this platform should allow the =
applications build on the platform to decide the tradeoffs of risk for encr=
ypting this or not.<br>


&gt;<br>
&gt; That isn=E2=80=99t the line we have taken in any other encryption disc=
ussion.<br>
&gt;<br>
&gt; Tim.<br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--047d7b33cddce9149404f3dccc44--


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On 5 Mar 2014, at 14:05, Cullen Jennings (fluffy) <fluffy@cisco.com> =
wrote:

>=20
> Well if people don=92t want to allow the application to choose to =
encrypt this header or not, I will be arguing for MUST NOT encrypt the =
RTP headers - all of them.=20
>=20
> A recurring theme at IETF is that people trying to protect E2E =
privacy, do things that result in the real world having the only way =
they can build applications is to build middle boxes that pretend to to =
be an end to the things on both sides. This means that the middle boxes =
end up with 100% ability to change everything and we loose all the =
protection we hoped to get with E2E security.=20

To me this smacks of special pleading, it would be very useful (for =
support reasons) if the javascript (optionally)
got access to the names and characteristics of the input devices. We =
decided as a group that a) the risks of fingerprinting out weighed the =
benefits b) that we didn=92t need a control surface to expose an option =
for this.

I feel this is an equivalent case, but we seem to be looking at making a =
different decision.

I=92m confident that the desired result could be achieved securely with =
a combination of webAudio sending appropriate volume data down a data =
channel set up to the middle box for this purpose.
I realise that would require some extensive rework in the middle box, =
but thats what happens when you connect a middle box to an ostensibly =
peer-to-peer system.

Tim.



>=20
>=20
> On Mar 5, 2014, at 1:59 PM, tim panton <tim@phonefromhere.com> wrote:
>=20
>>=20
>> On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy) <fluffy@cisco.com> =
wrote:
>>=20
>>>=20
>>> I am opposed to mandating that Client-to-Mixer Audio Level RTP =
header is REQUIRED to be encrypted. It means that we can not really =
build conferencing system where the mixers do not have access to decrypt =
the media.
>>=20
>> I have to ask:
>> Does that matter enough to weaken user security for everyone else?=20
>>=20
>>=20
>>>=20
>>> I would like to propose that the JS application can control  if the =
header is encrypted or not. =20
>>=20
>> I am deeply uncomfortable about that. I am not a good enough =
cryptographer to know, but my instinct
>> is that exposing the fact that (for example) most of the top bits of =
the ulaw data are zeros in this packet
>> must be a risk.
>>=20
>> I=92m pretty confident that one could detect the pitch of the speaker =
and probably fingerprint that down
>> to individuals or languages with enough data.
>>=20
>>>=20
>>> I am aware of the paper that suggests these audio levels can reveal =
the contents of the conversation. There is some element of truth to =
this. There is also some element of truth to the point that if this was =
true, speech recognition system would work better than they do. =
Encrypting this or not is a risk that the application using the header =
extension needs to evaluate, and WebRTC system needs to provide the =
applications with a control surfaces that allows them to use this in the =
way the applications desires.
>>=20
>> I am also uncomfortable about the number of little ad-hoc control =
surfaces that we seem to be mandating
>> without an overall design.
>>=20
>> My take is that we should push this out to 2.0 when hopefully we will =
have a designed control surface in place.
>>=20
>>>=20
>>> I will note that an normal application that used this header could =
decide if it was encrypted or not, what is different in RTCWeb is that =
we are building a platform and my view is that this platform should =
allow the applications build on the platform to decide the tradeoffs of =
risk for encrypting this or not.=20
>>=20
>> That isn=92t the line we have taken in any other encryption =
discussion.
>>=20
>> Tim.
>>=20
>=20


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From: Justin Uberti <juberti@google.com>
Date: Wed, 5 Mar 2014 14:30:42 +0000
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To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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--047d7bd6ab54c34f2c04f3dcde1c
Content-Type: text/plain; charset=UTF-8

I think the key thing is that it's not clear when creating a new
PeerConnection whether it should make new CNAMEs or use existing ones.

Compare the two scenarios below:

#1: audio and video over 2 PCs to the same destination
pc1 = new RTCPeerConnection();
pc2 = new RTCPeerConnection();
pc1.addStream(audioStream);
pc2.addStream(videoStream);
doCall(pc1);
doCall(pc2);
// cnames in pc1 and pc2 should be the same

#2: audio call over PC1 to destination 1, followed by audio call over PC2
to destination 2
pc1 = new RTCPeerConnection();
pc1.addStream(audioStream);
doCall(pc1);
pc2 = new RTCPeerConnection();
pc2.addStream(audioStream);
doCall(pc2);
// cnames in pc1 and pc2 should be different

I am inclined to make CNAMEs per-PeerConnection (i.e. enforce scenario #2
behavior) for 1.0, as it has a smaller downside.



On Wed, Mar 5, 2014 at 1:37 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 5 March 2014 12:44, Magnus Westerlund <magnus.westerlund@ericsson.com>
> wrote:
> > Martin, you talked about linking in this context. I wonder if there
> > really are an issue with linking as this is all in the same
> > communication context. Can you please make clear your concerns?
>
> I'm mostly concerned about being able to communicate with multiple
> people from the same page without revealing a linkage between sessions
> based on cert or CNAME.  It may be that we need API hooks to control
> this.
>
> As I said, I haven't thought it through fully.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7bd6ab54c34f2c04f3dcde1c
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">I think the key thing is that it&#39;s not clear when crea=
ting a new PeerConnection whether it should make new CNAMEs or use existing=
 ones.<div><br></div><div>Compare the two scenarios below:</div><div><br></=
div>

<div>#1: audio and video over 2 PCs to the same destination</div><div>pc1 =
=3D new RTCPeerConnection();</div><div>pc2 =3D new RTCPeerConnection();</di=
v><div>pc1.addStream(audioStream);</div><div>pc2.addStream(videoStream);</d=
iv>

<div>doCall(pc1);</div><div>doCall(pc2);</div><div>// cnames in pc1 and pc2=
 should be the same</div><div><br></div><div>#2: audio call over PC1 to des=
tination 1, followed by audio call over PC2 to destination 2</div><div>

pc1 =3D new RTCPeerConnection();</div><div>pc1.addStream(audioStream);</div=
><div>doCall(pc1);</div><div><div>pc2 =3D new RTCPeerConnection();</div><di=
v>pc2.addStream(audioStream);</div><div>doCall(pc2);</div></div><div>// cna=
mes in pc1 and pc2 should be different</div>

<div><br></div><div>I am inclined to make CNAMEs per-PeerConnection (i.e. e=
nforce scenario #2 behavior) for 1.0, as it has a smaller downside.</div><d=
iv><br></div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_q=
uote">

On Wed, Mar 5, 2014 at 1:37 PM, Martin Thomson <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmai=
l.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div class=3D"">On 5 March 2014 12:44, Magnus Westerlund &lt;<a href=3D"mai=
lto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a>&gt; =
wrote:<br>
&gt; Martin, you talked about linking in this context. I wonder if there<br=
>
&gt; really are an issue with linking as this is all in the same<br>
&gt; communication context. Can you please make clear your concerns?<br>
<br>
</div>I&#39;m mostly concerned about being able to communicate with multipl=
e<br>
people from the same page without revealing a linkage between sessions<br>
based on cert or CNAME. =C2=A0It may be that we need API hooks to control<b=
r>
this.<br>
<br>
As I said, I haven&#39;t thought it through fully.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--047d7bd6ab54c34f2c04f3dcde1c--


From nobody Wed Mar  5 06:33:35 2014
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On 5 Mar 2014, at 14:25, Justin Uberti <juberti@google.com> wrote:

> So there are three things here:
> 1) MUST the implementation offer encrypted header extensions? (i.e. =
mandatory-to-implement)
> 2) MUST the implementation use encrypted header extensions? (i.e. =
mandatory-to-use)
> 3) MUST the implementation expose an API to control this? (i.e. no SDP =
munging needed)
>=20
> I think we want yes for #1, no for #2, and #3 is potentially =
interesting but out of scope for 1.0.
> That gives encrypted headers for audio on by default, but remote =
parties can negotiate this off using RFC 6904 mechanisms.
>=20

In order to agree with that I=92d need to be persuaded that there isn=92t =
a known-plaintext attack risk there
(e.g. detecting a muted mic therefore knowing the ulaw content of a =
packet), since
any weakening  in DTLS also impacts the data channel and video channels =
- assuming bundle.

Like I say, I=92m not enough of a cryptographer to know.

T.



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: tim panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On Mar 5, 2014, at 2:33 PM, tim panton <tim@phonefromhere.com> wrote:

>=20
> On 5 Mar 2014, at 14:25, Justin Uberti <juberti@google.com> wrote:
>=20
>> So there are three things here:
>> 1) MUST the implementation offer encrypted header extensions? (i.e. mand=
atory-to-implement)
>> 2) MUST the implementation use encrypted header extensions? (i.e. mandat=
ory-to-use)
>> 3) MUST the implementation expose an API to control this? (i.e. no SDP m=
unging needed)
>>=20
>> I think we want yes for #1, no for #2, and #3 is potentially interesting=
 but out of scope for 1.0.
>> That gives encrypted headers for audio on by default, but remote parties=
 can negotiate this off using RFC 6904 mechanisms.
>>=20
>=20
> In order to agree with that I=92d need to be persuaded that there isn=92t=
 a known-plaintext attack risk there
> (e.g. detecting a muted mic therefore knowing the ulaw content of a packe=
t), since
> any weakening  in DTLS also impacts the data channel and video channels -=
 assuming bundle.
>=20
> Like I say, I=92m not enough of a cryptographer to know.

No - it does not weaken DTLS to know the plaintext (and if it did, DTLS wou=
ld have series security problems )=


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From: Justin Uberti <juberti@google.com>
Date: Wed, 5 Mar 2014 14:40:01 +0000
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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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--089e0111bce0147fc204f3dd009a
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Agree with Cullen (and FWIW the SRTP encryption uses keys vended from the
DTLS master secret, so I think even if you are able to crack the SRTP
crypto it would not be trivial to go backwards up to the DTLS key material)


On Wed, Mar 5, 2014 at 2:35 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> On Mar 5, 2014, at 2:33 PM, tim panton <tim@phonefromhere.com> wrote:
>
> >
> > On 5 Mar 2014, at 14:25, Justin Uberti <juberti@google.com> wrote:
> >
> >> So there are three things here:
> >> 1) MUST the implementation offer encrypted header extensions? (i.e.
> mandatory-to-implement)
> >> 2) MUST the implementation use encrypted header extensions? (i.e.
> mandatory-to-use)
> >> 3) MUST the implementation expose an API to control this? (i.e. no SDP
> munging needed)
> >>
> >> I think we want yes for #1, no for #2, and #3 is potentially
> interesting but out of scope for 1.0.
> >> That gives encrypted headers for audio on by default, but remote
> parties can negotiate this off using RFC 6904 mechanisms.
> >>
> >
> > In order to agree with that I=E2=80=99d need to be persuaded that there=
 isn=E2=80=99t a
> known-plaintext attack risk there
> > (e.g. detecting a muted mic therefore knowing the ulaw content of a
> packet), since
> > any weakening  in DTLS also impacts the data channel and video channels
> - assuming bundle.
> >
> > Like I say, I=E2=80=99m not enough of a cryptographer to know.
>
> No - it does not weaken DTLS to know the plaintext (and if it did, DTLS
> would have series security problems )

--089e0111bce0147fc204f3dd009a
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<div dir=3D"ltr">Agree with Cullen (and FWIW the SRTP encryption uses keys =
vended from the DTLS master secret, so I think even if you are able to crac=
k the SRTP crypto it would not be trivial to go backwards up to the DTLS ke=
y material)</div>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, Mar 5=
, 2014 at 2:35 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt;<a href=
=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</sp=
an> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"HOEnZb"><div class=3D"h5"><br>
On Mar 5, 2014, at 2:33 PM, tim panton &lt;<a href=3D"mailto:tim@phonefromh=
ere.com">tim@phonefromhere.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt; On 5 Mar 2014, at 14:25, Justin Uberti &lt;<a href=3D"mailto:juberti@g=
oogle.com">juberti@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; So there are three things here:<br>
&gt;&gt; 1) MUST the implementation offer encrypted header extensions? (i.e=
. mandatory-to-implement)<br>
&gt;&gt; 2) MUST the implementation use encrypted header extensions? (i.e. =
mandatory-to-use)<br>
&gt;&gt; 3) MUST the implementation expose an API to control this? (i.e. no=
 SDP munging needed)<br>
&gt;&gt;<br>
&gt;&gt; I think we want yes for #1, no for #2, and #3 is potentially inter=
esting but out of scope for 1.0.<br>
&gt;&gt; That gives encrypted headers for audio on by default, but remote p=
arties can negotiate this off using RFC 6904 mechanisms.<br>
&gt;&gt;<br>
&gt;<br>
&gt; In order to agree with that I=E2=80=99d need to be persuaded that ther=
e isn=E2=80=99t a known-plaintext attack risk there<br>
&gt; (e.g. detecting a muted mic therefore knowing the ulaw content of a pa=
cket), since<br>
&gt; any weakening =C2=A0in DTLS also impacts the data channel and video ch=
annels - assuming bundle.<br>
&gt;<br>
&gt; Like I say, I=E2=80=99m not enough of a cryptographer to know.<br>
<br>
</div></div>No - it does not weaken DTLS to know the plaintext (and if it d=
id, DTLS would have series security problems )</blockquote></div><br></div>

--089e0111bce0147fc204f3dd009a--


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To: Justin Uberti <juberti@google.com>
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On Mar 5, 2014, at 2:25 PM, Justin Uberti <juberti@google.com> wrote:

> So there are three things here:
> 1) MUST the implementation offer encrypted header extensions? (i.e. manda=
tory-to-implement)
I=92m fine with yes

> 2) MUST the implementation use encrypted header extensions? (i.e. mandato=
ry-to-use)
I think this needs to be No. Never mind my current use case - if the far en=
d does not negotiate RFC 6094 with the browser, the media and headers shoul=
d not fail.=20

> 3) MUST the implementation expose an API to control this? (i.e. no SDP mu=
nging needed)
I prefer control surfaces to SDP munging but as you say, I could live with =
SDP munging in 1.0=20

>=20
> I think we want yes for #1, no for #2, and #3 is potentially interesting =
but out of scope for 1.0.
> That gives encrypted headers for audio on by default, but remote parties =
can negotiate this off using RFC 6904 mechanisms.
>=20


seems workable=20


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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On 5 Mar 2014, at 14:40, Justin Uberti <juberti@google.com> wrote:

> Agree with Cullen (and FWIW the SRTP encryption uses keys vended from =
the DTLS master secret, so I think even if you are able to crack the =
SRTP crypto it would not be trivial to go backwards up to the DTLS key =
material)
>=20
>=20

Yep, agreed, my mistake.
T.


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>From the transports draft, S 3.4:


   In order to deal with situations where one party is on an IPv4
   network and the other party is on an IPv6 network, TURN extensions
   for IPv6 [RFC6156 <http://tools.ietf.org/html/rfc6156>] MUST be supported.


I think this needs clarification. Does this mean that we should
allocate v6 candidates from all v4 interfaces (for compat with v6-only
endpoints), as well as allocate v4 candidates from all v6 interfaces
(for compat with v4-only endpoints), and if so, which path (v6-to-TURN
or v6-from-TURN) should be prioritized?


Note also that these v4->v6 or v6->v4 allocations will be dependent on
the SSODA work being discussed in TRAM, since this is the only way to
allocate v4 and v6 TURN from a single host candidate.


Also worth noting:

With the current default settings, a single STUN and TURN server, and
assuming a dual-stack host with two network interfaces, a
PeerConnection with audio, video, and data will generate

5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) *
2 (IP protocols) = 80 candidates

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<div dir=3D"ltr"><pre class=3D"" style=3D"font-size:1em;margin-top:0px;marg=
in-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"=
>From the transports draft, S 3.4:</font></pre><pre class=3D"" style=3D"fon=
t-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">

<br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bott=
om:0px;color:rgb(0,0,0)">   In order to deal with situations where one part=
y is on an IPv4
   network and the other party is on an IPv6 network, TURN extensions
   for IPv6 [<a href=3D"http://tools.ietf.org/html/rfc6156" title=3D"&quot;=
Traversal Using Relays around NAT (TURN) Extension for IPv6&quot;">RFC6156<=
/a>] MUST be supported.</pre><pre class=3D"" style=3D"font-size:1em;margin-=
top:0px;margin-bottom:0px;color:rgb(0,0,0)">

<br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bott=
om:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif">I thin=
k this needs clarification. Does this mean that we should allocate v6 candi=
dates from all v4 interfaces (for compat with v6-only endpoints), as well a=
s allocate v4 candidates from all v6 interfaces (for compat with v4-only en=
dpoints), and if so, which path (v6-to-TURN or v6-from-TURN) should be prio=
ritized?</font></pre>

<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"><br></font></pre=
><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;co=
lor:rgb(0,0,0)">

<font face=3D"arial, helvetica, sans-serif">Note also that these v4-&gt;v6 =
or v6-&gt;v4 allocations will be dependent on the SSODA work being discusse=
d in TRAM, since this is the only way to allocate v4 and v6 TURN from a sin=
gle host candidate.</font></pre>

<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"><br></font></pre=
><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;co=
lor:rgb(0,0,0)">

<font face=3D"arial, helvetica, sans-serif">Also worth noting:</font></pre>=
<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif">With the current=
 default settings, a single STUN and TURN server, and assuming a dual-stack=
 host with two network interfaces, a PeerConnection with audio, video, and =
data will generate</font></pre>

<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif">5 (components) *=
 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) * 2 (IP protocols) =3D =
80 candidates</font></pre>

<pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;col=
or:rgb(0,0,0)"><br></pre><pre class=3D"" style=3D"font-size:1em;margin-top:=
0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, san=
s-serif"><br>

</font></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-b=
ottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"" style=3D"font-size:1e=
m;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=3D"arial, h=
elvetica, sans-serif"><br>

</font></pre><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-b=
ottom:0px;color:rgb(0,0,0)"><font face=3D"arial, helvetica, sans-serif"><br=
></font></pre></div>

--089e013c6f9c2f5c3804f3dd3cff--


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On Wed, Mar 5, 2014 at 2:41 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
>
> On Mar 5, 2014, at 2:25 PM, Justin Uberti <juberti@google.com> wrote:
>
> > So there are three things here:
> > 1) MUST the implementation offer encrypted header extensions? (i.e.
> mandatory-to-implement)
> I=E2=80=99m fine with yes
>
> > 2) MUST the implementation use encrypted header extensions? (i.e.
> mandatory-to-use)
> I think this needs to be No. Never mind my current use case - if the far
> end does not negotiate RFC 6094 with the browser, the media and headers
> should not fail.
>
> > 3) MUST the implementation expose an API to control this? (i.e. no SDP
> munging needed)
> I prefer control surfaces to SDP munging but as you say, I could live wit=
h
> SDP munging in 1.0
>
> >
> > I think we want yes for #1, no for #2, and #3 is potentially interestin=
g
> but out of scope for 1.0.
> > That gives encrypted headers for audio on by default, but remote partie=
s
> can negotiate this off using RFC 6904 mechanisms.
> >
>
>
> seems workable
>

Where should this be codified (in which I-D?)

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Mar 5, 2014 at 2:41 PM, Cullen Jennings (fluffy) <span dir=
=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@c=
isco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D""><br>
<br>
On Mar 5, 2014, at 2:25 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@goo=
gle.com">juberti@google.com</a>&gt; wrote:<br>
<br>
&gt; So there are three things here:<br>
&gt; 1) MUST the implementation offer encrypted header extensions? (i.e. ma=
ndatory-to-implement)<br>
</div>I=E2=80=99m fine with yes<br>
<div class=3D""><br>
&gt; 2) MUST the implementation use encrypted header extensions? (i.e. mand=
atory-to-use)<br>
</div>I think this needs to be No. Never mind my current use case - if the =
far end does not negotiate RFC 6094 with the browser, the media and headers=
 should not fail.<br>
<div class=3D""><br>
&gt; 3) MUST the implementation expose an API to control this? (i.e. no SDP=
 munging needed)<br>
</div>I prefer control surfaces to SDP munging but as you say, I could live=
 with SDP munging in 1.0<br>
<div class=3D""><br>
&gt;<br>
&gt; I think we want yes for #1, no for #2, and #3 is potentially interesti=
ng but out of scope for 1.0.<br>
&gt; That gives encrypted headers for audio on by default, but remote parti=
es can negotiate this off using RFC 6904 mechanisms.<br>
&gt;<br>
<br>
<br>
</div>seems workable<br></blockquote><div><br></div><div>Where should this =
be codified (in which I-D?)=C2=A0</div></div><br></div></div>

--089e0129534efa4cfb04f3dd4110--


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El 05/03/2014 13:32, Cullen Jennings (fluffy) escribió:
> I am opposed to mandating that Client-to-Mixer Audio Level RTP header is REQUIRED to be encrypted. It means that we can not really build conferencing system where the mixers do not have access to decrypt the media.
>
>
Hi Cullen,

I am not sure I am following you. AFAIK DTLS is only possible to be 
setup between two peers, so in a multiconference solution the mixer will 
anyway have to establish a DTLS/SRTP session which each peer and 
decrypt/encrypt before sending to any of participants in the conference. 
In case of a full meshed conference, there is no need of a mixer at all, 
and the logic could be directly implemented on each of the endpoints.

Am I missing something? Please correct me if I am wrong.

Best regards
Sergio


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--089e0111bce0681e4e04f3dd50fd
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>From the transports draft, Section 3.4:

   TURN TCP candidates [RFC6062 <http://tools.ietf.org/html/rfc6062>]
SHOULD be supported; this allows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)


I don't think we want to do this. The optimization of having a single
vs two TURN servers in the .09% case is not worth the implementation
or runtime cost of allocating TURN TCP candidates. It requires a TCP
connection to the TURN server, which we would otherwise not do except
in fallback cases.

--089e0111bce0681e4e04f3dd50fd
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<div dir=3D"ltr">From the transports draft, Section 3.4:<div><br></div><div=
><pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;co=
lor:rgb(0,0,0)">   TURN TCP candidates [<a href=3D"http://tools.ietf.org/ht=
ml/rfc6062" title=3D"&quot;Traversal Using Relays around NAT (TURN) Extensi=
ons for TCP Allocations&quot;">RFC6062</a>] SHOULD be supported; this allow=
s
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)</pre><pre class=3D"" style=3D"font-size:1em;margin-top:0p=
x;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"" style=3D"fo=
nt-size:1em;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><font face=
=3D"arial, helvetica, sans-serif">I don&#39;t think we want to do this. The=
 optimization of having a single vs two TURN servers in the .09% case is no=
t worth the implementation or runtime cost of allocating TURN TCP candidate=
s. It requires a TCP connection to the TURN server, which we would otherwis=
e not do except in fallback cases.</font></pre>

</div></div>

--089e0111bce0681e4e04f3dd50fd--


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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On 5 Mar 2014, at 14:41, Cullen Jennings (fluffy) <fluffy@cisco.com> =
wrote:

>=20
>=20
> On Mar 5, 2014, at 2:25 PM, Justin Uberti <juberti@google.com> wrote:
>=20
>> So there are three things here:
>> 1) MUST the implementation offer encrypted header extensions? (i.e. =
mandatory-to-implement)
> I=92m fine with yes
>=20
>> 2) MUST the implementation use encrypted header extensions? (i.e. =
mandatory-to-use)
> I think this needs to be No. Never mind my current use case - if the =
far end does not negotiate RFC 6094 with the browser, the media and =
headers should not fail.=20
>=20
>> 3) MUST the implementation expose an API to control this? (i.e. no =
SDP munging needed)
> I prefer control surfaces to SDP munging but as you say, I could live =
with SDP munging in 1.0=20
>=20
>>=20
>> I think we want yes for #1, no for #2, and #3 is potentially =
interesting but out of scope for 1.0.
>> That gives encrypted headers for audio on by default, but remote =
parties can negotiate this off using RFC 6904 mechanisms.
>>=20
>=20
>=20
> seems workable=20

For the record, I still don=92t like the idea that the far end can =
bid-down encryption on this header, but I suppose if my app cares it =
will have to not offer to do that header extension at all. (Hopefully it =
is a malleable SDP field).

Tim.=


From nobody Wed Mar  5 07:21:10 2014
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Le 2014-03-05 14:56, Justin Uberti a écrit :
> I think this needs clarification. Does this mean that we should allocate v6 candidates from all v4 interfaces (for compat with v6-only endpoints), as well as allocate v4 candidates from all v6 interfaces (for compat with v4-only endpoints), and if so, which path (v6-to-TURN or v6-from-TURN) should be prioritized?

You must always allocate both v4 and v6 relayed candidates, irrespective
of what other candidates you have.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca


From nobody Wed Mar  5 07:21:19 2014
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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, tim panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Transports: RFC 6062 support
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Agree on the cost concern.  Not sure I fully understand the context though.
 Is ~.09% the estimated percentage of times that TCP will be the only way
to get through UDP-blocking firewalls, or am I mis-understanding the use
case?

If it does refer to the UDP-blocking, what data is it based on?  I do know
large enterprise firewalls skew significantly higher on UDP blocking, but I
suspect SMB and residential perhaps are more UDP-permissive by default (but
have never seen extensive data that is not biased by the sampling).




On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <juberti@google.com> wrote:

> From the transports draft, Section 3.4:
>
>    TURN TCP candidates [RFC6062 <http://tools.ietf.org/html/rfc6062>] SHO=
ULD be supported; this allows
>    applications to achieve peer-to-peer communication when both parties
>    are behind UDP-blocking firewalls using a single TURN server.  (In
>    this case, one can also achieve communication using two TURN servers
>    that use TCP between the server and the client, and UDP between the
>    TURN servers.)
>
>
> I don't think we want to do this. The optimization of having a single vs =
two TURN servers in the .09% case is not worth the implementation or runtim=
e cost of allocating TURN TCP candidates. It requires a TCP connection to t=
he TURN server, which we would otherwise not do except in fallback cases.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Agree on the cost concern. =A0Not sure I fully understand =
the context though. =A0Is ~.09% the estimated percentage of times that TCP =
will be the only way to get through UDP-blocking firewalls, or am I mis-und=
erstanding the use case? =A0<div>
<br></div><div>If it does refer to the UDP-blocking, what data is it based =
on? =A0I do know large enterprise firewalls skew significantly higher on UD=
P blocking, but I suspect SMB and residential perhaps are more UDP-permissi=
ve by default (but have never seen extensive data that is not biased by the=
 sampling).</div>
<div><br></div><div><br></div></div><div class=3D"gmail_extra"><br><br><div=
 class=3D"gmail_quote">On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">ju=
berti@google.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">From the transports draft, =
Section 3.4:<div><br></div><div><pre style=3D"font-size:1em;margin-bottom:0=
px;margin-top:0px">
   TURN TCP candidates [<a href=3D"http://tools.ietf.org/html/rfc6062" titl=
e=3D"&quot;Traversal Using Relays around NAT (TURN) Extensions for TCP Allo=
cations&quot;" target=3D"_blank">RFC6062</a>] SHOULD be supported; this all=
ows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)</pre><pre style=3D"font-size:1em;margin-bottom:0px;margin=
-top:0px"><br></pre><pre style=3D"font-size:1em;margin-bottom:0px;margin-to=
p:0px"><font face=3D"arial, helvetica, sans-serif">I don&#39;t think we wan=
t to do this. The optimization of having a single vs two TURN servers in th=
e .09% case is not worth the implementation or runtime cost of allocating T=
URN TCP candidates. It requires a TCP connection to the TURN server, which =
we would otherwise not do except in fallback cases.</font></pre>


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<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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From: Justin Uberti <juberti@google.com>
Date: Wed, 5 Mar 2014 15:26:41 +0000
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On Wed, Mar 5, 2014 at 3:21 PM, Simon Perreault <simon.perreault@viagenie.c=
a
> wrote:

> Le 2014-03-05 14:56, Justin Uberti a =C3=A9crit :
> > I think this needs clarification. Does this mean that we should allocat=
e
> v6 candidates from all v4 interfaces (for compat with v6-only endpoints),
> as well as allocate v4 candidates from all v6 interfaces (for compat with
> v4-only endpoints), and if so, which path (v6-to-TURN or v6-from-TURN)
> should be prioritized?
>
> You must always allocate both v4 and v6 relayed candidates, irrespective
> of what other candidates you have.
>

I am fine with that - this needs to be added to the transports document
though.

Any thoughts on the prioritization question?

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Mar 5, 2014 at 3:21 PM, Simon Perreault <span dir=3D"ltr">&=
lt;<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D"_blank">simon.p=
erreault@viagenie.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2014-03-05 14:56, Justin Uberti a =C3=A9c=
rit :<br>
<div class=3D"">&gt; I think this needs clarification. Does this mean that =
we should allocate v6 candidates from all v4 interfaces (for compat with v6=
-only endpoints), as well as allocate v4 candidates from all v6 interfaces =
(for compat with v4-only endpoints), and if so, which path (v6-to-TURN or v=
6-from-TURN) should be prioritized?<br>


<br>
</div>You must always allocate both v4 and v6 relayed candidates, irrespect=
ive<br>
of what other candidates you have.<br></blockquote><div><br></div><div>I am=
 fine with that - this needs to be added to the transports document though.=
</div><div><br></div><div>Any thoughts on the prioritization question?=C2=
=A0</div>

</div></div></div>

--089e0118490ef448ea04f3dda6ae--


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I am assuming 3% as the case where TCP is needed to get through
UDP-blocking firewalls, based on empirical data from QUIC testing in Chrome
over a very large population.

Since TURN TCP candidates (not to be confused with TURN/TCP to connect to
the TURN server) are only useful in the case where BOTH sides are behind
such firewalls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN TCP
candidates are useful. As an optimization, I don't think this is worth
worrying about.


On Wed, Mar 5, 2014 at 3:26 PM, Gil Zino <gilzino@gmail.com> wrote:

> Agree on the cost concern.  Not sure I fully understand the context
> though.  Is ~.09% the estimated percentage of times that TCP will be the
> only way to get through UDP-blocking firewalls, or am I mis-understanding
> the use case?
>
> If it does refer to the UDP-blocking, what data is it based on?  I do kno=
w
> large enterprise firewalls skew significantly higher on UDP blocking, but=
 I
> suspect SMB and residential perhaps are more UDP-permissive by default (b=
ut
> have never seen extensive data that is not biased by the sampling).
>
>
>
>
> On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <juberti@google.com> wrote=
:
>
>> From the transports draft, Section 3.4:
>>
>>
>>    TURN TCP candidates [RFC6062 <http://tools.ietf.org/html/rfc6062>] SH=
OULD be supported; this allows
>>    applications to achieve peer-to-peer communication when both parties
>>    are behind UDP-blocking firewalls using a single TURN server.  (In
>>    this case, one can also achieve communication using two TURN servers
>>    that use TCP between the server and the client, and UDP between the
>>    TURN servers.)
>>
>>
>> I don't think we want to do this. The optimization of having a single vs=
 two TURN servers in the .09% case is not worth the implementation or runti=
me cost of allocating TURN TCP candidates. It requires a TCP connection to =
the TURN server, which we would otherwise not do except in fallback cases.
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

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<div dir=3D"ltr">I am assuming 3% as the case where TCP is needed to get th=
rough UDP-blocking firewalls, based on empirical data from QUIC testing in =
Chrome over a very large population.<div><br></div><div>Since TURN TCP cand=
idates (not to be confused with TURN/TCP to connect to the TURN server) are=
 only useful in the case where BOTH sides are behind such firewalls, 3% * 3=
% =3D=3D .09% is the fraction of calls where TURN TCP candidates are useful=
. As an optimization, I don&#39;t think this is worth worrying about.</div>


</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed,=
 Mar 5, 2014 at 3:26 PM, Gil Zino <span dir=3D"ltr">&lt;<a href=3D"mailto:g=
ilzino@gmail.com" target=3D"_blank">gilzino@gmail.com</a>&gt;</span> wrote:=
<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex">

<div dir=3D"ltr">Agree on the cost concern. =C2=A0Not sure I fully understa=
nd the context though. =C2=A0Is ~.09% the estimated percentage of times tha=
t TCP will be the only way to get through UDP-blocking firewalls, or am I m=
is-understanding the use case? =C2=A0<div>


<br></div><div>If it does refer to the UDP-blocking, what data is it based =
on? =C2=A0I do know large enterprise firewalls skew significantly higher on=
 UDP blocking, but I suspect SMB and residential perhaps are more UDP-permi=
ssive by default (but have never seen extensive data that is not biased by =
the sampling).</div>


<div><br></div><div><br></div></div><div class=3D"gmail_extra"><br><br><div=
 class=3D"gmail_quote"><div><div class=3D"h5">On Wed, Mar 5, 2014 at 10:02 =
AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.co=
m" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br>


</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5"><div dir=
=3D"ltr">From the transports draft, Section 3.4:<div><br></div><div><pre st=
yle=3D"font-size:1em;margin-bottom:0px;margin-top:0px">

   TURN TCP candidates [<a href=3D"http://tools.ietf.org/html/rfc6062" titl=
e=3D"&quot;Traversal Using Relays around NAT (TURN) Extensions for TCP Allo=
cations&quot;" target=3D"_blank">RFC6062</a>] SHOULD be supported; this all=
ows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)</pre><pre style=3D"font-size:1em;margin-bottom:0px;margin=
-top:0px"><br></pre><pre style=3D"font-size:1em;margin-bottom:0px;margin-to=
p:0px"><font face=3D"arial, helvetica, sans-serif">I don&#39;t think we wan=
t to do this. The optimization of having a single vs two TURN servers in th=
e .09% case is not worth the implementation or runtime cost of allocating T=
URN TCP candidates. It requires a TCP connection to the TURN server, which =
we would otherwise not do except in fallback cases.</font></pre>




</div></div>
<br></div></div>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div>

--001a1136563834ce1504f3ddb8d9--


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On Mar 5, 2014, at 2:25 PM, Justin Uberti <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

So there are three things here:
1) MUST the implementation offer encrypted header extensions? (i.e. mandato=
ry-to-implement)
2) MUST the implementation use encrypted header extensions? (i.e. mandatory=
-to-use)
3) MUST the implementation expose an API to control this? (i.e. no SDP mung=
ing needed)

I think we want yes for #1, no for #2, and #3 is potentially interesting bu=
t out of scope for 1.0.
That gives encrypted headers for audio on by default, but remote parties ca=
n negotiate this off using RFC 6904 mechanisms.

Note that in 6904=92s procedures, an offerer has to explicitly offer both e=
ncrypted and unencrypted versions of a header extension if it wants to allo=
w the answerer to choose between them.

To support this use case, in RTCWeb terms, this would mean that a browser w=
ould a) have to offer both the encrypted and unencrypted versions of the he=
ader extension, and b) answer an offer that offered only the unencrypted he=
ader by agreeing to use it.

That said, I=92m not sure I believe in Cullen=92s use case =97 there=92s to=
o much other information that this unauthorized conference mixer wouldn=92t=
 be able to do.  (e.g., request I-frames, or detect them, for switching; or=
 generate proper RTCP reports for sometimes-on streams).

On Wed, Mar 5, 2014 at 2:05 PM, Cullen Jennings (fluffy) <fluffy@cisco.com<=
mailto:fluffy@cisco.com>> wrote:

Well if people don=92t want to allow the application to choose to encrypt t=
his header or not, I will be arguing for MUST NOT encrypt the RTP headers -=
 all of them.

A recurring theme at IETF is that people trying to protect E2E privacy, do =
things that result in the real world having the only way they can build app=
lications is to build middle boxes that pretend to to be an end to the thin=
gs on both sides. This means that the middle boxes end up with 100% ability=
 to change everything and we loose all the protection we hoped to get with =
E2E security.


On Mar 5, 2014, at 1:59 PM, tim panton <tim@phonefromhere.com<mailto:tim@ph=
onefromhere.com>> wrote:

>
> On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy) <fluffy@cisco.com<mailt=
o:fluffy@cisco.com>> wrote:
>
>>
>> I am opposed to mandating that Client-to-Mixer Audio Level RTP header is=
 REQUIRED to be encrypted. It means that we can not really build conferenci=
ng system where the mixers do not have access to decrypt the media.
>
> I have to ask:
> Does that matter enough to weaken user security for everyone else?
>
>
>>
>> I would like to propose that the JS application can control  if the head=
er is encrypted or not.
>
> I am deeply uncomfortable about that. I am not a good enough cryptographe=
r to know, but my instinct
> is that exposing the fact that (for example) most of the top bits of the =
ulaw data are zeros in this packet
> must be a risk.
>
> I=92m pretty confident that one could detect the pitch of the speaker and=
 probably fingerprint that down
> to individuals or languages with enough data.
>
>>
>> I am aware of the paper that suggests these audio levels can reveal the =
contents of the conversation. There is some element of truth to this. There=
 is also some element of truth to the point that if this was true, speech r=
ecognition system would work better than they do. Encrypting this or not is=
 a risk that the application using the header extension needs to evaluate, =
and WebRTC system needs to provide the applications with a control surfaces=
 that allows them to use this in the way the applications desires.
>
> I am also uncomfortable about the number of little ad-hoc control surface=
s that we seem to be mandating
> without an overall design.
>
> My take is that we should push this out to 2.0 when hopefully we will hav=
e a designed control surface in place.
>
>>
>> I will note that an normal application that used this header could decid=
e if it was encrypted or not, what is different in RTCWeb is that we are bu=
ilding a platform and my view is that this platform should allow the applic=
ations build on the platform to decide the tradeoffs of risk for encrypting=
 this or not.
>
> That isn=92t the line we have taken in any other encryption discussion.
>
> Tim.
>

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space;">
<br>
<div>
<div>On Mar 5, 2014, at 2:25 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com">juberti@google.com</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div dir=3D"ltr">So there are three things here:
<div>1) MUST the implementation offer encrypted header extensions? (i.e. ma=
ndatory-to-implement)</div>
<div>2) MUST the implementation use encrypted header extensions? (i.e. mand=
atory-to-use)</div>
<div>3) MUST the implementation expose an API to control this? (i.e. no SDP=
 munging needed)<br>
</div>
<div><br>
</div>
<div>I think we want yes for #1, no for #2, and #3 is potentially interesti=
ng but out of scope for 1.0.</div>
<div>That gives encrypted headers for audio on by default, but remote parti=
es can negotiate this off using RFC 6904 mechanisms.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Note that in 6904=92s procedures, an offerer has to explicitly offer b=
oth encrypted and unencrypted versions of a header extension if it wants to=
 allow the answerer to choose between them.</div>
<div><br>
</div>
<div>To support this use case, in RTCWeb terms, this would mean that a brow=
ser would a) have to offer both the encrypted and unencrypted versions of t=
he header extension, and b) answer an offer that offered only the unencrypt=
ed header by agreeing to use it.</div>
<div><br>
</div>
<div>That said, I=92m not sure I believe in Cullen=92s use case =97 there=
=92s too much other information that this unauthorized conference mixer wou=
ldn=92t be able to do. &nbsp;(e.g., request I-frames, or detect them, for s=
witching; or generate proper RTCP reports for sometimes-on
 streams).&nbsp;</div>
<br>
<blockquote type=3D"cite">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">On Wed, Mar 5, 2014 at 2:05 PM, Cullen Jennings =
(fluffy)
<span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank"=
>fluffy@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Well if people don=92t want to allow the application to choose to encrypt t=
his header or not, I will be arguing for MUST NOT encrypt the RTP headers -=
 all of them.<br>
<br>
A recurring theme at IETF is that people trying to protect E2E privacy, do =
things that result in the real world having the only way they can build app=
lications is to build middle boxes that pretend to to be an end to the thin=
gs on both sides. This means that
 the middle boxes end up with 100% ability to change everything and we loos=
e all the protection we hoped to get with E2E security.<br>
<div class=3D"HOEnZb">
<div class=3D"h5"><br>
<br>
On Mar 5, 2014, at 1:59 PM, tim panton &lt;<a href=3D"mailto:tim@phonefromh=
ere.com">tim@phonefromhere.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt; On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy) &lt;<a href=3D"mailt=
o:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I am opposed to mandating that Client-to-Mixer Audio Level RTP hea=
der is REQUIRED to be encrypted. It means that we can not really build conf=
erencing system where the mixers do not have access to decrypt the media.<b=
r>
&gt;<br>
&gt; I have to ask:<br>
&gt; Does that matter enough to weaken user security for everyone else?<br>
&gt;<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I would like to propose that the JS application can control &nbsp;=
if the header is encrypted or not.<br>
&gt;<br>
&gt; I am deeply uncomfortable about that. I am not a good enough cryptogra=
pher to know, but my instinct<br>
&gt; is that exposing the fact that (for example) most of the top bits of t=
he ulaw data are zeros in this packet<br>
&gt; must be a risk.<br>
&gt;<br>
&gt; I=92m pretty confident that one could detect the pitch of the speaker =
and probably fingerprint that down<br>
&gt; to individuals or languages with enough data.<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I am aware of the paper that suggests these audio levels can revea=
l the contents of the conversation. There is some element of truth to this.=
 There is also some element of truth to the point that if this was true, sp=
eech recognition system would work better
 than they do. Encrypting this or not is a risk that the application using =
the header extension needs to evaluate, and WebRTC system needs to provide =
the applications with a control surfaces that allows them to use this in th=
e way the applications desires.<br>
&gt;<br>
&gt; I am also uncomfortable about the number of little ad-hoc control surf=
aces that we seem to be mandating<br>
&gt; without an overall design.<br>
&gt;<br>
&gt; My take is that we should push this out to 2.0 when hopefully we will =
have a designed control surface in place.<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; I will note that an normal application that used this header could=
 decide if it was encrypted or not, what is different in RTCWeb is that we =
are building a platform and my view is that this platform should allow the =
applications build on the platform to
 decide the tradeoffs of risk for encrypting this or not.<br>
&gt;<br>
&gt; That isn=92t the line we have taken in any other encryption discussion=
.<br>
&gt;<br>
&gt; Tim.<br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
https://www.ietf.org/mailman/listinfo/rtcweb<br>
</blockquote>
</div>
<br>
</body>
</html>

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From nobody Wed Mar  5 07:56:08 2014
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Since you asked for more...

Le 2014-03-05 14:56, Justin Uberti a écrit :
> which path (v6-to-TURN or v6-from-TURN) should be prioritized?

I'm sorry, I don't understand this question.

> Note also that these v4->v6 or v6->v4 allocations will be dependent on the SSODA work being discussed in TRAM, since this is the only way to allocate v4 and v6 TURN from a single host candidate.

I don't understand what you mean.

> Also worth noting:
> 
> With the current default settings, a single STUN and TURN server, and assuming a dual-stack host with two network interfaces, a PeerConnection with audio, video, and data will generate
> 
> 5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) * 2 (IP protocols) = 80 candidates

It's....... it's beautiful!

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca


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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Transports: RFC 6062 support
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--_000_4A409D06511D424D8285E38B3E08292Dskypenet_
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+1 given that it still works for those 0.9%, it just hits two TURN servers.=
.. And with luck, that actually is better because if they each select TURN =
servers well you get TCP on the short legs and UDP on the long leg.

Matthew Kaufman

Sent from my iPad

On Mar 5, 2014, at 7:32 AM, "Justin Uberti" <juberti@google.com<mailto:jube=
rti@google.com>> wrote:

I am assuming 3% as the case where TCP is needed to get through UDP-blockin=
g firewalls, based on empirical data from QUIC testing in Chrome over a ver=
y large population.

Since TURN TCP candidates (not to be confused with TURN/TCP to connect to t=
he TURN server) are only useful in the case where BOTH sides are behind suc=
h firewalls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN TCP ca=
ndidates are useful. As an optimization, I don't think this is worth worryi=
ng about.


On Wed, Mar 5, 2014 at 3:26 PM, Gil Zino <gilzino@gmail.com<mailto:gilzino@=
gmail.com>> wrote:
Agree on the cost concern.  Not sure I fully understand the context though.=
  Is ~.09% the estimated percentage of times that TCP will be the only way =
to get through UDP-blocking firewalls, or am I mis-understanding the use ca=
se?

If it does refer to the UDP-blocking, what data is it based on?  I do know =
large enterprise firewalls skew significantly higher on UDP blocking, but I=
 suspect SMB and residential perhaps are more UDP-permissive by default (bu=
t have never seen extensive data that is not biased by the sampling).




On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <juberti@google.com<mailto:j=
uberti@google.com>> wrote:
>From the transports draft, Section 3.4:



   TURN TCP candidates [RFC6062<http://tools.ietf.org/html/rfc6062>] SHOULD=
 be supported; this allows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)


I don't think we want to do this. The optimization of having a single vs tw=
o TURN servers in the .09% case is not worth the implementation or runtime =
cost of allocating TURN TCP candidates. It requires a TCP connection to the=
 TURN server, which we would otherwise not do except in fallback cases.

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb



_______________________________________________
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rtcweb@ietf.org<mailto:rtcweb@ietf.org>
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>&#43;1 given that it still works for those 0.9%, it just hits two TURN=
 servers... And with luck, that actually is better because if they each sel=
ect TURN servers well you get TCP on the short legs and UDP on the long leg=
.<br>
<br>
Matthew Kaufman
<div><br>
</div>
<div>Sent from my iPad</div>
</div>
<div><br>
On Mar 5, 2014, at 7:32 AM, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto=
:juberti@google.com">juberti@google.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">I am assuming 3% as the case where TCP is needed to get th=
rough UDP-blocking firewalls, based on empirical data from QUIC testing in =
Chrome over a very large population.
<div><br>
</div>
<div>Since TURN TCP candidates (not to be confused with TURN/TCP to connect=
 to the TURN server) are only useful in the case where BOTH sides are behin=
d such firewalls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN T=
CP candidates are useful. As an optimization,
 I don't think this is worth worrying about.</div>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Wed, Mar 5, 2014 at 3:26 PM, Gil Zino <span d=
ir=3D"ltr">
&lt;<a href=3D"mailto:gilzino@gmail.com" target=3D"_blank">gilzino@gmail.co=
m</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"ltr">Agree on the cost concern. &nbsp;Not sure I fully understa=
nd the context though. &nbsp;Is ~.09% the estimated percentage of times tha=
t TCP will be the only way to get through UDP-blocking firewalls, or am I m=
is-understanding the use case? &nbsp;
<div><br>
</div>
<div>If it does refer to the UDP-blocking, what data is it based on? &nbsp;=
I do know large enterprise firewalls skew significantly higher on UDP block=
ing, but I suspect SMB and residential perhaps are more UDP-permissive by d=
efault (but have never seen extensive
 data that is not biased by the sampling).</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">
<div>
<div class=3D"h5">On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <span dir=
=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">jubert=
i@google.com</a>&gt;</span> wrote:<br>
</div>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div>
<div class=3D"h5">
<div dir=3D"ltr">From the transports draft, Section 3.4:
<div><br>
</div>
<div>
<pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px">
   TURN TCP candidates [<a href=3D"http://tools.ietf.org/html/rfc6062" titl=
e=3D"&quot;Traversal Using Relays around NAT (TURN) Extensions for TCP Allo=
cations&quot;" target=3D"_blank">RFC6062</a>] SHOULD be supported; this all=
ows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)</pre>
<pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"><br></pre>
<pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"><font face=3D=
"arial, helvetica, sans-serif">I don't think we want to do this. The optimi=
zation of having a single vs two TURN servers in the .09% case is not worth=
 the implementation or runtime cost of allocating TURN TCP candidates. It r=
equires a TCP connection to the TURN server, which we would otherwise not d=
o except in fallback cases.</font></pre>
</div>
</div>
<br>
</div>
</div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
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On 5 March 2014 14:30, Justin Uberti <juberti@google.com> wrote:
> I am inclined to make CNAMEs per-PeerConnection (i.e. enforce scenario #2
> behavior) for 1.0, as it has a smaller downside.

I think that I am OK with this.  It might still be possible to
synchronize in the first scenario, but it would require that the
playback protect itself against drift, because CNAME wouldn't be
providing a positive indication that synchronization is safe.


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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On 5 March 2014 14:40, Justin Uberti <juberti@google.com> wrote:
> Agree with Cullen (and FWIW the SRTP encryption uses keys vended from the
> DTLS master secret, so I think even if you are able to crack the SRTP crypto
> it would not be trivial to go backwards up to the DTLS key material)

Yes.  DTLS-SRTP extracts keys, so working backwards to the master
secret would mean that not only have you cracked SRTP, but that you
can also produce a HMAC preimage at will.


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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On 5 March 2014 14:25, Justin Uberti <juberti@google.com> wrote:
> 3) MUST the implementation expose an API to control this? (i.e. no SDP
> munging needed)

I think perhaps we might split this finer and discuss whether the
control is offered for isolated streams or not.


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Date: Wed, 05 Mar 2014 17:23:07 +0100
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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El 05/03/2014 16:45, Jonathan Lennox escribió:
>
> That said, I'm not sure I believe in Cullen's use case --- there's too 
> much other information that this unauthorized conference mixer 
> wouldn't be able to do.  (e.g., request I-frames, or detect them, for 
> switching; or generate proper RTCP reports for sometimes-on streams).

I have deeper doubts, I don't really know how a RTP router/mixer would 
be able to work without acting as DTLS endpoint and 
decripting/encripting the RTP data.  Or is there a way to setup/share a 
DTLS session between more than two peers so the same DTLS/SRTP packet is 
decriptable by more than one receiver?

Best regards
Sergio

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    <div class="moz-cite-prefix">El 05/03/2014 16:45, Jonathan Lennox
      escribi&oacute;:<br>
    </div>
    <blockquote
      cite="mid:E7C174FB-B137-4E6D-81AC-06C8C2B30FE1@vidyo.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <br>
      <div>That said, I&#8217;m not sure I believe in Cullen&#8217;s use case &#8212;
        there&#8217;s too much other information that this unauthorized
        conference mixer wouldn&#8217;t be able to do. &nbsp;(e.g., request
        I-frames, or detect them, for switching; or generate proper RTCP
        reports for sometimes-on streams).&nbsp;<br>
      </div>
    </blockquote>
    <br>
    I have deeper doubts, I don't really know how a RTP router/mixer
    would be able to work without acting as DTLS endpoint and
    decripting/encripting the RTP data.&nbsp; Or is there a way to
    setup/share a DTLS session between more than two peers so the same
    DTLS/SRTP packet is decriptable by more than one receiver?<br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------050100020608060907050107--


From nobody Wed Mar  5 08:25:38 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 5 Mar 2014 17:25:10 +0100
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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2014-03-05 16:00 GMT+01:00 Sergio Garcia Murillo
<sergio.garcia.murillo@gmail.com>:
> El 05/03/2014 13:32, Cullen Jennings (fluffy) escribi=C3=B3:
>
>> I am opposed to mandating that Client-to-Mixer Audio Level RTP header is
>> REQUIRED to be encrypted. It means that we can not really build conferen=
cing
>> system where the mixers do not have access to decrypt the media.
>>
>>
> Hi Cullen,
>
> I am not sure I am following you. AFAIK DTLS is only possible to be setup
> between two peers, so in a multiconference solution the mixer will anyway
> have to establish a DTLS/SRTP session which each peer and decrypt/encrypt
> before sending to any of participants in the conference. In case of a ful=
l
> meshed conference, there is no need of a mixer at all, and the logic coul=
d
> be directly implemented on each of the endpoints.


Hi Cullen, could you please clarify this? I agree with Sergio: The
mixer (or a media-server before it) has a DTLS and SRTP session with
every participant, so it *already* has to decrypt the received SRTP
(and hence it can decrypt the encrypted RTP header as well) before
processing the media in a mixing fashion (or whichever it does with
it).

Thanks.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>


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From: Justin Uberti <juberti@google.com>
Date: Wed, 5 Mar 2014 16:26:56 +0000
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Subject: [rtcweb] On ptimes and maxptimes
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>From the session this morning, I recall this consensus:

   - We will add recommendations on the frame sizes to use for PCMU/A to
   the rtcweb audio codec I-D (20, 30, 60 ms frames).
   - We will consider adding an a=maxptime attribute that represents the
   "minimum maxptime" of all the offered codecs. That is, if both Opus
   (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a maxptime=60
   SHOULD be included.
      - Since maxptime spans all codecs, this means that some modes (e.g.
      Opus 120ms) will be unavailable, unless only Opus is offered.
   - We will consider adding an a=ptime attribute, set to the receiver's
   preferred frame size to receive. Again, this value spans all codecs, so the
   specified value may not be viable for all codecs (e.g. 30 ms works for PCMU
   but not for Opus)

After thinking about this some, I'm not sure the maxptime/ptime values will
lead to any different behavior than if they were not specified. Since there
is a complexity cost (especially if the values need to change based on
which codecs are offered), is there a clear upside here?

--001a1133757678c54c04f3de7e6c
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<div dir=3D"ltr">From the session this morning, I recall this consensus:<di=
v><ul><li>We will add recommendations on the frame sizes to use for PCMU/A =
to the rtcweb audio codec I-D (20, 30, 60 ms frames).</li><li>We will consi=
der adding an a=3Dmaxptime attribute that represents the &quot;minimum maxp=
time&quot; of all the offered codecs. That is, if both Opus (maxptime of 12=
0ms) and PCMU (maxptime of 60) are offered, a maxptime=3D60 SHOULD be inclu=
ded.</li>

<ul><li>Since maxptime spans all codecs, this means that some modes (e.g. O=
pus 120ms) will be unavailable, unless only Opus is offered.</li></ul><li>W=
e will consider adding an a=3Dptime attribute, set to the receiver&#39;s pr=
eferred frame size to receive. Again, this value spans all codecs, so the s=
pecified value may not be viable for all codecs (e.g. 30 ms works for PCMU =
but not for Opus)</li>

</ul><div>After thinking about this some, I&#39;m not sure the maxptime/pti=
me values will lead to any different behavior than if they were not specifi=
ed. Since there is a complexity cost (especially if the values need to chan=
ge based on which codecs are offered), is there a clear upside here?</div>

</div></div>

--001a1133757678c54c04f3de7e6c--


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From: Justin Uberti <juberti@google.com>
Date: Wed, 5 Mar 2014 16:27:55 +0000
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Subject: Re: [rtcweb] Transports: RFC 6062 support
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Good point on isolating TCP to the short legs.


On Wed, Mar 5, 2014 at 3:55 PM, Matthew Kaufman (SKYPE) <
matthew.kaufman@skype.net> wrote:

>  +1 given that it still works for those 0.9%, it just hits two TURN
> servers... And with luck, that actually is better because if they each
> select TURN servers well you get TCP on the short legs and UDP on the lon=
g
> leg.
>
> Matthew Kaufman
>
>  Sent from my iPad
>
> On Mar 5, 2014, at 7:32 AM, "Justin Uberti" <juberti@google.com> wrote:
>
>   I am assuming 3% as the case where TCP is needed to get through
> UDP-blocking firewalls, based on empirical data from QUIC testing in Chro=
me
> over a very large population.
>
>  Since TURN TCP candidates (not to be confused with TURN/TCP to connect
> to the TURN server) are only useful in the case where BOTH sides are behi=
nd
> such firewalls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN T=
CP
> candidates are useful. As an optimization, I don't think this is worth
> worrying about.
>
>
> On Wed, Mar 5, 2014 at 3:26 PM, Gil Zino <gilzino@gmail.com> wrote:
>
>> Agree on the cost concern.  Not sure I fully understand the context
>> though.  Is ~.09% the estimated percentage of times that TCP will be the
>> only way to get through UDP-blocking firewalls, or am I mis-understandin=
g
>> the use case?
>>
>>  If it does refer to the UDP-blocking, what data is it based on?  I do
>> know large enterprise firewalls skew significantly higher on UDP blockin=
g,
>> but I suspect SMB and residential perhaps are more UDP-permissive by
>> default (but have never seen extensive data that is not biased by the
>> sampling).
>>
>>
>>
>>
>>  On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <juberti@google.com>wrot=
e:
>>
>>>  From the transports draft, Section 3.4:
>>>
>>>     TURN TCP candidates [RFC6062 <http://tools.ietf.org/html/rfc6062>] =
SHOULD be supported; this allows
>>>    applications to achieve peer-to-peer communication when both parties
>>>    are behind UDP-blocking firewalls using a single TURN server.  (In
>>>    this case, one can also achieve communication using two TURN servers
>>>    that use TCP between the server and the client, and UDP between the
>>>    TURN servers.)
>>>
>>>
>>> I don't think we want to do this. The optimization of having a single v=
s two TURN servers in the .09% case is not worth the implementation or runt=
ime cost of allocating TURN TCP candidates. It requires a TCP connection to=
 the TURN server, which we would otherwise not do except in fallback cases.
>>>
>>>
>>>  _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>   _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Good point on isolating TCP to the short legs.</div><div c=
lass=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, Mar 5, 2014=
 at 3:55 PM, Matthew Kaufman (SKYPE) <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:matthew.kaufman@skype.net" target=3D"_blank">matthew.kaufman@skype.net</a=
>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">



<div dir=3D"auto">
<div>+1 given that it still works for those 0.9%, it just hits two TURN ser=
vers... And with luck, that actually is better because if they each select =
TURN servers well you get TCP on the short legs and UDP on the long leg.<br=
>


<br>
Matthew Kaufman
<div><br>
</div>
<div>Sent from my iPad</div>
</div><div><div class=3D"h5">
<div><br>
On Mar 5, 2014, at 7:32 AM, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto=
:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt; wrote:<br=
>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr">I am assuming 3% as the case where TCP is needed to get th=
rough UDP-blocking firewalls, based on empirical data from QUIC testing in =
Chrome over a very large population.
<div><br>
</div>
<div>Since TURN TCP candidates (not to be confused with TURN/TCP to connect=
 to the TURN server) are only useful in the case where BOTH sides are behin=
d such firewalls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN T=
CP candidates are useful. As an optimization,
 I don&#39;t think this is worth worrying about.</div>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Wed, Mar 5, 2014 at 3:26 PM, Gil Zino <span d=
ir=3D"ltr">
&lt;<a href=3D"mailto:gilzino@gmail.com" target=3D"_blank">gilzino@gmail.co=
m</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"ltr">Agree on the cost concern. =C2=A0Not sure I fully understa=
nd the context though. =C2=A0Is ~.09% the estimated percentage of times tha=
t TCP will be the only way to get through UDP-blocking firewalls, or am I m=
is-understanding the use case? =C2=A0
<div><br>
</div>
<div>If it does refer to the UDP-blocking, what data is it based on? =C2=A0=
I do know large enterprise firewalls skew significantly higher on UDP block=
ing, but I suspect SMB and residential perhaps are more UDP-permissive by d=
efault (but have never seen extensive
 data that is not biased by the sampling).</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">
<div>
<div>On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <span dir=3D"ltr">&lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt;</span> wrote:<br>
</div>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div>
<div>
<div dir=3D"ltr">From the transports draft, Section 3.4:
<div><br>
</div>
<div>
<pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px">   TURN TCP c=
andidates [<a href=3D"http://tools.ietf.org/html/rfc6062" title=3D"&quot;Tr=
aversal Using Relays around NAT (TURN) Extensions for TCP Allocations&quot;=
" target=3D"_blank">RFC6062</a>] SHOULD be supported; this allows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)</pre>
<pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"><br></pre>
<pre style=3D"font-size:1em;margin-bottom:0px;margin-top:0px"><font face=3D=
"arial, helvetica, sans-serif">I don&#39;t think we want to do this. The op=
timization of having a single vs two TURN servers in the .09% case is not w=
orth the implementation or runtime cost of allocating TURN TCP candidates. =
It requires a TCP connection to the TURN server, which we would otherwise n=
ot do except in fallback cases.</font></pre>


</div>
</div>
<br>
</div>
</div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org<=
/a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_b=
lank">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</div></div></div>

</blockquote></div><br></div>

--001a1136b262f1281604f3de8165--


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(TURN) relayed candidates MUST NOT be prioritized as it is always =
possible to other high priority candidates succeed for the WebRTC =
session. Also, TURN candidates is not required to be allocated for all =
the session. Hope trickle ICE solve these problems as the allocation =
shall happen.

=20

IMO, it is very much possible to implement in the WebRTC network =
(server/endpoint) without TURN server and with reduced WebRTC session =
setup time.

=20

Thanks

Partha

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Justin Uberti
Sent: Wednesday, March 05, 2014 8:57 PM
To: Simon Perreault
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Transports questions on IPv6

=20

=20

=20

On Wed, Mar 5, 2014 at 3:21 PM, Simon Perreault =
<simon.perreault@viagenie.ca> wrote:

Le 2014-03-05 14:56, Justin Uberti a =C3=A9crit :

> I think this needs clarification. Does this mean that we should =
allocate v6 candidates from all v4 interfaces (for compat with v6-only =
endpoints), as well as allocate v4 candidates from all v6 interfaces =
(for compat with v4-only endpoints), and if so, which path (v6-to-TURN =
or v6-from-TURN) should be prioritized?

You must always allocate both v4 and v6 relayed candidates, irrespective
of what other candidates you have.

=20

I am fine with that - this needs to be added to the transports document =
though.

=20

Any thoughts on the prioritization question?=20


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 </o:shapelayout></xml><![endif]-->
</head>

<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>(TURN) relayed candidates MUST NOT be prioritized as it =
is
always possible to other high priority candidates succeed for the WebRTC =
session.
Also, TURN candidates is not required to be allocated for all the =
session. Hope
trickle ICE solve these problems as the allocation shall =
happen.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>IMO, it is very much possible to implement in the WebRTC =
network
(server/endpoint) without TURN server and with reduced WebRTC session =
setup
time.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<o:p></o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb
[mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Justin Uberti<br>
<b>Sent:</b> Wednesday, March 05, 2014 8:57 PM<br>
<b>To:</b> Simon Perreault<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Transports questions on =
IPv6<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>On Wed, Mar 5, 2014 at 3:21 PM, Simon Perreault =
&lt;<a
href=3D"mailto:simon.perreault@viagenie.ca" =
target=3D"_blank">simon.perreault@viagenie.ca</a>&gt;
wrote:<o:p></o:p></p>

<p class=3DMsoNormal>Le 2014-03-05 14:56, Justin Uberti a =C3=A9crit =
:<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>&gt; I think this =
needs
clarification. Does this mean that we should allocate v6 candidates from =
all v4
interfaces (for compat with v6-only endpoints), as well as allocate v4
candidates from all v6 interfaces (for compat with v4-only endpoints), =
and if
so, which path (v6-to-TURN or v6-from-TURN) should be =
prioritized?<o:p></o:p></p>

</div>

<p class=3DMsoNormal>You must always allocate both v4 and v6 relayed =
candidates,
irrespective<br>
of what other candidates you have.<o:p></o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>I am fine with that - this needs to be added to the
transports document though.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Any thoughts on the prioritization =
question?&nbsp;<o:p></o:p></p>

</div>

</div>

</div>

</div>

</div>

</div>

</body>

</html>

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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 5 Mar 2014 16:30:19 +0000
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Subject: Re: [rtcweb] Transports: RFC 6062 support
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Justin said:

"Since TURN TCP candidates (not to be confused with TURN/TCP to connect to
the TURN server) are only useful in the case where BOTH sides are behind
such firewalls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN TCP
candidates are useful. As an optimization, I don't think this is worth
worrying about."

[BA] +1.

If TURN/TCP isn't sufficient to traverse the firewall then you've typically
got a problem that TURN TCP candidates won't solve either.


On Wed, Mar 5, 2014 at 3:31 PM, Justin Uberti <juberti@google.com> wrote:

> I am assuming 3% as the case where TCP is needed to get through
> UDP-blocking firewalls, based on empirical data from QUIC testing in Chro=
me
> over a very large population.
>
> Since TURN TCP candidates (not to be confused with TURN/TCP to connect to
> the TURN server) are only useful in the case where BOTH sides are behind
> such firewalls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN T=
CP
> candidates are useful. As an optimization, I don't think this is worth
> worrying about.
>
>
> On Wed, Mar 5, 2014 at 3:26 PM, Gil Zino <gilzino@gmail.com> wrote:
>
>> Agree on the cost concern.  Not sure I fully understand the context
>> though.  Is ~.09% the estimated percentage of times that TCP will be the
>> only way to get through UDP-blocking firewalls, or am I mis-understandin=
g
>> the use case?
>>
>> If it does refer to the UDP-blocking, what data is it based on?  I do
>> know large enterprise firewalls skew significantly higher on UDP blockin=
g,
>> but I suspect SMB and residential perhaps are more UDP-permissive by
>> default (but have never seen extensive data that is not biased by the
>> sampling).
>>
>>
>>
>>
>> On Wed, Mar 5, 2014 at 10:02 AM, Justin Uberti <juberti@google.com>wrote=
:
>>
>>> From the transports draft, Section 3.4:
>>>
>>>
>>>    TURN TCP candidates [RFC6062 <http://tools.ietf.org/html/rfc6062>] S=
HOULD be supported; this allows
>>>    applications to achieve peer-to-peer communication when both parties
>>>    are behind UDP-blocking firewalls using a single TURN server.  (In
>>>    this case, one can also achieve communication using two TURN servers
>>>    that use TCP between the server and the client, and UDP between the
>>>    TURN servers.)
>>>
>>>
>>> I don't think we want to do this. The optimization of having a single v=
s two TURN servers in the .09% case is not worth the implementation or runt=
ime cost of allocating TURN TCP candidates. It requires a TCP connection to=
 the TURN server, which we would otherwise not do except in fallback cases.
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><div>Justin said: </div><div><br></div><div>&quot;Since TU=
RN TCP candidates (not to be confused with TURN/TCP to connect to the TURN =
server) are only useful in the case where BOTH sides are behind such firewa=
lls, 3% * 3% =3D=3D .09% is the fraction of calls where TURN TCP candidates=
 are useful. As an optimization, I don&#39;t think this is worth worrying a=
bout.&quot;</div>

<div><br></div><div>[BA] +1.=A0</div><div><br></div><div>If TURN/TCP isn&#3=
9;t sufficient to traverse the=A0firewall then you&#39;ve typically got a p=
roblem=A0that TURN TCP candidates won&#39;t solve either. =A0</div></div><d=
iv class=3D"gmail_extra">

<br><br><div class=3D"gmail_quote">On Wed, Mar 5, 2014 at 3:31 PM, Justin U=
berti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D=
"_blank">juberti@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">

<div dir=3D"ltr">I am assuming 3% as the case where TCP is needed to get th=
rough UDP-blocking firewalls, based on empirical data from QUIC testing in =
Chrome over a very large population.<div><br></div><div>Since TURN TCP cand=
idates (not to be confused with TURN/TCP to connect to the TURN server) are=
 only useful in the case where BOTH sides are behind such firewalls, 3% * 3=
% =3D=3D .09% is the fraction of calls where TURN TCP candidates are useful=
. As an optimization, I don&#39;t think this is worth worrying about.</div>




</div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><b=
r><br><div class=3D"gmail_quote">On Wed, Mar 5, 2014 at 3:26 PM, Gil Zino <=
span dir=3D"ltr">&lt;<a href=3D"mailto:gilzino@gmail.com" target=3D"_blank"=
>gilzino@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;padding=
-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-l=
eft-style:solid">

<div dir=3D"ltr">Agree on the cost concern. =A0Not sure I fully understand =
the context though. =A0Is ~.09% the estimated percentage of times that TCP =
will be the only way to get through UDP-blocking firewalls, or am I mis-und=
erstanding the use case? =A0<div>




<br></div><div>If it does refer to the UDP-blocking, what data is it based =
on? =A0I do know large enterprise firewalls skew significantly higher on UD=
P blocking, but I suspect SMB and residential perhaps are more UDP-permissi=
ve by default (but have never seen extensive data that is not biased by the=
 sampling).</div>




<div><br></div><div><br></div></div><div class=3D"gmail_extra"><br><br><div=
 class=3D"gmail_quote"><div><div>On Wed, Mar 5, 2014 at 10:02 AM, Justin Ub=
erti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"=
_blank">juberti@google.com</a>&gt;</span> wrote:<br>




</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:=
1px;border-left-style:solid"><div><div><div dir=3D"ltr">From the transports=
 draft, Section 3.4:<div>

<br></div><div><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px=
">
   TURN TCP candidates [<a title=3D"&quot;Traversal Using Relays around NAT=
 (TURN) Extensions for TCP Allocations&quot;" href=3D"http://tools.ietf.org=
/html/rfc6062" target=3D"_blank">RFC6062</a>] SHOULD be supported; this all=
ows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)</pre><pre style=3D"font-size:1em;margin-top:0px;margin-bo=
ttom:0px"><br></pre><pre style=3D"font-size:1em;margin-top:0px;margin-botto=
m:0px"><font face=3D"arial, helvetica, sans-serif">I don&#39;t think we wan=
t to do this. The optimization of having a single vs two TURN servers in th=
e .09% case is not worth the implementation or runtime cost of allocating T=
URN TCP candidates. It requires a TCP connection to the TURN server, which =
we would otherwise not do except in fallback cases.</font></pre>






</div></div>
<br></div></div>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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EKT is one possible answer to this concern.

By the way: If and when we specify EKT for RTCWEB, there is a simple rule w=
e should establish: A "browser" must not initiate the use of EKT. That way,=
 browser to browser sessions would be unable to force a known key... That c=
ould only happen when talking to non-browser endpoints (likely mixers)

Matthew Kaufman

Sent from my iPad

> On Mar 5, 2014, at 8:23 AM, "Sergio Garcia Murillo" <sergio.garcia.murill=
o@gmail.com> wrote:
>=20
> El 05/03/2014 16:45, Jonathan Lennox escribi=F3:
>>=20
>> That said, I=92m not sure I believe in Cullen=92s use case =97 there=92s=
 too much other information that this unauthorized conference mixer wouldn=
=92t be able to do.  (e.g., request I-frames, or detect them, for switching=
; or generate proper RTCP reports for sometimes-on streams).=20
>=20
> I have deeper doubts, I don't really know how a RTP router/mixer would be=
 able to work without acting as DTLS endpoint and decripting/encripting the=
 RTP data.  Or is there a way to setup/share a DTLS session between more th=
an two peers so the same DTLS/SRTP packet is decriptable by more than one r=
eceiver?
>=20
> Best regards
> Sergio
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Date: Wed, 05 Mar 2014 17:41:00 +0100
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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In that case, we could link the usage of EKT to unencripted rtp header 
extensions (and not only audio-to-mixer level), moreover as you suggest 
that it would not be enabled in P2P communications and only when a mixer 
is involved.

Best regards
Sergio
El 05/03/2014 17:34, Matthew Kaufman (SKYPE) escribió:
> EKT is one possible answer to this concern.
>
> By the way: If and when we specify EKT for RTCWEB, there is a simple rule we should establish: A "browser" must not initiate the use of EKT. That way, browser to browser sessions would be unable to force a known key... That could only happen when talking to non-browser endpoints (likely mixers)
>
> Matthew Kaufman
>
> Sent from my iPad
>
>> On Mar 5, 2014, at 8:23 AM, "Sergio Garcia Murillo" <sergio.garcia.murillo@gmail.com> wrote:
>>
>> El 05/03/2014 16:45, Jonathan Lennox escribió:
>>> That said, I’m not sure I believe in Cullen’s use case — there’s too much other information that this unauthorized conference mixer wouldn’t be able to do.  (e.g., request I-frames, or detect them, for switching; or generate proper RTCP reports for sometimes-on streams).
>> I have deeper doubts, I don't really know how a RTP router/mixer would be able to work without acting as DTLS endpoint and decripting/encripting the RTP data.  Or is there a way to setup/share a DTLS session between more than two peers so the same DTLS/SRTP packet is decriptable by more than one receiver?
>>
>> Best regards
>> Sergio
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


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On 5 March 2014 16:34, Matthew Kaufman (SKYPE)
<matthew.kaufman@skype.net> wrote:
> By the way: If and when we specify EKT for RTCWEB, there is a simple rule=
 we should establish: A "browser" must not initiate the use of EKT. That wa=
y, browser to browser sessions would be unable to force a known key... That=
 could only happen when talking to non-browser endpoints (likely mixers)

I've talked to several people about this and we all agree on this
point.  It's important, if browsers ever implement EKT.

We might also require that "private" sessions not accept attempts to
use EKT too.


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On 5 March 2014 15:55, Matthew Kaufman (SKYPE)
<matthew.kaufman@skype.net> wrote:
> +1 given that it still works for those 0.9%, it just hits two TURN
> servers... And with luck, that actually is better because if they each
> select TURN servers well you get TCP on the short legs and UDP on the long
> leg.


Agree.


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From: Justin Uberti <juberti@google.com>
Date: Wed, 5 Mar 2014 18:16:00 +0000
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Subject: Re: [rtcweb] Transports questions on IPv6
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On Wed, Mar 5, 2014 at 3:55 PM, Simon Perreault <simon.perreault@viagenie.c=
a
> wrote:

> Since you asked for more...
>
> Le 2014-03-05 14:56, Justin Uberti a =C3=A9crit :
> > which path (v6-to-TURN or v6-from-TURN) should be prioritized?
>
> I'm sorry, I don't understand this question.
>

Imagine I am a v6-only endpoint. Via IPv6, I allocate 2 ports on the TURN
server, v4 and v6. The remote client is v4-only, and via IPv4 it also
allocates v4 and v6 on its TURN server. So we need to relay to connect v4
and v6. Which path should be preferred?

a) Caller -- TURN/v6 --> TURN Server(Caller, v4 alloc) <-- v4 -- Callee
[ Caller candidate is relay, Callee candidate is host ]
b) Caller -- v6 --> TURN Server(Callee, v6 alloc) <-- TURN/v4 -- Callee
[ Caller candidate is host, Callee candidate is relay ]

I worked this through and I think ICE will choose (b), due to the (G>D?1:0)
tiebreaker, which seems right to me. So no action needed :)



> > Note also that these v4->v6 or v6->v4 allocations will be dependent on
> the SSODA work being discussed in TRAM, since this is the only way to
> allocate v4 and v6 TURN from a single host candidate.
>
> I don't understand what you mean.
>

I was assuming that WebRTC endpoints will need to do cross-family
allocations, to handle the case where they are talking to a v4-only or
v6-only endpoint and they are not v4- or v6- capable. If so, the transports
draft will need to reference the SSODA work once it becomes a WG item,
since I don't think WebRTC endpoints should create additional sockets to do
cross-family allocations (we have enough sockets already, as shown below).


> > Also worth noting:
> >
> > With the current default settings, a single STUN and TURN server, and
> assuming a dual-stack host with two network interfaces, a PeerConnection
> with audio, video, and data will generate
> >
> > 5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) * 2
> (IP protocols) =3D 80 candidates
>
> It's....... it's beautiful!
>
> Simon
> --
> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> STUN/TURN server               --> http://numb.viagenie.ca
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Mar 5, 2014 at 3:55 PM, Simon Perreault <span dir=3D"ltr">&=
lt;<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D"_blank">simon.p=
erreault@viagenie.ca</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Since you asked for more...<br>
<div><br>
Le 2014-03-05 14:56, Justin Uberti a =C3=A9crit :<br>
</div><div>&gt; which path (v6-to-TURN or v6-from-TURN) should be prioritiz=
ed?<br>
<br>
</div>I&#39;m sorry, I don&#39;t understand this question.<br></blockquote>=
<div><br></div><div>Imagine I am a v6-only endpoint. Via IPv6, I allocate 2=
 ports on the TURN server, v4 and v6. The remote client is v4-only, and via=
 IPv4 it also allocates v4 and v6 on its TURN server. So we need to relay t=
o connect v4 and v6. Which path should be preferred?</div>

<div><br></div><div>a) Caller -- TURN/v6 --&gt; TURN Server(Caller, v4 allo=
c) &lt;-- v4 -- Callee</div><div>[ Caller candidate is relay, Callee candid=
ate is host ]</div><div>b) Caller -- v6 --&gt; TURN Server(Callee, v6 alloc=
) &lt;-- TURN/v4 -- Callee</div>

<div>[ Caller candidate is host, Callee candidate is relay ]</div><div><br>=
</div><div>I worked this through and I think ICE will choose (b), due to th=
e (G&gt;D?1:0) tiebreaker, which seems right to me. So no action needed :)<=
/div>

<div><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div><br>
&gt; Note also that these v4-&gt;v6 or v6-&gt;v4 allocations will be depend=
ent on the SSODA work being discussed in TRAM, since this is the only way t=
o allocate v4 and v6 TURN from a single host candidate.<br>
<br>
</div>I don&#39;t understand what you mean.<br></blockquote><div><br></div>=
<div>I was assuming that WebRTC endpoints will need to do cross-family allo=
cations, to handle the case where they are talking to a v4-only or v6-only =
endpoint and they are not v4- or v6- capable. If so, the transports draft w=
ill need to reference the SSODA work once it becomes a WG item, since I don=
&#39;t think WebRTC endpoints should create additional sockets to do cross-=
family allocations (we have enough sockets already, as shown below).</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<div><br>
&gt; Also worth noting:<br>
&gt;<br>
&gt; With the current default settings, a single STUN and TURN server, and =
assuming a dual-stack host with two network interfaces, a PeerConnection wi=
th audio, video, and data will generate<br>
&gt;<br>
&gt; 5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) * =
2 (IP protocols) =3D 80 candidates<br>
<br>
</div>It&#39;s....... it&#39;s beautiful!<br>
<div><div><br>
Simon<br>
--<br>
DTN made easy, lean, and smart --&gt; <a href=3D"http://postellation.viagen=
ie.ca" target=3D"_blank">http://postellation.viagenie.ca</a><br>
NAT64/DNS64 open-source =C2=A0 =C2=A0 =C2=A0 =C2=A0--&gt; <a href=3D"http:/=
/ecdysis.viagenie.ca" target=3D"_blank">http://ecdysis.viagenie.ca</a><br>
STUN/TURN server =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 --&gt; <a=
 href=3D"http://numb.viagenie.ca" target=3D"_blank">http://numb.viagenie.ca=
</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--047d7bdc969c81b35304f3e0040e--


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On 5 March 2014 18:16, Justin Uberti <juberti@google.com> wrote:
> I worked this through and I think ICE will choose (b), due to the (G>D?1:0)
> tiebreaker, which seems right to me. So no action needed :)

Isn't it the case that you can assign different local priorities to
these to get the result that you need/want such that the tie breaker
is not necessary.  I would suggest that ICE would prefer the
v6-all-the-way option, but the happy eyeballs stuff might cause the
precedence to flip.


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On 5 March 2014 15:55, Simon Perreault <simon.perreault@viagenie.ca> wrote:
>> 5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) * 2 (IP protocols) = 80 candidates
>
> It's....... it's beautiful!

Indeed!  Now imagine pairing those.


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Le 2014-03-05 18:34, Martin Thomson a Ã©crit :
> On 5 March 2014 15:55, Simon Perreault <simon.perreault@viagenie.ca> wrote:
>>> 5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) * 2 (IP protocols) = 80 candidates
>>
>> It's....... it's beautiful!
> 
> Indeed!  Now imagine pairing those.

5 * ( 16 * 15 / 2 ) = 600 pairs

A marvel of engineering!

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca


From nobody Wed Mar  5 11:26:30 2014
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Le 2014-03-05 15:55, Matthew Kaufman (SKYPE) a écrit :
> if they each select TURN servers well you get TCP on the short legs and
> UDP on the long leg.

How would they "select TURN servers well"?

We have a milestone for solving that problem in TRAM, and no draft
currently adopted for it.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca


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--089e013a1b227d64a704f3e10c18
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On Wed, Mar 5, 2014 at 6:53 PM, Simon Perreault <simon.perreault@viagenie.c=
a
> wrote:

> Le 2014-03-05 18:34, Martin Thomson a =C3=A9crit :
> > On 5 March 2014 15:55, Simon Perreault <simon.perreault@viagenie.ca>
> wrote:
> >>> 5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interfaces) *
> 2 (IP protocols) =3D 80 candidates
> >>
> >> It's....... it's beautiful!
> >
> > Indeed!  Now imagine pairing those.
>
> 5 * ( 16 * 15 / 2 ) =3D 600 pairs
>
> A marvel of engineering!
>

At 20 ms paced checks, this is... 12 seconds.

(Fortunately many pairs don't make sense, e.g. v4->v6, udp->tcp, stun->*,
and can be omitted)

>
> Simon
> --
> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> STUN/TURN server               --> http://numb.viagenie.ca
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Mar 5, 2014 at 6:53 PM, Simon Perreault <span dir=3D"ltr">&=
lt;<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D"_blank">simon.p=
erreault@viagenie.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2014-03-05 18:34, Martin Thomson a =C3=A9=
crit :<br>
<div class=3D"">&gt; On 5 March 2014 15:55, Simon Perreault &lt;<a href=3D"=
mailto:simon.perreault@viagenie.ca">simon.perreault@viagenie.ca</a>&gt; wro=
te:<br>
&gt;&gt;&gt; 5 (components) * 4 (local-udp+stun+turn+local-tcp) * 2 (interf=
aces) * 2 (IP protocols) =3D 80 candidates<br>
&gt;&gt;<br>
&gt;&gt; It&#39;s....... it&#39;s beautiful!<br>
&gt;<br>
&gt; Indeed! =C2=A0Now imagine pairing those.<br>
<br>
</div>5 * ( 16 * 15 / 2 ) =3D 600 pairs<br>
<br>
A marvel of engineering!<br></blockquote><div><br></div><div>At 20 ms paced=
 checks, this is... 12 seconds.</div><div><br></div><div>(Fortunately many =
pairs don&#39;t make sense, e.g. v4-&gt;v6, udp-&gt;tcp, stun-&gt;*, and ca=
n be omitted)=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D"im HOEnZb"><br>
Simon<br>
--<br>
DTN made easy, lean, and smart --&gt; <a href=3D"http://postellation.viagen=
ie.ca" target=3D"_blank">http://postellation.viagenie.ca</a><br>
NAT64/DNS64 open-source =C2=A0 =C2=A0 =C2=A0 =C2=A0--&gt; <a href=3D"http:/=
/ecdysis.viagenie.ca" target=3D"_blank">http://ecdysis.viagenie.ca</a><br>
STUN/TURN server =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 --&gt; <a=
 href=3D"http://numb.viagenie.ca" target=3D"_blank">http://numb.viagenie.ca=
</a><br>
<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">_____________________________=
__________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--089e013a1b227d64a704f3e10c18--


From nobody Wed Mar  5 11:49:22 2014
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On 3/4/14 7:24 PM, Makaraju, Maridi Raju (Raju) wrote:
>>>> 2. Section 4:
>>>>
>>>> The method
>>>>     used to determine which side uses odd or even is based on the
>>>>     underlying DTLS connection role when used in WebRTC, with the side
>>>>     acting as the DTLS client using even stream identifiers.
>>>>
>>>> Isn't this unnecessary using the vague word of WebRTC instead of simply
>>>> pointing to the DTLS roles of the established data channel?
>>> The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and therefore
>>> you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
>>> or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
>>> In that case DTLS is not used and you can not refer to the DTLS role.
>>> That is why the restriction is used.
>>
>> So data channels could work over SCTP/IP or SCTP/UDP/IP, but in fact
>> can't solely because the choice of even/odd role is dependent on DTLS
>> connection role?
>>
>> Couldn't you find a way to choose the even/odd role based solely on the
>> SCTP layer and the SDP? Then data channels could be used over those
>> other stacks.
> [Raju]
> +1
> a=setup could be used to determine this, but the final determination is done only at SDP answer (which is same for DTLS).
> For DTLS/SCTP case since both DTLS and SCTP layers use the same a=setup, both DTLS and SCTP client/servers roles match.
> So, DCEP changing text to indicate role determination be done per SCTP client/server role instead of DTLS role may just work and is compatible with existing text.
> [/Raju]

+1

	Thanks,
	Paul


From nobody Wed Mar  5 12:40:13 2014
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Simon Perreault <simon.perreault@viagenie.ca>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Transports: RFC 6062 support
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/Ey5u7y0cmKw9sbLLIYAmGIf-TqY
Subject: Re: [rtcweb] Transports: RFC 6062 support
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TRAM is solving the "without the assistance of the JavaScript" case (findin=
g corporate servers or ISP servers)... I'm sure that some of the larger RTC=
WEB application providers will be more than capable of figuring out which T=
URN servers will be best to assign to their customers.

Matthew Kaufman

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Simon
> Perreault
> Sent: Wednesday, March 5, 2014 7:26 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Transports: RFC 6062 support
>=20
> Le 2014-03-05 15:55, Matthew Kaufman (SKYPE) a =E9crit :
> > if they each select TURN servers well you get TCP on the short legs
> > and UDP on the long leg.
>=20
> How would they "select TURN servers well"?
>=20
> We have a milestone for solving that problem in TRAM, and no draft
> currently adopted for it.
>=20
> Simon
> --
> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> STUN/TURN server               --> http://numb.viagenie.ca
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Le 2014-03-05 20:39, Matthew Kaufman (SKYPE) a écrit :
> TRAM is solving the "without the assistance of the JavaScript" case (finding corporate servers or ISP servers)... I'm sure that some of the larger RTCWEB application providers will be more than capable of figuring out which TURN servers will be best to assign to their customers.

Right. So the "TCP on the short legs, UDP on the long leg" argument
currently only holds for large application providers.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca


From nobody Thu Mar  6 01:29:00 2014
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From: Martin Thomson <martin.thomson@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] HTTP POST for IdP
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As discussed in the meeting, I've put together a proposal for handling
IdP interactions using HTTP POST.

This adds a direct translation of the exchanges that are sent over the
Message Channel to JSON over HTTP POST.  This allows servers or
non-browser applications to use the IdP.

More details:
 - the "id" and "origin" parameters are only used for the MessageChannel mapping
   - for "id", HTTP provides adequate request-response correlation
   - for "origin", the HTTP Origin header should suffice
 - there is a new field added to the identity assertion that
identifies the supported interaction models: "proxy", "http" or both,
included in an array
    - on this point, we should discuss in the W3C whether we will add
this parameter to setIdentityProvider
 - there are some structural changes required to make this flow properly
 - I've referenced the new JSON RFC

Actual change request:
https://github.com/ekr/ietf-drafts/pull/14/files


From nobody Thu Mar  6 01:56:31 2014
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTP usage: supporting RTP ECN?
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Date: Thu, 6 Mar 2014 09:56:15 +0000
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Hi,

draft-ietf-rtcweb-rtp-usage-12 doesn't have any description regarding to RT=
P ECN RFC6679. I'm wondering why. But from my point of view, it's worth som=
e discussion. The usage of ECN is a trend though it is not universal in cur=
rent network. Supporting it in browsers for RTP media transport may be good=
 for future.

BR,
Rachel

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<body lang=3D"ZH-CN" link=3D"blue" vlink=3D"purple" style=3D"text-justify-t=
rim:punctuation">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">draft-ietf-rtcweb-rtp-usage-12 =
doesn&#8217;t have any description regarding to RTP ECN RFC6679. I&#8217;m =
wondering why. But from my point of view, it&#8217;s worth some discussion.=
 The usage of ECN is a trend though it is not universal
 in current network. Supporting it in browsers for RTP media transport may =
be good for future.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">BR,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Rachel<o:p></o:p></span></p>
</div>
</body>
</html>

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From: Justin Uberti <juberti@google.com>
Date: Thu, 6 Mar 2014 13:11:22 +0000
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I would expect that corp or ISP servers would also be located very close to
the endpoint.


On Thu, Mar 6, 2014 at 9:11 AM, Simon Perreault <simon.perreault@viagenie.c=
a
> wrote:

> Le 2014-03-05 20:39, Matthew Kaufman (SKYPE) a =C3=A9crit :
> > TRAM is solving the "without the assistance of the JavaScript" case
> (finding corporate servers or ISP servers)... I'm sure that some of the
> larger RTCWEB application providers will be more than capable of figuring
> out which TURN servers will be best to assign to their customers.
>
> Right. So the "TCP on the short legs, UDP on the long leg" argument
> currently only holds for large application providers.
>
> Simon
> --
> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> STUN/TURN server               --> http://numb.viagenie.ca
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">I would expect that corp or ISP servers would also be loca=
ted very close to the endpoint.</div><div class=3D"gmail_extra"><br><br><di=
v class=3D"gmail_quote">On Thu, Mar 6, 2014 at 9:11 AM, Simon Perreault <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D=
"_blank">simon.perreault@viagenie.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2014-03-05 20:39, Matthew Kaufman (SKYPE)=
 a =C3=A9crit :<br>
<div class=3D"">&gt; TRAM is solving the &quot;without the assistance of th=
e JavaScript&quot; case (finding corporate servers or ISP servers)... I&#39=
;m sure that some of the larger RTCWEB application providers will be more t=
han capable of figuring out which TURN servers will be best to assign to th=
eir customers.<br>


<br>
</div>Right. So the &quot;TCP on the short legs, UDP on the long leg&quot; =
argument<br>
currently only holds for large application providers.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Simon<br>
--<br>
DTN made easy, lean, and smart --&gt; <a href=3D"http://postellation.viagen=
ie.ca" target=3D"_blank">http://postellation.viagenie.ca</a><br>
NAT64/DNS64 open-source =C2=A0 =C2=A0 =C2=A0 =C2=A0--&gt; <a href=3D"http:/=
/ecdysis.viagenie.ca" target=3D"_blank">http://ecdysis.viagenie.ca</a><br>
STUN/TURN server =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 --&gt; <a=
 href=3D"http://numb.viagenie.ca" target=3D"_blank">http://numb.viagenie.ca=
</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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Le 2014-03-06 13:11, Justin Uberti a Ã©crit :
> I would expect that corp or ISP servers would also be located very close
> to the endpoint.

Right, but my point is that we still don't have a way for those corp- or
ISP-provided servers to be discovered by the endpoint. We had a draft
presented in TRAM yesterday, but it was not adopted. So I guess we're at
a point where we need ideas.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca


From nobody Thu Mar  6 05:22:54 2014
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From: Justin Uberti <juberti@google.com>
Date: Thu, 6 Mar 2014 13:22:16 +0000
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On Wed, Mar 5, 2014 at 3:45 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

>
>  On Mar 5, 2014, at 2:25 PM, Justin Uberti <juberti@google.com> wrote:
>
>  So there are three things here:
> 1) MUST the implementation offer encrypted header extensions? (i.e.
> mandatory-to-implement)
> 2) MUST the implementation use encrypted header extensions? (i.e.
> mandatory-to-use)
> 3) MUST the implementation expose an API to control this? (i.e. no SDP
> munging needed)
>
>  I think we want yes for #1, no for #2, and #3 is potentially interesting
> but out of scope for 1.0.
> That gives encrypted headers for audio on by default, but remote parties
> can negotiate this off using RFC 6904 mechanisms.
>
>
>  Note that in 6904=E2=80=99s procedures, an offerer has to explicitly off=
er both
> encrypted and unencrypted versions of a header extension if it wants to
> allow the answerer to choose between them.
>
>  To support this use case, in RTCWeb terms, this would mean that a
> browser would a) have to offer both the encrypted and unencrypted version=
s
> of the header extension, and b) answer an offer that offered only the
> unencrypted header by agreeing to use it.
>

That is a good point, and implies that for any header extension that
recommends encryption (e.g. RFC 6464 for audio level), we have the
following choices:
a) Implementations MUST offer only the encrypted version, and MUST accept
only the encrypted version
b) Implementations MUST offer both the encrypted and unencrypted versions,
and MUST accept both

Of these, a) is the less complicated option.

--bcaec5469505e6574604f3f00762
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Mar 5, 2014 at 3:45 PM, Jonathan Lennox <span dir=3D"ltr">&=
lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jonathan@vidyo.c=
om</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">



<div style=3D"word-wrap:break-word">
<br>
<div><div class=3D"">
<div>On Mar 5, 2014, at 2:25 PM, Justin Uberti &lt;<a href=3D"mailto:jubert=
i@google.com" target=3D"_blank">juberti@google.com</a>&gt; wrote:</div>
<br>
</div><div class=3D""><blockquote type=3D"cite">
<div dir=3D"ltr">So there are three things here:
<div>1) MUST the implementation offer encrypted header extensions? (i.e. ma=
ndatory-to-implement)</div>
<div>2) MUST the implementation use encrypted header extensions? (i.e. mand=
atory-to-use)</div>
<div>3) MUST the implementation expose an API to control this? (i.e. no SDP=
 munging needed)<br>
</div>
<div><br>
</div>
<div>I think we want yes for #1, no for #2, and #3 is potentially interesti=
ng but out of scope for 1.0.</div>
<div>That gives encrypted headers for audio on by default, but remote parti=
es can negotiate this off using RFC 6904 mechanisms.</div>
</div>
</blockquote>
<div><br>
</div>
</div><div>Note that in 6904=E2=80=99s procedures, an offerer has to explic=
itly offer both encrypted and unencrypted versions of a header extension if=
 it wants to allow the answerer to choose between them.</div>
<div><br>
</div>
<div>To support this use case, in RTCWeb terms, this would mean that a brow=
ser would a) have to offer both the encrypted and unencrypted versions of t=
he header extension, and b) answer an offer that offered only the unencrypt=
ed header by agreeing to use it.</div>

</div></div></blockquote><div><br></div><div>That is a good point, and impl=
ies that for any header extension that recommends encryption (e.g. RFC 6464=
 for audio level), we have the following choices:</div><div>a) Implementati=
ons MUST offer only the encrypted version, and MUST accept only the encrypt=
ed version</div>

<div>b) Implementations MUST offer both the encrypted and unencrypted versi=
ons, and MUST accept both</div><div><br></div><div>Of these, a) is the less=
 complicated option.</div><div><br></div><div><br></div></div></div></div>


--bcaec5469505e6574604f3f00762--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 6 Mar 2014 13:23:50 +0000
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True. But in trying to wrap up this thread, can we agree that TURN TCP
candidates are a SHOULD NOT?


On Thu, Mar 6, 2014 at 1:14 PM, Simon Perreault <simon.perreault@viagenie.c=
a
> wrote:

> Le 2014-03-06 13:11, Justin Uberti a =C3=A9crit :
> > I would expect that corp or ISP servers would also be located very clos=
e
> > to the endpoint.
>
> Right, but my point is that we still don't have a way for those corp- or
> ISP-provided servers to be discovered by the endpoint. We had a draft
> presented in TRAM yesterday, but it was not adopted. So I guess we're at
> a point where we need ideas.
>
> Simon
> --
> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> STUN/TURN server               --> http://numb.viagenie.ca
>

--047d7b343f9a812df704f3f00d9e
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<div dir=3D"ltr">True. But in trying to wrap up this thread, can we agree t=
hat TURN TCP candidates are a SHOULD NOT?</div><div class=3D"gmail_extra"><=
br><br><div class=3D"gmail_quote">On Thu, Mar 6, 2014 at 1:14 PM, Simon Per=
reault <span dir=3D"ltr">&lt;<a href=3D"mailto:simon.perreault@viagenie.ca"=
 target=3D"_blank">simon.perreault@viagenie.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2014-03-06 13:11, Justin Uberti a =C3=A9c=
rit :<br>
<div class=3D"">&gt; I would expect that corp or ISP servers would also be =
located very close<br>
&gt; to the endpoint.<br>
<br>
</div>Right, but my point is that we still don&#39;t have a way for those c=
orp- or<br>
ISP-provided servers to be discovered by the endpoint. We had a draft<br>
presented in TRAM yesterday, but it was not adopted. So I guess we&#39;re a=
t<br>
a point where we need ideas.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Simon<br>
--<br>
DTN made easy, lean, and smart --&gt; <a href=3D"http://postellation.viagen=
ie.ca" target=3D"_blank">http://postellation.viagenie.ca</a><br>
NAT64/DNS64 open-source =C2=A0 =C2=A0 =C2=A0 =C2=A0--&gt; <a href=3D"http:/=
/ecdysis.viagenie.ca" target=3D"_blank">http://ecdysis.viagenie.ca</a><br>
STUN/TURN server =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 --&gt; <a=
 href=3D"http://numb.viagenie.ca" target=3D"_blank">http://numb.viagenie.ca=
</a><br>
</div></div></blockquote></div><br></div>

--047d7b343f9a812df704f3f00d9e--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 6 Mar 2014 13:31:11 +0000
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Subject: Re: [rtcweb] Issue with new Trickle ICE text
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As discussed in the meeting, this doesn't mandate Trickle ICE for the
application (which can just wait until gathering is complete, as indicated
in S 3.4.1). Trickle ICE is mandated for the implementation, but I thought
we had agreed that long ago so that applications can get consistent
behavior.

While the text in S 3.4.2 is new, it is consistent with the text in 4.1.2
and 4.1.5, which defines the behavior for createOffer and
setLocalDescription

createOffer:


   Calling this method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.


setLocalDescription:

   This API indirectly controls the candidate gathering process.  When a
   local description is supplied, and the number of transports currently
   in use does not match the number of transports needed by the local
   description, the PeerConnection will create transports as needed and
   begin gathering candidates for them.



On Wed, Mar 5, 2014 at 9:27 AM, Eric Rescorla <ekr@rtfm.com> wrote:

> S 3.4.2 of JSEP now reads:
>
>     JSEP applications typically inform the browser to begin ICE gathering
>     via the information supplied to setLocalDescription, as this is where
>     the app specifies the number of media streams to gather candidates
>     for.  However, to accelerate cases where the browser knows the number
>     of media streams to use ahead of time, the application MAY ask the
>     browser to gather a pool of potential ICE candidates to help ensure
>     rapid media setup.  When setLocalDescription is eventually called,
>     and the browser goes to gather the needed ICE candidates, it can
>     start by checking if any candidates are available in the pool.
>
> The issue here is that this basically mandates trickle ICE. If the
> pool is smaller than the number of independent ICE streams needed,
> then there is no way for you to "start gathering" (since this text
> says it happens with sLD) and therefore you cannot supply all of the
> candidates in the offer.
>
> If that's not the intention to require (and I don't believe we agreed
> to this semantic), then this needs to be rewritten.
>
> -Ekr
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">As discussed in the meeting, this doesn&#39;t mandate Tric=
kle ICE for the application (which can just wait until gathering is complet=
e, as indicated in S 3.4.1). Trickle ICE is mandated for the implementation=
, but I thought we had agreed that long ago so that applications can get co=
nsistent behavior.<div>

<br></div><div>While the text in S 3.4.2 is new, it is consistent with the =
text in 4.1.2 and 4.1.5, which defines the behavior for createOffer and set=
LocalDescription</div><div><br></div><div>createOffer:</div><div><pre style=
=3D"color:rgb(0,0,0)">

   Calling this method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.</pre></div><div><br></div><div>setLocalD=
escription:</div><div><pre style=3D"color:rgb(0,0,0)">   This API indirectl=
y controls the candidate gathering process.  When a
   local description is supplied, and the number of transports currently
   in use does not match the number of transports needed by the local
   description, the PeerConnection will create transports as needed and
   begin gathering candidates for them.</pre></div></div><div class=3D"gmai=
l_extra"><br><br><div class=3D"gmail_quote">On Wed, Mar 5, 2014 at 9:27 AM,=
 Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=
=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">S 3.4.2 of JSEP now reads:<=
div><br></div><div><div>=C2=A0 =C2=A0 JSEP applications typically inform th=
e browser to begin ICE gathering<span style=3D"white-space:pre-wrap">	</spa=
n></div>

<div>=C2=A0 =C2=A0 via the information supplied to setLocalDescription, as =
this is where<span style=3D"white-space:pre-wrap">	</span></div>

<div>=C2=A0 =C2=A0 the app specifies the number of media streams to gather =
candidates<span style=3D"white-space:pre-wrap">	</span></div><div>=C2=A0 =
=C2=A0 for. =C2=A0However, to accelerate cases where the browser knows the =
number<span style=3D"white-space:pre-wrap">	</span></div>



<div>=C2=A0 =C2=A0 of media streams to use ahead of time, the application M=
AY ask the<span style=3D"white-space:pre-wrap">	</span></div><div>=C2=A0 =
=C2=A0 browser to gather a pool of potential ICE candidates to help ensure<=
span style=3D"white-space:pre-wrap">	</span></div>



<div>=C2=A0 =C2=A0 rapid media setup. =C2=A0When setLocalDescription is eve=
ntually called,<span style=3D"white-space:pre-wrap">	</span></div><div>=C2=
=A0 =C2=A0 and the browser goes to gather the needed ICE candidates, it can=
<span style=3D"white-space:pre-wrap">	</span></div>



<div>=C2=A0 =C2=A0 start by checking if any candidates are available in the=
 pool.</div><div><br></div><div>The issue here is that this basically manda=
tes trickle ICE. If the</div><div>pool is smaller than the number of indepe=
ndent ICE streams needed,</div>



<div>then there is no way for you to &quot;start gathering&quot; (since thi=
s text</div><div>says it happens with sLD) and therefore you cannot supply =
all of the</div><div>candidates in the offer.</div><div><br></div><div>



If that&#39;s not the intention to require (and I don&#39;t believe we agre=
ed</div><div>to this semantic), then this needs to be rewritten.</div></div=
><div><br></div><div>-Ekr</div><div><br></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a11c1da06bd45a204f3f027ec--


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Le 2014-03-06 13:23, Justin Uberti a Ã©crit :
> can we agree that TURN TCP candidates are a SHOULD NOT?

Not a SHOULD, sure. SHOULD NOT, no.

Simon
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On 6 March 2014 13:22, Justin Uberti <juberti@google.com> wrote:
> a) Implementations MUST offer only the encrypted version, and MUST accept
> only the encrypted version
> b) Implementations MUST offer both the encrypted and unencrypted versions,
> and MUST accept both

"either"?

>
> Of these, a) is the less complicated option.

I agree.


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--bcaec520f689e0d2fd04f3f073b6
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On Thu, Mar 6, 2014 at 1:49 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 6 March 2014 13:22, Justin Uberti <juberti@google.com> wrote:
> > a) Implementations MUST offer only the encrypted version, and MUST accept
> > only the encrypted version
> > b) Implementations MUST offer both the encrypted and unencrypted
> versions,
> > and MUST accept both
>
> "either"?
>

Right, that is what I meant.

>
> >
> > Of these, a) is the less complicated option.
>
> I agree.
>

--bcaec520f689e0d2fd04f3f073b6
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Mar 6, 2014 at 1:49 PM, Martin Thomson <span dir=3D"ltr">&l=
t;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thom=
son@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 6 March 2014 13:22, Justi=
n Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&g=
t; wrote:<br>


&gt; a) Implementations MUST offer only the encrypted version, and MUST acc=
ept<br>
&gt; only the encrypted version<br>
&gt; b) Implementations MUST offer both the encrypted and unencrypted versi=
ons,<br>
&gt; and MUST accept both<br>
<br>
</div>&quot;either&quot;?<br></blockquote><div><br></div><div>Right, that i=
s what I meant.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D""><br>
&gt;<br>
&gt; Of these, a) is the less complicated option.<br>
<br>
</div>I agree.<br>
</blockquote></div><br></div></div>

--bcaec520f689e0d2fd04f3f073b6--


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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On Mar 6, 2014, at 1:22 PM, Justin Uberti <juberti@google.com> wrote:

> That is a good point, and implies that for any header extension that reco=
mmends encryption (e.g. RFC 6464 for audio level), we have the following ch=
oices:
> a) Implementations MUST offer only the encrypted version, and MUST accept=
 only the encrypted version
> b) Implementations MUST offer both the encrypted and unencrypted versions=
, and MUST accept both
>=20
> Of these, a) is the less complicated option.

To be precise, 6904 says an answerer MUST NOT accept both encrypted and une=
ncrypted versions of the same header (in the same answer).  I think option =
b) should say that Implementations MUST be prepared to accept either encryp=
ted and unencrypted.  [I.e., I agree with Martin, whose e-mail came in whil=
e I was writing this.]

If we go this way, there=92s then a W3C question of whether to provide an A=
PI knob to choose between the encrypted and unencrypted version of the head=
er extension.


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On 6 March 2014 13:54, Jonathan Lennox <jonathan@vidyo.com> wrote:
> If we go this way, there=E2=80=99s then a W3C question of whether to prov=
ide an API knob to choose between the encrypted and unencrypted version of =
the header extension.

That's a very good reason to go with option a :)  I don't want to
expose that knob to the Internet.


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Hi,

I am not okay with changing 1). I think we MUST mandate implementation
of Header Extension encryption if you implement the audio level header
extensions.

When it comes to use level I am fine with reducing this to RECOMMENDED,
with a reasonable motivation, like enabling middleboxes that can avoid
to be in the security context to read these headers. Thus resulting in a
trade-off between what information one leaks, versus the impact of
having the middlebox in the security context. It may also need to be
made clear that the trade-off for this may further be changed as we
learn more about the feasibility of performing the attack of learning
the content of the voice communication based on the audio levels.

I realize that the above is mostly useful if one actually gets a control
knob for this. But, that is a question for the API.

>From my perspective, allowing for exceptions to the default encryption
on, when so indicated, is the best way forward.

Cheers

Magnus


On 2014-03-05 14:25, Justin Uberti wrote:
> So there are three things here:
> 1) MUST the implementation offer encrypted header extensions? (i.e.
> mandatory-to-implement)
> 2) MUST the implementation use encrypted header extensions? (i.e.
> mandatory-to-use)
> 3) MUST the implementation expose an API to control this? (i.e. no SDP
> munging needed)
> 
> I think we want yes for #1, no for #2, and #3 is potentially interesting
> but out of scope for 1.0.
> That gives encrypted headers for audio on by default, but remote parties
> can negotiate this off using RFC 6904 mechanisms.
> 
> 
> 
> On Wed, Mar 5, 2014 at 2:05 PM, Cullen Jennings (fluffy)
> <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
> 
> 
>     Well if people don’t want to allow the application to choose to
>     encrypt this header or not, I will be arguing for MUST NOT encrypt
>     the RTP headers - all of them.
> 
>     A recurring theme at IETF is that people trying to protect E2E
>     privacy, do things that result in the real world having the only way
>     they can build applications is to build middle boxes that pretend to
>     to be an end to the things on both sides. This means that the middle
>     boxes end up with 100% ability to change everything and we loose all
>     the protection we hoped to get with E2E security.
> 
> 
>     On Mar 5, 2014, at 1:59 PM, tim panton <tim@phonefromhere.com
>     <mailto:tim@phonefromhere.com>> wrote:
> 
>     >
>     > On 5 Mar 2014, at 12:32, Cullen Jennings (fluffy)
>     <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>     >
>     >>
>     >> I am opposed to mandating that Client-to-Mixer Audio Level RTP
>     header is REQUIRED to be encrypted. It means that we can not really
>     build conferencing system where the mixers do not have access to
>     decrypt the media.
>     >
>     > I have to ask:
>     > Does that matter enough to weaken user security for everyone else?
>     >
>     >
>     >>
>     >> I would like to propose that the JS application can control  if
>     the header is encrypted or not.
>     >
>     > I am deeply uncomfortable about that. I am not a good enough
>     cryptographer to know, but my instinct
>     > is that exposing the fact that (for example) most of the top bits
>     of the ulaw data are zeros in this packet
>     > must be a risk.
>     >
>     > I’m pretty confident that one could detect the pitch of the
>     speaker and probably fingerprint that down
>     > to individuals or languages with enough data.
>     >
>     >>
>     >> I am aware of the paper that suggests these audio levels can
>     reveal the contents of the conversation. There is some element of
>     truth to this. There is also some element of truth to the point that
>     if this was true, speech recognition system would work better than
>     they do. Encrypting this or not is a risk that the application using
>     the header extension needs to evaluate, and WebRTC system needs to
>     provide the applications with a control surfaces that allows them to
>     use this in the way the applications desires.
>     >
>     > I am also uncomfortable about the number of little ad-hoc control
>     surfaces that we seem to be mandating
>     > without an overall design.
>     >
>     > My take is that we should push this out to 2.0 when hopefully we
>     will have a designed control surface in place.
>     >
>     >>
>     >> I will note that an normal application that used this header
>     could decide if it was encrypted or not, what is different in RTCWeb
>     is that we are building a platform and my view is that this platform
>     should allow the applications build on the platform to decide the
>     tradeoffs of risk for encrypting this or not.
>     >
>     > That isn’t the line we have taken in any other encryption discussion.
>     >
>     > Tim.
>     >
> 
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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On 6 March 2014 13:11, Justin Uberti <juberti@google.com> wrote:
> I would expect that corp or ISP servers would also be located very close to
> the endpoint.

Not in my experience.  At least not always.  At a previous job, I was
forced to use an HTTP proxy that was on the other side of the planet.
We were unable to get official clearance for a local breakout.


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Subject: [rtcweb] Preserving stream isolation when traversing the network
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There was a bit of confusion yesterday regarding stream isolation.
Mostly the reasons for its existence.  This email is an attempt to
explain the rationale in more detail, as well as to explore some of
the options.

# Problem Statement

This is the feature that we rely on for creating end-to-end protected
and authenticated calls.  We rely on the browser preventing
applications from accessing these streams so that they can't be
modified, recorded, sent to or through servers, rendered to canvas,
pushed into web audio, etc...

The basic idea of isolation is already an important part of the web
security model.  Streams that come from sources other than WebRTC
already require protections of this sort.  For instance, <video> tags
from different origins can't be rendered to canvas, and web audio
cannot take the output of <audio> if it was cross origin (though CORS
can be used to work around this limitation).

The problem I outlined was that any isolation property is not
preserved by WebRTC.  This provides a trivial means of circumvention
of isolated streams.  Sending media using WebRTC effectively strips
away isolation.

# Solution Options

As I described in the working group session, I see two ways to solve
this problem.  One in-band in TLS, the other in signaling.

We seemed to have general agreement that a signal in (D)TLS would be
best.  Signaling creates extra attack modes.  This more closely binds
the isolation guarantee to the security context.

To that end, I've submitted an extension draft to the TLS WG.  I
recommend that you read it and provide comments on its applicability
to RTCWEB here, and comments on the mechanism itself over in TLS.

http://datatracker.ietf.org/doc/draft-thomson-tls-acp/

There is another option I really just thought of, and that is to add
an authenticated parameter to RTP for this purpose.  Probably as an
RTP header extension.  This has different properties, mostly which
apply to the scope question.

# Scope

We agreed yesterday that identity assertions would be scoped to the
session level and that a=identity is session scoped.

I am going to propose that stream isolation is also scoped to the
session.  This implies that all streams from both peers MUST be either
all isolated, or that they are all accessible to the application.

There's an argument that you could scope this to a BUNDLE, but that
seems like unnecessary complication to me.  For instance, we don't
have any API artifacts at this level, upon which this property could
be attached.

# Data Channels

The question as to whether data channels are affected by this
isolation is an interesting one.  Data channels are inherently NEVER
isolated (though we could work on that for some use cases, e.g., file
sharing, I don't think that we're ready for that today.)

I would like to say that data channels can't be created on a
PeerConnection that has isolated streams.  It's logically cleaner that
way.  If we do ever decide to do confidential file transfer (for
example), it would also enable that without new contortions.

I'm not going to get militant on this point though if someone has a
plausible reason to allow use of data channels in the same PC as
isolated streams.


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On Mar 6, 2014, at 1:57 PM, Magnus Westerlund <magnus.westerlund@ericsson.c=
om> wrote:

> Hi,
>=20
> I am not okay with changing 1). I think we MUST mandate implementation
> of Header Extension encryption if you implement the audio level header
> extensions.
>=20
> When it comes to use level I am fine with reducing this to RECOMMENDED,
> with a reasonable motivation, like enabling middleboxes that can avoid
> to be in the security context to read these headers. Thus resulting in a
> trade-off between what information one leaks, versus the impact of
> having the middlebox in the security context. It may also need to be
> made clear that the trade-off for this may further be changed as we
> learn more about the feasibility of performing the attack of learning
> the content of the voice communication based on the audio levels.
>=20
> I realize that the above is mostly useful if one actually gets a control
> knob for this. But, that is a question for the API.
>=20
> From my perspective, allowing for exceptions to the default encryption
> on, when so indicated, is the best way forward.
>=20
> Cheers
>=20
> Magnus

I agree with Magnus - must implement is fine with me, but because using thi=
s feature can have really bad privacy implications, I don=92t want this as =
MUST use. I want the application to be able choose if it uses the feature o=
r not. And of course, the spec needs to have sufficient guidance on the ris=
ks.=20






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On the scoping question, suppose SideA creates isolated streams (that 
can only be sent to SideB) while SideB creates unprotected streams to 
send to SideA.  Are SideA's streams treated as unprotected, or are 
SideB's treated as isolated?  Normally I would think that the weakest 
restriction would win, so that both sides' streams would be unprotected, 
but it would come as quite a surprise to A that something B did could 
cause his streams  to lose isolation (in his own browser, I mean - A 
obviously can't control what B does with the stream once it gets there.)

- Jim
On 3/6/2014 9:21 AM, Martin Thomson wrote:
> There was a bit of confusion yesterday regarding stream isolation.
> Mostly the reasons for its existence.  This email is an attempt to
> explain the rationale in more detail, as well as to explore some of
> the options.
>
> # Problem Statement
>
> This is the feature that we rely on for creating end-to-end protected
> and authenticated calls.  We rely on the browser preventing
> applications from accessing these streams so that they can't be
> modified, recorded, sent to or through servers, rendered to canvas,
> pushed into web audio, etc...
>
> The basic idea of isolation is already an important part of the web
> security model.  Streams that come from sources other than WebRTC
> already require protections of this sort.  For instance, <video> tags
> from different origins can't be rendered to canvas, and web audio
> cannot take the output of <audio> if it was cross origin (though CORS
> can be used to work around this limitation).
>
> The problem I outlined was that any isolation property is not
> preserved by WebRTC.  This provides a trivial means of circumvention
> of isolated streams.  Sending media using WebRTC effectively strips
> away isolation.
>
> # Solution Options
>
> As I described in the working group session, I see two ways to solve
> this problem.  One in-band in TLS, the other in signaling.
>
> We seemed to have general agreement that a signal in (D)TLS would be
> best.  Signaling creates extra attack modes.  This more closely binds
> the isolation guarantee to the security context.
>
> To that end, I've submitted an extension draft to the TLS WG.  I
> recommend that you read it and provide comments on its applicability
> to RTCWEB here, and comments on the mechanism itself over in TLS.
>
> http://datatracker.ietf.org/doc/draft-thomson-tls-acp/
>
> There is another option I really just thought of, and that is to add
> an authenticated parameter to RTP for this purpose.  Probably as an
> RTP header extension.  This has different properties, mostly which
> apply to the scope question.
>
> # Scope
>
> We agreed yesterday that identity assertions would be scoped to the
> session level and that a=identity is session scoped.
>
> I am going to propose that stream isolation is also scoped to the
> session.  This implies that all streams from both peers MUST be either
> all isolated, or that they are all accessible to the application.
>
> There's an argument that you could scope this to a BUNDLE, but that
> seems like unnecessary complication to me.  For instance, we don't
> have any API artifacts at this level, upon which this property could
> be attached.
>
> # Data Channels
>
> The question as to whether data channels are affected by this
> isolation is an interesting one.  Data channels are inherently NEVER
> isolated (though we could work on that for some use cases, e.g., file
> sharing, I don't think that we're ready for that today.)
>
> I would like to say that data channels can't be created on a
> PeerConnection that has isolated streams.  It's logically cleaner that
> way.  If we do ever decide to do confidential file transfer (for
> example), it would also enable that without new contortions.
>
> I'm not going to get militant on this point though if someone has a
> plausible reason to allow use of data channels in the same PC as
> isolated streams.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

-- 
Jim Barnett
Genesys


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: solutions for pervasive survelance
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On Mar 5, 2014, at 3:45 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

> That said, I=92m not sure I believe in Cullen=92s use case =97 there=92s =
too much other information that this unauthorized conference mixer wouldn=
=92t be able to do.  (e.g., request I-frames, or detect them, for switching=
; or generate proper RTCP reports for sometimes-on streams).=20

I talked to Jonathan about this offline and I think there are solutions to =
these issue using EKT but it really needs a draft to explain the overall sy=
stem. I will probably do that before Toronto. =


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Would be good to think about whether the default should be isolated (with a=
 special way for sites to ask the browser to relax the restriction) or not =
isolated (with a way to ask for isolation). The traditional way for "the we=
b" is to do the latter, but I think by now we've seen why we might have wis=
hed otherwise.

Matthew Kaufman

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Martin
> Thomson
> Sent: Thursday, March 6, 2014 2:22 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Preserving stream isolation when traversing the network
>=20
> There was a bit of confusion yesterday regarding stream isolation.
> Mostly the reasons for its existence.  This email is an attempt to explai=
n the
> rationale in more detail, as well as to explore some of the options.
>=20
> # Problem Statement
>=20
> This is the feature that we rely on for creating end-to-end protected and
> authenticated calls.  We rely on the browser preventing applications from
> accessing these streams so that they can't be modified, recorded, sent to=
 or
> through servers, rendered to canvas, pushed into web audio, etc...
>=20
> The basic idea of isolation is already an important part of the web secur=
ity
> model.  Streams that come from sources other than WebRTC already require
> protections of this sort.  For instance, <video> tags from different orig=
ins
> can't be rendered to canvas, and web audio cannot take the output of
> <audio> if it was cross origin (though CORS can be used to work around th=
is
> limitation).
>=20
> The problem I outlined was that any isolation property is not preserved b=
y
> WebRTC.  This provides a trivial means of circumvention of isolated strea=
ms.
> Sending media using WebRTC effectively strips away isolation.
>=20
> # Solution Options
>=20
> As I described in the working group session, I see two ways to solve this
> problem.  One in-band in TLS, the other in signaling.
>=20
> We seemed to have general agreement that a signal in (D)TLS would be best=
.
> Signaling creates extra attack modes.  This more closely binds the isolat=
ion
> guarantee to the security context.
>=20
> To that end, I've submitted an extension draft to the TLS WG.  I recommen=
d
> that you read it and provide comments on its applicability to RTCWEB here=
,
> and comments on the mechanism itself over in TLS.
>=20
> http://datatracker.ietf.org/doc/draft-thomson-tls-acp/
>=20
> There is another option I really just thought of, and that is to add an
> authenticated parameter to RTP for this purpose.  Probably as an RTP head=
er
> extension.  This has different properties, mostly which apply to the scop=
e
> question.
>=20
> # Scope
>=20
> We agreed yesterday that identity assertions would be scoped to the sessi=
on
> level and that a=3Didentity is session scoped.
>=20
> I am going to propose that stream isolation is also scoped to the session=
.  This
> implies that all streams from both peers MUST be either all isolated, or =
that
> they are all accessible to the application.
>=20
> There's an argument that you could scope this to a BUNDLE, but that seems
> like unnecessary complication to me.  For instance, we don't have any API
> artifacts at this level, upon which this property could be attached.
>=20
> # Data Channels
>=20
> The question as to whether data channels are affected by this isolation i=
s an
> interesting one.  Data channels are inherently NEVER isolated (though we
> could work on that for some use cases, e.g., file sharing, I don't think =
that
> we're ready for that today.)
>=20
> I would like to say that data channels can't be created on a PeerConnecti=
on
> that has isolated streams.  It's logically cleaner that way.  If we do ev=
er decide
> to do confidential file transfer (for example), it would also enable that
> without new contortions.
>=20
> I'm not going to get militant on this point though if someone has a plaus=
ible
> reason to allow use of data channels in the same PC as isolated streams.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Mar  6 07:43:16 2014
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On 03/05/2014 11:04 AM, Martin Thomson wrote:
> I haven't thought about this in detail yet, but I wanted to ensure
> that this issue is tracked.  I think that our chairs might be able to
> create an issue in some tracker.  Can I request that they do that for
> this issue?

This is the issue of whether one should (MUST/SHOULD/MAY/MUST NOT) use
the same CNAME for RTP media streams created in different PeerConnections?

In particular, if one does

  pc1.addStream(s)
  pc2.addStream(s)
  pc1.<connect procedure to X>
  pc2.<connect procedure to Y>

will the same CNAME be shown to both X and Y, or not?
And does the answer change if X and Y are the same (for some value of
"same")?


>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: DSCP marking for STUN packets
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ICE says that an endpoint SHOULD apply the same DSCP marking as media packe=
ts to connectivity checks. This looks insufficient for RTCWEB, since it doe=
sn't cover (at least) the following:
- connectivity checks performed for a stream using different drop pecedence=
s (draft-dhesikan-tsvwg-rtcweb-qos).
- connectivity checks performed for non-media/data-channel traffic.
- DSCP markings for consent checks.

Comments on where we need to specify them? The last one can probably go int=
o draft-ietf-rtcweb-stun-consent-freshness?

Muthu

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<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">ICE says that an endpoint SHOULD apply the same DSCP marki=
ng as media packets to connectivity checks. This looks insufficient for RTC=
WEB, since it doesn't cover (at least) the following:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- connectivity checks performed for a stream using differe=
nt drop pecedences (draft-dhesikan-tsvwg-rtcweb-qos).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- connectivity checks performed for non-media/data-channel=
 traffic.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- DSCP markings for consent checks.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Comments on where we need to specify them? The last one ca=
n probably go into draft-ietf-rtcweb-stun-consent-freshness?<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Muthu<o:p></o:p></span></p>
</div>
</body>
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From nobody Thu Mar  6 23:01:52 2014
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From: Martin Thomson <martin.thomson@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On Mar 6, 2014 3:43 PM, "Harald Alvestrand" <harald@alvestrand.no> wrote:
> This is the issue of whether one should (MUST/SHOULD/MAY/MUST NOT) use
> the same CNAME for RTP media streams created in different PeerConnections?

I think that, as Justin has pointed out, we should say 'MUST be different'.
That way, we are able to protect against linkability trivially.

The synchronization case I raised isn't important enough, or difficult
enough to address without CNAME that we would need to risk the greater
exposure that re-use would bring.

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<p dir="ltr"><br>
On Mar 6, 2014 3:43 PM, &quot;Harald Alvestrand&quot; &lt;<a href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
&gt; This is the issue of whether one should (MUST/SHOULD/MAY/MUST NOT) use<br>
&gt; the same CNAME for RTP media streams created in different PeerConnections?</p>
<p dir="ltr">I think that, as Justin has pointed out, we should say &#39;MUST be different&#39;. That way, we are able to protect against linkability trivially.</p>
<p dir="ltr">The synchronization case I raised isn&#39;t important enough, or difficult enough to address without CNAME that we would need to risk the greater exposure that re-use would bring.<br>
</p>

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On Mar 6, 2014 2:38 PM, "Jim Barnett" <1jhbarnett@gmail.com> wrote:
>
> On the scoping question, suppose SideA creates isolated streams (that can
only be sent to SideB) while SideB creates unprotected streams to send to
SideA.  Are SideA's streams treated as unprotected, or are SideB's treated
as isolated?  Normally I would think that the weakest restriction would
win, so that both sides' streams would be unprotected, but it would come as
quite a surprise to A that something B did could cause his streams  to lose
isolation (in his own browser, I mean - A obviously can't control what B
does with the stream once it gets there.)

The idea is that =C3=84=C4=BB=C5=81 streams/tracks would be protected and t=
hat means that
both sides would have to apply the same treatment.

That has several consequences, some of which could be considered negative.
There are a few API niceties that could be added here to mitigate these.  I
omitted those from the first analysis, but I've added them here.

For streams that change between isolated and non-isolated, there are two
ways we handle this.  We already have guidance that streams should send
black/silence if they are (or become) isolated.  And streams that become
non-isolated fail safe.  These protections are the most important, because
they are fundamental properties required by the web sandbox.

The scenario you describe is where each side starts with a different idea
about what isolation properties are required.  Without any other
mechanisms, this fails safe - the (D)TLS handshake would be required to
fail when implementations detect a mismatch in isolate state.

In order to avoid issues with this, we probably need to consider the
addition of a flag in SDP.  That flag would be advisory only, but it would
ensure that you could surface an error in createAnswer that indicates this
mismatch prior to having the problem blow up in less obvious ways.

BTW, we don't have to, but for mixed streams on one side, we should add an
error to createOffer/createAnswer to block the creation of SDP when there
are mixed isolation on streams.  Again, that avoids the issue of having
things break in non-obvious ways later.

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<div dir=3D"ltr"><p dir=3D"ltr"><br>
On Mar 6, 2014 2:38 PM, &quot;Jim Barnett&quot; &lt;<a href=3D"mailto:1jhba=
rnett@gmail.com" target=3D"_blank">1jhbarnett@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On the scoping question, suppose SideA creates isolated streams (that =
can only be sent to SideB) while SideB creates unprotected streams to send =
to SideA. =C2=A0Are SideA&#39;s streams treated as unprotected, or are Side=
B&#39;s treated as isolated? =C2=A0Normally I would think that the weakest =
restriction would win, so that both sides&#39; streams would be unprotected=
, but it would come as quite a surprise to A that something B did could cau=
se his streams =C2=A0to lose isolation (in his own browser, I mean - A obvi=
ously can&#39;t control what B does with the stream once it gets there.)</p=
>


<p dir=3D"ltr">The idea is that =C3=84=C4=BB=C5=81 streams/tracks would be =
protected and that means that both sides would have to apply the same treat=
ment.</p>
<p dir=3D"ltr">That has several consequences, some of which could be consid=
ered negative.=C2=A0 There are a few API niceties that could be added here =
to mitigate these.=C2=A0 I omitted those from the first analysis, but I&#39=
;ve added them here.<br>
</p><p>For streams that change between isolated and non-isolated, there are=
 two ways we handle this.=C2=A0 We already have guidance that streams shoul=
d send black/silence if they are (or become) isolated.=C2=A0 And streams th=
at become non-isolated fail safe.=C2=A0 These protections are the most impo=
rtant, because they are fundamental properties required by the web sandbox.=
<br>
</p><p>The scenario you describe is where each side starts with a different=
 idea about what isolation properties are required.=C2=A0 Without any other=
 mechanisms, this fails safe - the (D)TLS handshake would be required to fa=
il when implementations detect a mismatch in isolate state.<br>
</p><p>In order to avoid issues with this, we probably need to consider the=
 addition of a flag in SDP.=C2=A0 That flag would be advisory only, but it =
would ensure that you could surface an error in createAnswer that indicates=
 this mismatch prior to having the problem blow up in less obvious ways.<br=
>
</p><p>BTW, we don&#39;t have to, but for mixed streams on one side, we=20
should add an error to createOffer/createAnswer to block the creation of
 SDP when there are mixed isolation on streams.=C2=A0 Again, that avoids th=
e issue of having things break in non-obvious ways later.<br></p></div>

--089e010d89e0b6b54904f400bfa9--


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Subject: Re: [rtcweb] Preserving stream isolation when traversing the network
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On 6 March 2014 15:03, Matthew Kaufman (SKYPE)
<matthew.kaufman@skype.net> wrote:
> Would be good to think about whether the default should be isolated (with=
 a special way for sites to ask the browser to relax the restriction) or no=
t isolated (with a way to ask for isolation). The traditional way for "the =
web" is to do the latter, but I think by now we've seen why we might have w=
ished otherwise.

We've talked about this in the past.  There are two aspects that we've
considered: whether to prompt for access to isolated streams (we
decided that this could be considered creepy), and whether to default
to isolation.  I don't think that we can realistically default to
isolation at this point.


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On 6 March 2014 14:21, Martin Thomson <martin.thomson@gmail.com> wrote:
> There is another option I really just thought of, and that is to add
> an authenticated parameter to RTP for this purpose.  Probably as an
> RTP header extension.  This has different properties, mostly which
> apply to the scope question.

Having thought about this, I think that we can discount it.  A
security property of this sort needs to be reliably attached to the
stream.  That means an extension in every SRTP packet if we do it that
way.  In the (D)TLS handshake, it costs much less.

If we wanted isolation on a much more granular level, then this would
be OK, but I don't think that we should be making it that granular.


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On 7 Mar 2014, at 09:21, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 6 March 2014 15:03, Matthew Kaufman (SKYPE)
> <matthew.kaufman@skype.net> wrote:
>> Would be good to think about whether the default should be isolated =
(with a special way for sites to ask the browser to relax the =
restriction) or not isolated (with a way to ask for isolation). The =
traditional way for "the web" is to do the latter, but I think by now =
we've seen why we might have wished otherwise.
>=20
> We've talked about this in the past.  There are two aspects that we've
> considered: whether to prompt for access to isolated streams (we
> decided that this could be considered creepy), and whether to default
> to isolation.  I don't think that we can realistically default to
> isolation at this point.

I'm skeptical about the usefulness of isolation. If I understand it =
correctly, it would disable the ability
of a user interface to do any webGL or webAudio processing on the audio =
or video signals.
Many of the use cases I see use these APIs to do things like add =
titlebars, mix audio streams play audio prompts etc,
isolation reduces webRTC to  door-intercom like functionality, whilst =
there are good usages of that, I suspect they are in the=20
minority.

So I'd oppose any move to make streams isolated by default.

T.


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On 7 Mar 2014, at 09:59, Martin Thomson <martin.thomson@gmail.com> wrote:

> On 6 March 2014 14:21, Martin Thomson <martin.thomson@gmail.com> wrote:
>> There is another option I really just thought of, and that is to add
>> an authenticated parameter to RTP for this purpose.  Probably as an
>> RTP header extension.  This has different properties, mostly which
>> apply to the scope question.
> 
> Having thought about this, I think that we can discount it.  A
> security property of this sort needs to be reliably attached to the
> stream.  That means an extension in every SRTP packet if we do it that
> way.  In the (D)TLS handshake, it costs much less.

Where in the DTLS handshake? There is a risk that if it is too early, 
then we are raising an un encrypted flag saying to the DPI crowd 
"look at me, I'm interesting".

The whole thrust of the IETF's 'encrypt all the things' is to make encryption
default and therefor unexceptional, we shouldn't do anything that makes
isolated calls visibly 'special'.

T.


> 
> If we wanted isolation on a much more granular level, then this would
> be OK, but I don't think that we should be making it that granular.
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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I agree with the conclusion, however:

On 7 March 2014 10:19, Tim Panton <tim@phonefromhere.com> wrote:
> Many of the use cases I see use these APIs to do things like add titlebars, mix audio streams play audio prompts etc,

The stuff you cite can be achieved without having to of webgl or
webaudio.  But I agree with the principle: there are cases where
isolation is inconvenient.


From nobody Fri Mar  7 02:26:07 2014
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On 7 March 2014 10:23, Tim Panton <tim@phonefromhere.com> wrote:
> Where in the DTLS handshake? There is a risk that if it is too early,
> then we are raising an un encrypted flag saying to the DPI crowd
> "look at me, I'm interesting".

Ahh yes, and I wish them luck.

So if you read the draft I referenced (it's short), it's a TLS extension.

In TLS 1.2, those are in the clear.

But in TLS 1.3 we are encrypting extensions.  I expect that browsers
will have TLS 1.3 fairly quickly.


From nobody Fri Mar  7 02:32:11 2014
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My preference is for the former with the caveat of "asking the browser =
to relax the restriction" not involve ye olde generic popup asking the =
user for permission.  The former fits in better with the current =
behavior of audio/video tags.

-jim

On Mar 6, 2014, at 3:03 PM, Matthew Kaufman (SKYPE) =
<matthew.kaufman@skype.net> wrote:

> Would be good to think about whether the default should be isolated =
(with a special way for sites to ask the browser to relax the =
restriction) or not isolated (with a way to ask for isolation). The =
traditional way for "the web" is to do the latter, but I think by now =
we've seen why we might have wished otherwise.
>=20
> Matthew Kaufman
>=20
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Martin
>> Thomson
>> Sent: Thursday, March 6, 2014 2:22 PM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Preserving stream isolation when traversing the =
network
>>=20
>> There was a bit of confusion yesterday regarding stream isolation.
>> Mostly the reasons for its existence.  This email is an attempt to =
explain the
>> rationale in more detail, as well as to explore some of the options.
>>=20
>> # Problem Statement
>>=20
>> This is the feature that we rely on for creating end-to-end protected =
and
>> authenticated calls.  We rely on the browser preventing applications =
from
>> accessing these streams so that they can't be modified, recorded, =
sent to or
>> through servers, rendered to canvas, pushed into web audio, etc...
>>=20
>> The basic idea of isolation is already an important part of the web =
security
>> model.  Streams that come from sources other than WebRTC already =
require
>> protections of this sort.  For instance, <video> tags from different =
origins
>> can't be rendered to canvas, and web audio cannot take the output of
>> <audio> if it was cross origin (though CORS can be used to work =
around this
>> limitation).
>>=20
>> The problem I outlined was that any isolation property is not =
preserved by
>> WebRTC.  This provides a trivial means of circumvention of isolated =
streams.
>> Sending media using WebRTC effectively strips away isolation.
>>=20
>> # Solution Options
>>=20
>> As I described in the working group session, I see two ways to solve =
this
>> problem.  One in-band in TLS, the other in signaling.
>>=20
>> We seemed to have general agreement that a signal in (D)TLS would be =
best.
>> Signaling creates extra attack modes.  This more closely binds the =
isolation
>> guarantee to the security context.
>>=20
>> To that end, I've submitted an extension draft to the TLS WG.  I =
recommend
>> that you read it and provide comments on its applicability to RTCWEB =
here,
>> and comments on the mechanism itself over in TLS.
>>=20
>> http://datatracker.ietf.org/doc/draft-thomson-tls-acp/
>>=20
>> There is another option I really just thought of, and that is to add =
an
>> authenticated parameter to RTP for this purpose.  Probably as an RTP =
header
>> extension.  This has different properties, mostly which apply to the =
scope
>> question.
>>=20
>> # Scope
>>=20
>> We agreed yesterday that identity assertions would be scoped to the =
session
>> level and that a=3Didentity is session scoped.
>>=20
>> I am going to propose that stream isolation is also scoped to the =
session.  This
>> implies that all streams from both peers MUST be either all isolated, =
or that
>> they are all accessible to the application.
>>=20
>> There's an argument that you could scope this to a BUNDLE, but that =
seems
>> like unnecessary complication to me.  For instance, we don't have any =
API
>> artifacts at this level, upon which this property could be attached.
>>=20
>> # Data Channels
>>=20
>> The question as to whether data channels are affected by this =
isolation is an
>> interesting one.  Data channels are inherently NEVER isolated (though =
we
>> could work on that for some use cases, e.g., file sharing, I don't =
think that
>> we're ready for that today.)
>>=20
>> I would like to say that data channels can't be created on a =
PeerConnection
>> that has isolated streams.  It's logically cleaner that way.  If we =
do ever decide
>> to do confidential file transfer (for example), it would also enable =
that
>> without new contortions.
>>=20
>> I'm not going to get militant on this point though if someone has a =
plausible
>> reason to allow use of data channels in the same PC as isolated =
streams.
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Fri Mar  7 02:57:31 2014
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On 03/07/2014 11:23 AM, Martin Thomson wrote:
> I agree with the conclusion, however:
>
> On 7 March 2014 10:19, Tim Panton <tim@phonefromhere.com> wrote:
>> Many of the use cases I see use these APIs to do things like add titlebars, mix audio streams play audio prompts etc,
> The stuff you cite can be achieved without having to of webgl or
> webaudio.  But I agree with the principle: there are cases where
> isolation is inconvenient.

I'd put this a bit more strongly:

Every single non-trivial application that uses WebRTC today is going to
fail if isolation is turned on by default.

Some of the things that will fail:

- Content-sensitive video overlays ("funny hats")
- Color enhancements that attempt to "fix" videos that are "too dark" /
"too light" and so on
- Bluescreening / background replacement
- Volume measurement via WebAudio
- "Voice effects" (also applied via WebAudio)
- Retransmission ("mesh conferencing")
- Any type of recording

All of these are being done by Javascript if they are done at all today.

If data channels are included in the isolate-by-default zone too, of
course every application that uses data channels will fail too - at the
moment there is *nothing* you can do with a data channel that doesn't
expose the data to the Javascript.

I don't think we've thought through the threat model here (well, Martin
may have, but I don't think we have a joint understanding of it), or
what the things are that we won't be able to do without inevitably
exposing our streams to the kinds of manipulation described here.

The threat model I heard at the meeting was that the sender wants a
guarantee from the receiver's browser that it will keep the streams
confidential against Javascript running in the receiver. It gains no
security against compromised receivers, or receivers that participate in
the RTCWEB protocol dialog but aren't Javascript-executing browsers - in
that case, the promise is simply irrelevant.

The promise might  be relevant for some use cases - in particular, the
use case of interactive videoconferencing where the users trust each
other, but not their Javascript, and where they're willing to give up
funny hats, recording and the other things listed above.

I think that's a very specialized use case.

-- 
Surveillance is pervasive. Go Dark.


From nobody Fri Mar  7 03:04:52 2014
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On 7 March 2014 10:57, Harald Alvestrand <harald@alvestrand.no> wrote:
> I think that's a very specialized use case.

While I agree with much of the rest of your argument, casting this as
specialized does our future users a disservice, I think.

I appreciate that some companies consider their "value-add" to be so
compelling that isolation is something they would not consider.  That
doesn't mean that everyone will.


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On 03/07/2014 12:04 PM, Martin Thomson wrote:
> On 7 March 2014 10:57, Harald Alvestrand <harald@alvestrand.no> wrote:
>> I think that's a very specialized use case.
> While I agree with much of the rest of your argument, casting this as
> specialized does our future users a disservice, I think.
>
> I appreciate that some companies consider their "value-add" to be so
> compelling that isolation is something they would not consider.  That
> doesn't mean that everyone will.
There may be people in both camps. Probably will be.

I don't think that "companies' value-add" matters in this situation.
What matters is whether JS access to the media is necessary to achieve
the applications' purpose.

We've seen virtual drum kits that you play by waving your hands in front
of a camera. That application simply can't be made without JS access to
media; it's inherent in what the function is.

That's my point - there are many such applications.


-- 
Surveillance is pervasive. Go Dark.


From nobody Fri Mar  7 12:19:56 2014
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On Mar 5, 2014, at 10:04 AM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> I haven't thought about this in detail yet, but I wanted to ensure
> that this issue is tracked.  I think that our chairs might be able to
> create an issue in some tracker.  Can I request that they do that for
> this issue?

About half a year ago, AVTCORE published an updated recommendation for =
CNAME as RFC7022.  Is its guidance insufficient?

-d


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On 3/7/2014 6:16 AM, Harald Alvestrand wrote:
> On 03/07/2014 12:04 PM, Martin Thomson wrote:
>> On 7 March 2014 10:57, Harald Alvestrand <harald@alvestrand.no> wrote:
>>> I think that's a very specialized use case.
>> While I agree with much of the rest of your argument, casting this as
>> specialized does our future users a disservice, I think.
>>
>> I appreciate that some companies consider their "value-add" to be so
>> compelling that isolation is something they would not consider.  That
>> doesn't mean that everyone will.
> There may be people in both camps. Probably will be.
>
> I don't think that "companies' value-add" matters in this situation.
> What matters is whether JS access to the media is necessary to achieve
> the applications' purpose.

Right.  The example given by Cullen in previous conversations was that 
if someone is using a service provided by Large Company to have internal 
discussions, they want some assurance that Large Company (serving the JS 
code to them for both ends of the conference) doesn't have access to the 
media streams.

I presented an early look at this issue and the use of origins in the W3 
part of the Feb 2012 IETF/W3 interim in MV.

The more general case is a service that wants to provide people with 
secure calls can use isolated streams (modulo the ability of the 
participants to verify through browser UI that they're being used), or 
conversely the ability to tell the browser not to let the JS application 
(*or* the application at the other side!) access the mediastreams.  
(Note that some uses might break, or (if the app was written knowing the 
user can do this) would disable features. That's fine)

This is solely to protect against the JS.  It does not protect against 
other attacks or the far-end user themselves (who could always have a 
camera/mic pointed at the screen).  Even if you trust the application 
provider, in some cases they may have been hacked, or they may have been 
coerced into spying on you (how *would* that happen....)

The ability to do this is important to some people, for the same sort of 
reasons why someone I worked with building videophones kept a post-it 
note over the camera of the videophone on her desk; she didn't feel safe 
without it.

> We've seen virtual drum kits that you play by waving your hands in front
> of a camera. That application simply can't be made without JS access to
> media; it's inherent in what the function is.
>
> That's my point - there are many such applications.

Sure.  I don't think anyone is seriously proposing making isolated 
streams the default (or aren't anymore).

-- 
Randell Jesup -- rjesup a t mozilla d o t com


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To: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 7 March 2014 20:14, Dan Wing <dwing@cisco.com> wrote:
> About half a year ago, AVTCORE published an updated recommendation for CNAME as RFC7022.  Is its guidance insufficient?

Necessary, but not sufficient.  Unfortunately, the definition of a
"session", while sufficient for the description in 7022, is not quite
precise enough for this use case.  The implication there is that it
means "RTP session", which is both not at the right level of
granularity, and not directly actionable.

I also note this little gem:

   A longer-term persistent RTCP CNAME is sometimes useful to facilitate
   third-party monitoring, consistent with [RFC3550].


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From: Justin Uberti <juberti@google.com>
Date: Mon, 10 Mar 2014 05:16:23 +0000
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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I think that the existing guidance should be sufficient in non-multiplexing
cases (i.e. BUNDLE). For consent checks, I think the same DSCP markings as
any other ICE connectivity checks should be used; for ICE-for-SCTP checks,
the same DSCP markings as media packets (i.e. the SCTP traffic) should be
used.

If multiplexing is being performed, the connectivity checks probably should
use the highest DSCP value being used by the multiplexed media.


On Fri, Mar 7, 2014 at 4:35 AM, Muthu Arul Mozhi Perumal (mperumal) <
mperumal@cisco.com> wrote:

>  ICE says that an endpoint SHOULD apply the same DSCP marking as media
> packets to connectivity checks. This looks insufficient for RTCWEB, since
> it doesn't cover (at least) the following:
>
> - connectivity checks performed for a stream using different drop
> pecedences (draft-dhesikan-tsvwg-rtcweb-qos).
>
> - connectivity checks performed for non-media/data-channel traffic.
>
> - DSCP markings for consent checks.
>
>
>
> Comments on where we need to specify them? The last one can probably go
> into draft-ietf-rtcweb-stun-consent-freshness?
>
>
>
> Muthu
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">I think that the existing guidance should be sufficient in=
 non-multiplexing cases (i.e. BUNDLE). For consent checks, I think the same=
 DSCP markings as any other ICE connectivity checks should be used; for ICE=
-for-SCTP checks, the same DSCP markings as media packets (i.e. the SCTP tr=
affic) should be used.<div>

<br></div><div>If multiplexing is being performed, the connectivity checks =
probably should use the highest DSCP value being used by the multiplexed me=
dia.</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quot=
e">

On Fri, Mar 7, 2014 at 4:35 AM, Muthu Arul Mozhi Perumal (mperumal) <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_blank">mper=
umal@cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">







<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">ICE says that an endpoint SHOULD apply the same DSCP marki=
ng as media packets to connectivity checks. This looks insufficient for RTC=
WEB, since it doesn&#39;t cover (at least) the following:<u></u><u></u></sp=
an></p>


<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- connectivity checks performed for a stream using differe=
nt drop pecedences (draft-dhesikan-tsvwg-rtcweb-qos).<u></u><u></u></span><=
/p>


<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- connectivity checks performed for non-media/data-channel=
 traffic.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- DSCP markings for consent checks.<u></u><u></u></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Comments on where we need to specify them? The last one ca=
n probably go into draft-ietf-rtcweb-stun-consent-freshness?<span class=3D"=
HOEnZb"><font color=3D"#888888"><u></u><u></u></font></span></span></p>

<span class=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Muthu<u></u><u></u></span></p>
</font></span></div>
</div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--bcaec51d25b491cc6904f439b5c9--


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From: Justin Uberti <juberti@google.com>
Date: Mon, 10 Mar 2014 06:17:31 +0000
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In my JSEP preso during the RTCWEB session last Wednesday, I discussed
adding a new BUNDLE policy to JSEP. This policy, which would be called
something like =E2=80=9Cforce-bundle=E2=80=9D, would force the use of the s=
ame ports for
BUNDLEd m-sections, as well as RTCP mux. When one knows a priori that the
other side supports BUNDLE, this would reduce candidate gathering even over
the currently defined =E2=80=9Cmax-bundle=E2=80=9D policy, since there woul=
d be no need to
gather RTCP ports at all, and also avoid the need for the BUNDLE fixup
offer (where the ports are updated to be the same for all BUNDLEd
m-sections).

To explain the differences here, here are some example offers for an
audio+video call:

Default bundle policy - separate ports for audio, video, RTP and RTCP:
m=3Daudio 10000
a=3Drtcp:10001
a=3Drtcp-mux
m=3Dvideo 11000
a=3Drtcp:11001
a=3Drtcp-mux

Existing "max-bundle" policy - video port is 0 because bundle-only, but
both RTP and RTCP ports allocated for audio:
m=3Daudio 10000
a=3Drtcp:10001
a=3Drtcp-mux
m=3Dvideo 0
a=3Dbundle-only
a=3Drtcp-mux

The new =E2=80=9Cforce-bundle=E2=80=9D policy - only a single set of ports =
allocated -
audio, video, RTP, RTCP share the same port.
m=3Daudio 10000
a=3Drtcp:10000
a=3Drtcp-mux
m=3Dvideo 10000
a=3Drtcp:10000
a=3Drtcp-mux

During the discussion, the ability to avoid RTCP port gathering was seen as
valuable, but not the ability to avoid the fixup offer. However, after
thinking about it a bit, I don=E2=80=99t think that there is a viable solut=
ion that
avoids RTCP port gathering that *doesn=E2=80=99t* use a single set of ports=
. The
only other option would be to do something like this:

m=3Daudio 10000
a=3Drtcp:10000 # same as m=3Daudio port
a=3Drtcp-mux
m=3Dvideo 0
a=3Dbundle-only
a=3Drtcp-mux

but this will almost surely will result in a malfunction when talking to a
non-rtcp-mux client, due to the same port being used for RTP and RTCP, and
still requires the fixup offer, and as such is net worse than the
force-bundle approach indicated above.

So I think we the force-bundle approach is the one we want to minimize port
gathering. The only question is whether we should keep max-bundle as well.
To be specific, we need to decide between:

a) Keep max-bundle as it is (separate RTP and RTCP ports allocated), and
add a new force-bundle policy that creates an offer with a single port
shared between all m-sections (RTP and RTCP). max-bundle remains nominally
backwards compatible with non-BUNDLE/non-rtcp-mux endpoints, in that it
will still send a single audio stream. force-bundle is not backwards
compatible with non-BUNDLE endpoints.

b) Add force-bundle, but get rid of max-bundle.

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<div dir=3D"ltr">







<p class=3D"">In my JSEP preso during the RTCWEB session last Wednesday, I =
discussed adding a new BUNDLE policy to JSEP. This policy, which would be c=
alled something like =E2=80=9Cforce-bundle=E2=80=9D, would force the use of=
 the same ports for BUNDLEd m-sections, as well as RTCP mux. When one knows=
 a priori that the other side supports BUNDLE, this would reduce candidate =
gathering even over the currently defined =E2=80=9Cmax-bundle=E2=80=9D poli=
cy, since there would be no need to gather RTCP ports at all, and also avoi=
d the need for the BUNDLE fixup offer (where the ports are updated to be th=
e same for all BUNDLEd m-sections).</p>


<p class=3D"">To explain the differences here, here are some example offers=
 for an audio+video call:</p>
<p class=3D"">Default bundle policy - separate ports for audio, video, RTP =
and RTCP:<br>m=3Daudio 10000<br>a=3Drtcp:10001<br>a=3Drtcp-mux<br>m=3Dvideo=
 11000<br>a=3Drtcp:11001<br>a=3Drtcp-mux</p>
<p class=3D"">Existing &quot;max-bundle&quot; policy - video port is 0 beca=
use bundle-only, but both RTP and RTCP ports allocated for audio:<br>m=3Dau=
dio 10000<br>a=3Drtcp:10001<br>a=3Drtcp-mux<br>m=3Dvideo 0<br>a=3Dbundle-on=
ly<br>a=3Drtcp-mux</p>


<p class=3D"">The new =E2=80=9Cforce-bundle=E2=80=9D policy - only a single=
 set of ports allocated - audio, video, RTP, RTCP share the same port.<br>m=
=3Daudio 10000<br>a=3Drtcp:10000<br>a=3Drtcp-mux<br>m=3Dvideo 10000<br>a=3D=
rtcp:10000<br>a=3Drtcp-mux</p>


<p class=3D"">During the discussion, the ability to avoid RTCP port gatheri=
ng was seen as valuable, but not the ability to avoid the fixup offer. Howe=
ver, after thinking about it a bit, I don=E2=80=99t think that there is a v=
iable solution that avoids RTCP port gathering that *doesn=E2=80=99t* use a=
 single set of ports. The only other option would be to do something like t=
his:</p>


<p class=3D"">m=3Daudio 10000<br>a=3Drtcp:10000 # same as m=3Daudio port<br=
>a=3Drtcp-mux<br>m=3Dvideo 0<br>a=3Dbundle-only<br>a=3Drtcp-mux</p>
<p class=3D"">but this will almost surely will result in a malfunction when=
 talking to a non-rtcp-mux client, due to the same port being used for RTP =
and RTCP, and still requires the fixup offer, and as such is net worse than=
 the force-bundle approach indicated above.</p>


<p class=3D"">So I think we the force-bundle approach is the one we want to=
 minimize port gathering. The only question is whether we should keep max-b=
undle as well. To be specific, we need to decide between:</p>
<p class=3D"">a) Keep max-bundle as it is (separate RTP and RTCP ports allo=
cated), and add a new force-bundle policy that creates an offer with a sing=
le port shared between all m-sections (RTP and RTCP). max-bundle remains no=
minally backwards compatible with non-BUNDLE/non-rtcp-mux endpoints, in tha=
t it will still send a single audio stream. force-bundle is not backwards c=
ompatible with non-BUNDLE endpoints.</p>


<p class=3D"">b) Add force-bundle, but get rid of max-bundle.</p></div>

--20cf30780f6234c99304f43a9015--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: BUNDLE with implicit rtcp-mux
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] DSCP marking for STUN packets
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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--_000_E721D8C6A2E1544DB2DEBC313AF54DE22E3108E4xmbrcdx02ciscoc_--


From nobody Mon Mar 10 01:51:46 2014
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Date: Mon, 10 Mar 2014 08:51:36 +0000
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From: Dave Taht <dave.taht@gmail.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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--e89a8f642e1e0c5a6004f43cb6f4
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In all this talk about drop precedence and so on I'd like to point people
at some details of some currently deployed packet scheduling and aqm
systems.

http://snapon.lab.bufferbloat.net/~d/draft-taht-home-gateway-best-practices-00.html
On Mar 10, 2014 1:36 AM, "Muthu Arul Mozhi Perumal (mperumal)" <
mperumal@cisco.com> wrote:

>  |I think that the existing guidance should be sufficient in
> non-multiplexing
>
> |cases (i.e. BUNDLE).
>
>
>
> I don't think so. Even for non-multiplexing cases, a same stream could use
> different drop precedence for different packets, as described in
> draft-dhesikan-tsvwg-rtcweb-qos
>
>
>
>    The combination of priority input and multiple precedence levels
>
>    within a data class provides flexibility for an implementation in
>
>    deciding the importance of the stream and packets within a stream.
>
>    For example, if I frames are more important than the P frames then
>
>    the I frames can be marked with a DSCP with the lower drop
>
>    precedence.
>
>
>
> My take is that STUN connectivity/consent checks SHOULD use the lowest
> drop precedence within a given data type and priority.
>
>
>
> |For consent checks, I think the same DSCP markings as any other ICE
> connectivity
>
> |checks should be used.
>
>
>
> Agree.
>
>
>
> |If multiplexing is being performed, the connectivity checks probably
> should
>
> |use the highest DSCP value being used by the multiplexed media.
>
>
>
> For multiplexed media, I guess there is no WG consensus on how to mark the
> stream, yet (draft-ietf-mmusic-sdp-mux-attributes lists two possible
> options). But, I agree with your proposal in general, since these STUN
> packets are a kind of (media path) signaling packets and signaling packets
> are generally given higher priority..
>
>
>
> Muthu
>
>
>
> *From:* Justin Uberti [mailto:juberti@google.com]
> *Sent:* Monday, March 10, 2014 10:46 AM
> *To:* Muthu Arul Mozhi Perumal (mperumal)
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] DSCP marking for STUN packets
>
>
>
> I think that the existing guidance should be sufficient in
> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think the same
> DSCP markings as any other ICE connectivity checks should be used; for
> ICE-for-SCTP checks, the same DSCP markings as media packets (i.e. the SCTP
> traffic) should be used.
>
>
>
> If multiplexing is being performed, the connectivity checks probably
> should use the highest DSCP value being used by the multiplexed media.
>
>
>
> On Fri, Mar 7, 2014 at 4:35 AM, Muthu Arul Mozhi Perumal (mperumal) <
> mperumal@cisco.com> wrote:
>
> ICE says that an endpoint SHOULD apply the same DSCP marking as media
> packets to connectivity checks. This looks insufficient for RTCWEB, since
> it doesn't cover (at least) the following:
>
> - connectivity checks performed for a stream using different drop
> pecedences (draft-dhesikan-tsvwg-rtcweb-qos).
>
> - connectivity checks performed for non-media/data-channel traffic.
>
> - DSCP markings for consent checks.
>
>
>
> Comments on where we need to specify them? The last one can probably go
> into draft-ietf-rtcweb-stun-consent-freshness?
>
>
>
> Muthu
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--e89a8f642e1e0c5a6004f43cb6f4
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<p dir=3D"ltr">In all this talk about drop precedence and so on I&#39;d lik=
e to point people at some details of some currently deployed packet schedul=
ing and aqm systems.</p>
<p dir=3D"ltr"><a href=3D"http://snapon.lab.bufferbloat.net/~d/draft-taht-h=
ome-gateway-best-practices-00.html">http://snapon.lab.bufferbloat.net/~d/dr=
aft-taht-home-gateway-best-practices-00.html</a></p>
<div class=3D"gmail_quote">On Mar 10, 2014 1:36 AM, &quot;Muthu Arul Mozhi =
Perumal (mperumal)&quot; &lt;<a href=3D"mailto:mperumal@cisco.com">mperumal=
@cisco.com</a>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">






<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|I think that the existing guidance should be sufficient i=
n non-multiplexing
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|cases (i.e. BUNDLE).<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">I don&#39;t think so. Even for non-multiplexing cases, a s=
ame stream could use different drop precedence for different packets, as de=
scribed in draft-dhesikan-tsvwg-rtcweb-qos<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=A0=A0 The combination of priority input and multiple prec=
edence levels<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=A0=A0 within a data class provides flexibility for an imp=
lementation in<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=A0=A0 deciding the importance of the stream and packets w=
ithin a stream.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=A0=A0 For example, if I frames are more important than th=
e P frames then<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=A0=A0 the I frames can be marked with a DSCP with the low=
er drop<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=A0=A0 precedence.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">My take is that STUN connectivity/consent checks SHOULD us=
e the lowest drop precedence within a given data type and priority.<u></u><=
u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|For consent checks, I think the same DSCP markings as any=
 other ICE connectivity
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|checks should be used.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Agree.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|If multiplexing is being performed, the connectivity chec=
ks probably should
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">|use the highest DSCP value being used by the multiplexed =
media.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">For multiplexed media, I guess there is no WG consensus on=
 how to mark the stream, yet (draft-ietf-mmusic-sdp-mux-attributes lists tw=
o possible options). But, I agree with your proposal
 in general, since these STUN packets are a kind of (media path) signaling =
packets and signaling packets are generally given higher priority..<u></u><=
u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Muthu
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><u></u>=A0<u></u></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Justin U=
berti [mailto:<a href=3D"mailto:juberti@google.com" target=3D"_blank">juber=
ti@google.com</a>]
<br>
<b>Sent:</b> Monday, March 10, 2014 10:46 AM<br>
<b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] DSCP marking for STUN packets<u></u><u></u></s=
pan></p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">I think that the existing guidance should be suffici=
ent in non-multiplexing cases (i.e. BUNDLE). For consent checks, I think th=
e same DSCP markings as any other ICE connectivity checks should be used; f=
or ICE-for-SCTP checks, the same DSCP
 markings as media packets (i.e. the SCTP traffic) should be used.<u></u><u=
></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">If multiplexing is being performed, the connectivity=
 checks probably should use the highest DSCP value being used by the multip=
lexed media.<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Fri, Mar 7, 2014 at 4:35 AM, Muthu Arul Mozhi Per=
umal (mperumal) &lt;<a href=3D"mailto:mperumal@cisco.com" target=3D"_blank"=
>mperumal@cisco.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">ICE says that an endpoint SHOULD apply the same DSCP marki=
ng as media packets to connectivity checks. This looks insufficient
 for RTCWEB, since it doesn&#39;t cover (at least) the following:</span><u>=
</u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- connectivity checks performed for a stream using differe=
nt drop pecedences (draft-dhesikan-tsvwg-rtcweb-qos).</span><u></u><u></u><=
/p>

<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- connectivity checks performed for non-media/data-channel=
 traffic.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- DSCP markings for consent checks.</span><u></u><u></u></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Comments on where we need to specify them? The last one ca=
n probably go into draft-ietf-rtcweb-stun-consent-freshness?</span><u></u><=
u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:#888888">=A0</span><span style=3D"color:#888888"><u><=
/u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:#888888">Muthu</span><span style=3D"color:#888888"><u=
></u><u></u></span></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
</div>
</div>
</div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div>

--e89a8f642e1e0c5a6004f43cb6f4--


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Subject: [rtcweb] Fwd: [tsvwg] WG Adoption Call for draft-dhesikan-tsvwg-rtcweb-qos - 24th March 2014
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WG,

For your information the TSVWG has stared an adoption call on
draft-dhesikan-tsvwg-rtcweb-qos. Please consider to support this
adoption call, and indicate your willingness to help working on this
document. Else, this WG will be without one piece of the solution
regarding DSCP usage that we clearly had interest before in.

Cheers

Magnus Westerlund
WG Chair

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The TSVWG Chairs are considering adoption of the above draft. This draft
has been discussed at previous IETF meetings and on the list.

At the recent London IETF meeting it received strong support from the
people present in the meeting room, and no comments against adopting this
draft as a TSVWG work item.

The WG chairs now invite any further comments on this adoption. If there
are concerns or comments about adoption please send comments to the list
or to the Chairs. We plan to make a final decision on this adoption on
24th March 2014.

Gorry, James, and David
TSVWG Co-Chairs.



--------------010105000008060306090105--


From nobody Mon Mar 10 06:59:22 2014
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To: Christer Holmberg <christer.holmberg@ericsson.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
References: <CAOJ7v-063mGE_zndtAqAMN2fJw7kWvUHX2SgyDMryfw_-aS-9w@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D1F1892@ESESSMB209.ericsson.se>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On 2014-03-10 09:24, Christer Holmberg wrote:
> Hi,
> 
>  
> 
> Before I comment on your proposal, let’s take a few steps back (maybe it
> can even solve your issue).
> 
>  
> 
> BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp
> attributes. I remember we had discussions about that, but I don’t
> remember the justification at the moment. But, I wonder whether that
> text is from before we made a decision that, unless both endpoints
> support BUNDLE, no endpoint will use a single address:port.
> 
>  
> 
> So, the question is: do we really need the SDP rtcp attribute?

I think it is a question of how you want the fallback to work. In the
case of bundle only lines, a non  bundle supporting peer will reject
them, thus no issues with what is written around RTCP-mux and a=rtcp in
those m= blocks. However, the first one if you want that to support
fallback that makes sense then you will need both a=rtcp-mux and an
a=rtcp (with a different port) to handle that with optimal backwards
compatibility.

Regarding the motivation for a=rtcp-mux and a=rtcp, I think it is
reasonably straight forward. We want a=rtcp-mux to minimize port usage,
i.e. ports=1. Not supporting a=rtcp-mux with bundle that would mean two
ports (RTP and RTCP) for the bundle group. To get the backwards
compatibility the usage of a=rtcp would be required.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Mon Mar 10 07:18:57 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

>> Before I comment on your proposal, let's take a few steps back (maybe=20
>> it can even solve your issue). =20
>>=20
>> BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp=20
>> attributes. I remember we had discussions about that, but I don't=20
>> remember the justification at the moment. But, I wonder whether that=20
>> text is from before we made a decision that, unless both endpoints=20
>> support BUNDLE, no endpoint will use a single address:port.
>>=20
>> =20
>>=20
>> So, the question is: do we really need the SDP rtcp attribute?
>
> I think it is a question of how you want the fallback to work. In the cas=
e of bundle only lines, a non  bundle supporting peer will reject them, thu=
s=20
> no issues with what is written around RTCP-mux and a=3Drtcp in those m=3D=
 blocks. However, the first one if you want that to support fallback that=20
> makes sense then you will need both a=3Drtcp-mux and an a=3Drtcp (with a =
different port) to handle that with optimal backwards compatibility.
>
> Regarding the motivation for a=3Drtcp-mux and a=3Drtcp, I think it is rea=
sonably straight forward. We want a=3Drtcp-mux to minimize port usage,=20
> i.e. ports=3D1. Not supporting a=3Drtcp-mux with bundle that would mean t=
wo ports (RTP and RTCP) for the bundle group. To get the backwards=20
> compatibility the usage of a=3Drtcp would be required.

Keep in mind that the offerer must always be prepared that the answerer doe=
s not support the rtcp-mux attribute (or the rtcp attribute).

So, at least my understanding is: no matter whether BUNDLE is used or not, =
if the answer does not contain a rtcp-mux (e.g. in case of fallback) attrib=
ute, the offerer must use the default "rtp+1" port.

Regards,

Christer


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Subject: Re: [rtcweb] DSCP marking for STUN packets
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Hi,
(as individual)

I think we need to start with what the goals are with the STUN packets.

In the initial connectivity establishment of ICE the purpose is to find
paths that works. To make that determination at all, I would argue to
maximize success one should use best effort markings as the network is
least likely to do something strange with such packets, including
dropping them for violating policy (even if they should just be
downgraded).

If one like to use STUN to determine the possibility to get DSCP != BE
packets through the network from a peer to peer perspective, then I
think one should consider specifying something like the STUN ECN check
options that are documented in Section 7.2.2 in RFC 6679.

To summarize, this sends an additional round of STUN packets after a
working path has been found to check ECN compliance in this path. It
includes and option indicating the ECT value entered, and the receiver
echoes the received value back. Thus enabling a sender to determine that
the path is ECT.

Something similar could be specified if desired to determine that at
least the network doesn't drop packets with DSCP != BE.

For consent freshness I don't think we need to do that on a per traffic
marking type level. Thus, the important is that it reaches the peer. And
the packet reaching the just needs to be statistically high enough
probability over a sequence of checks. Thus, I would say that the only
important factor here is to not mark it so that it has significant
higher drop probability. It also don't need to be minimal delay, thus EF
appears overkill and although AFx could be used, that should rather be
based on a traffic mix with a lot of AFx traffic.

In the end, it might be easier to not use other DSCP marking that BE for
the STUN, simple because it gives the most robust behavior and not
subject to network issues due to policing or DSCP remarking.

I think it will be important to take the considerations for using
multiple DSCP on the same 5-tuple that DART will document into account
here.

Cheers

Magnus Westerlund

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To: Christer Holmberg <christer.holmberg@ericsson.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On 2014-03-10 15:13, Christer Holmberg wrote:
> Hi,
> 
>>> Before I comment on your proposal, let's take a few steps back
>>> (maybe it can even solve your issue).
>>> 
>>> BUNDLE currently mandates the usage of both the SDP rtcp-mux and
>>> rtcp attributes. I remember we had discussions about that, but I
>>> don't remember the justification at the moment. But, I wonder
>>> whether that text is from before we made a decision that, unless
>>> both endpoints support BUNDLE, no endpoint will use a single
>>> address:port.
>>> 
>>> 
>>> 
>>> So, the question is: do we really need the SDP rtcp attribute?
>> 
>> I think it is a question of how you want the fallback to work. In
>> the case of bundle only lines, a non  bundle supporting peer will
>> reject them, thus no issues with what is written around RTCP-mux
>> and a=rtcp in those m= blocks. However, the first one if you want
>> that to support fallback that makes sense then you will need both
>> a=rtcp-mux and an a=rtcp (with a different port) to handle that
>> with optimal backwards compatibility.
>> 
>> Regarding the motivation for a=rtcp-mux and a=rtcp, I think it is
>> reasonably straight forward. We want a=rtcp-mux to minimize port
>> usage, i.e. ports=1. Not supporting a=rtcp-mux with bundle that
>> would mean two ports (RTP and RTCP) for the bundle group. To get
>> the backwards compatibility the usage of a=rtcp would be required.
> 
> Keep in mind that the offerer must always be prepared that the
> answerer does not support the rtcp-mux attribute (or the rtcp
> attribute).

Yes, that was what I meant with fallback in the above text. What you do
in case the peer doesn't support bundle.

> 
> So, at least my understanding is: no matter whether BUNDLE is used or
> not, if the answer does not contain a rtcp-mux (e.g. in case of
> fallback) attribute, the offerer must use the default "rtp+1" port.

I have a tendency to agree, which means that the first m= block in a
bunde-only group will be required to contain a=rtcp and if RTCP mux is
desired also that attribute.

I think a=rtcp could be skipped for a bound-only m= block due to the
requirement on supporting a=rtcp-mux if you do bundle.

Cheers

Magnus Westerlund

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

>>>> Before I comment on your proposal, let's take a few steps back=20
>>>> (maybe it can even solve your issue).
>>>>=20
>>>> BUNDLE currently mandates the usage of both the SDP rtcp-mux and=20
>>>> rtcp attributes. I remember we had discussions about that, but I=20
>>>> don't remember the justification at the moment. But, I wonder=20
>>>> whether that text is from before we made a decision that, unless=20
>>>> both endpoints support BUNDLE, no endpoint will use a single=20
>>>> address:port.
>>>>=20
>>>>=20
>>>>=20
>>>> So, the question is: do we really need the SDP rtcp attribute?
>>>=20
>>> I think it is a question of how you want the fallback to work. In the=20
>>> case of bundle only lines, a non  bundle supporting peer will reject=20
>>> them, thus no issues with what is written around RTCP-mux and a=3Drtcp=
=20
>>> in those m=3D blocks. However, the first one if you want that to=20
>>> support fallback that makes sense then you will need both a=3Drtcp-mux=
=20
>>> and an a=3Drtcp (with a different port) to handle that with optimal=20
>>> backwards compatibility.
>>>=20
>>> Regarding the motivation for a=3Drtcp-mux and a=3Drtcp, I think it is=20
>>> reasonably straight forward. We want a=3Drtcp-mux to minimize port=20
>>> usage, i.e. ports=3D1. Not supporting a=3Drtcp-mux with bundle that wou=
ld=20
>>> mean two ports (RTP and RTCP) for the bundle group. To get the=20
>>> backwards compatibility the usage of a=3Drtcp would be required.
>>=20
>> Keep in mind that the offerer must always be prepared that the=20
>> answerer does not support the rtcp-mux attribute (or the rtcp=20
>> attribute).
>
> Yes, that was what I meant with fallback in the above text. What you do i=
n case the peer doesn't support bundle.
>=20
>> So, at least my understanding is: no matter whether BUNDLE is used or=20
>> not, if the answer does not contain a rtcp-mux (e.g. in case of
>> fallback) attribute, the offerer must use the default "rtp+1" port.
>
> I have a tendency to agree, which means that the first m=3D block in a bu=
nde-only group will be required to contain a=3Drtcp and if RTCP mux is desi=
red also that attribute.
>
> I think a=3Drtcp could be skipped for a bound-only m=3D block due to the =
requirement on supporting a=3Drtcp-mux if you do bundle.

For bundle-only, I actually think BOTH can be skipped, because bundle-only =
will by default use whatever address:port is negotiated - both for RTP and =
RTCP.

But, for non-bundle-only, I still ask whether we really need a=3Drtcp. Isn'=
t a=3Drtcp-mux enough? If the answerer supports it, you will use it, otherw=
ise you will use rtp+1.

Otherwise, assume that the answerer supports BOTH a=3Drtcp and a=3Drtcp-mux=
. Which has higher "priority"? How does the answerer knows whether the offe=
rer wants to use rtcp-mux, or whether it wants to use whatever port is indi=
cated in a=3Drtcp?

Regards,

Christer


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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 2014-03-08 07:48, Martin Thomson wrote:
> On 7 March 2014 20:14, Dan Wing <dwing@cisco.com> wrote:
>> About half a year ago, AVTCORE published an updated recommendation
>> for CNAME as RFC7022.  Is its guidance insufficient?
> 
> Necessary, but not sufficient.  Unfortunately, the definition of a 
> "session", while sufficient for the description in 7022, is not
> quite precise enough for this use case.  The implication there is
> that it means "RTP session", which is both not at the right level of 
> granularity, and not directly actionable.

So, within a PeerConnection one have one or more RTP sessions, depending
on how one configures it. That RTP session MAY be interconnected with
other PeerConnection's RTP sessions by a middlebox.

As long as this is different end-points PCs there is little issues. The
interesting case here is if one end-point connects multiple PCs to the
same middlebox and starts seeing its own traffic due to the middlebox
not being built for this usage and the JS programmer thinks this is okay
to do.

The end-point can always detect this, as it knows all its used CNAMEs,
and can note that they are looped back to itself if this check is
implemented. If one have the same CNAME is both PC's RTP session
instances the RTP stack is more likely to detect this without code
changes as a loop.

The only reason I can see for having multiple PCs being connected to the
same middlebox would be if that makes the setup of QoS simpler, but I
think MST individual RTP sessions within the context of one PC is more
straight forward to ensure that one have unique 5-tuples per media stream.

> 
> I also note this little gem:
> 
> A longer-term persistent RTCP CNAME is sometimes useful to
> facilitate third-party monitoring, consistent with [RFC3550].

Yes, but that is clearly not suitable for WebRTC.

Cheers

Magnus Westerlund

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On 10 March 2014 15:42, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
>> A longer-term persistent RTCP CNAME is sometimes useful to
>> facilitate third-party monitoring, consistent with [RFC3550].
>
> Yes, but that is clearly not suitable for WebRTC.

I'm sure that's not the only thing it's unsuitable for.


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On 2014-03-10 15:44, Martin Thomson wrote:
> On 10 March 2014 15:42, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>>> A longer-term persistent RTCP CNAME is sometimes useful to
>>> facilitate third-party monitoring, consistent with [RFC3550].
>>
>> Yes, but that is clearly not suitable for WebRTC.
> 
> I'm sure that's not the only thing it's unsuitable for.
> 

No, but there are cases where this can be a desired property in someone
building an application. For example an astronomy sensory network that
deliver sensor data over RTP. In that case you are not talking about
human user's endpoints, and you also may have a private network and want
to make monitoring at key points simple. Thus having the sensors being
provided a persistent name is simple and straight forward.

cheers

Magnus Westerlund

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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On 2014-03-10 15:38, Christer Holmberg wrote:
> Hi,
> 
>>>>> Before I comment on your proposal, let's take a few steps
>>>>> back (maybe it can even solve your issue).
>>>>> 
>>>>> BUNDLE currently mandates the usage of both the SDP rtcp-mux
>>>>> and rtcp attributes. I remember we had discussions about
>>>>> that, but I don't remember the justification at the moment.
>>>>> But, I wonder whether that text is from before we made a
>>>>> decision that, unless both endpoints support BUNDLE, no
>>>>> endpoint will use a single address:port.
>>>>> 
>>>>> 
>>>>> 
>>>>> So, the question is: do we really need the SDP rtcp
>>>>> attribute?
>>>> 
>>>> I think it is a question of how you want the fallback to work.
>>>> In the case of bundle only lines, a non  bundle supporting peer
>>>> will reject them, thus no issues with what is written around
>>>> RTCP-mux and a=rtcp in those m= blocks. However, the first one
>>>> if you want that to support fallback that makes sense then you
>>>> will need both a=rtcp-mux and an a=rtcp (with a different port)
>>>> to handle that with optimal backwards compatibility.
>>>> 
>>>> Regarding the motivation for a=rtcp-mux and a=rtcp, I think it
>>>> is reasonably straight forward. We want a=rtcp-mux to minimize
>>>> port usage, i.e. ports=1. Not supporting a=rtcp-mux with bundle
>>>> that would mean two ports (RTP and RTCP) for the bundle group.
>>>> To get the backwards compatibility the usage of a=rtcp would be
>>>> required.
>>> 
>>> Keep in mind that the offerer must always be prepared that the 
>>> answerer does not support the rtcp-mux attribute (or the rtcp 
>>> attribute).
>> 
>> Yes, that was what I meant with fallback in the above text. What
>> you do in case the peer doesn't support bundle.
>> 
>>> So, at least my understanding is: no matter whether BUNDLE is
>>> used or not, if the answer does not contain a rtcp-mux (e.g. in
>>> case of fallback) attribute, the offerer must use the default
>>> "rtp+1" port.
>> 
>> I have a tendency to agree, which means that the first m= block in
>> a bunde-only group will be required to contain a=rtcp and if RTCP
>> mux is desired also that attribute.
>> 
>> I think a=rtcp could be skipped for a bound-only m= block due to
>> the requirement on supporting a=rtcp-mux if you do bundle.
> 
> For bundle-only, I actually think BOTH can be skipped, because
> bundle-only will by default use whatever address:port is negotiated -
> both for RTP and RTCP.

No, I see a path of grief to not explicitly signal things when you use
them. I also thought the a=rtcp-mux is a MUST to implement, not a MUST use.

> 
> But, for non-bundle-only, I still ask whether we really need a=rtcp.
> Isn't a=rtcp-mux enough? If the answerer supports it, you will use
> it, otherwise you will use rtp+1.

My understanding is that a=rtcp has been required in any non-private
network deployments due to NATs for the last 10 years. Thus, I don't see
how you can remove them and expect it to work, unless you are doing
a=rtcp-mux. Thus, my view would be that you can remove a=rtcp if you
know that the peer supprots a=rtcp-mux.

> 
> Otherwise, assume that the answerer supports BOTH a=rtcp and
> a=rtcp-mux. Which has higher "priority"? How does the answerer knows
> whether the offerer wants to use rtcp-mux, or whether it wants to use
> whatever port is indicated in a=rtcp?

That is a=rtcp-mux per RFC 5761 that has higher priority, please see
Section 5.1.1.

cheers

Magnus Westerlund

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----------------------------------------------------------------------
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 2014-03-05 17:07, Martin Thomson wrote:
> On 5 March 2014 14:30, Justin Uberti <juberti@google.com> wrote:
>> I am inclined to make CNAMEs per-PeerConnection (i.e. enforce scenario #2
>> behavior) for 1.0, as it has a smaller downside.
> 
> I think that I am OK with this.  It might still be possible to
> synchronize in the first scenario, but it would require that the
> playback protect itself against drift, because CNAME wouldn't be
> providing a positive indication that synchronization is safe.
> 

Please slow down here,

Can you please think through the linkage issue a bit first before
changing this property. I know that it might appear simple to mandate
different CNAMES on a Per PC basis, but I do think it has downsides.

One has to do with loop detection between the PCs, i.e. determine if one
receives one self's MSTs. The next one is that it will prevent someone
to easily determine if the MST in two different PCs are actually sharing
the same synchronization context. As the default really needs to not
synchronize RTP media with different SSRCs, having the JS app direct the
browser to please consider some set of CNAMEs as being equal will be a
problematic one, and require API surface.

I want to understand what issue with linkages that we have here before
determining what to do. So, different application contexts, i.e.
different browser TABs on an end-point will have different CNAMEs, thus
different applications will not allow linking between them. The linkage
we have is between PCs originating in the same JS application instance.
In cases where a set of end-points participate in a conference, they
will see only one CNAME from this endpoint.

I guess the linkage issue appears when one application have the endpoint
participate in two or more conferences either simultaneous or in
sequence (assuming that CNAME will be persistent as long as some PC
exists or during the JS application lifetime). In that case an external
endpoint participating in more than one of these conferences can see the
same CNAME appears in both conferences. And thus link any input
provided, if the applications doesn't otherwise reveal the user identity
etc. anyway.

I have a hard time determine the impact of this linkage issue, as it
will depend on the application context, and also the CNAMEs persistence
over time within a JS APP context.

Cheers

Magnus Westerlund

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On 2014-03-06 10:56, Huangyihong (Rachel) wrote:
> Hi,
> 
>  
> 
> draft-ietf-rtcweb-rtp-usage-12 doesn’t have any description regarding to
> RTP ECN RFC6679. I’m wondering why. But from my point of view, it’s
> worth some discussion. The usage of ECN is a trend though it is not
> universal in current network. Supporting it in browsers for RTP media
> transport may be good for future.
> 

Hi Rachel,
(as individual/author)

We did consider ECN for RTP and it was discussed a bit at the Stockholm
interim meeting in June 2012. My recollection of this is that the
complexity trade-off versus the potential gains and the risk for it
causing negative impact resulted in that we did not included it at this
stage.

I do think adding ECN would be a good thing, but we do need to know that
it is working reasonably well and provide a benefit sufficiently often.
I think the importance of AQM will enable more deployment of ECN also.
Thus enabling improved performance for real-time applications like
WebRTC based ones.

I would suggest that we revisit this question after the core set of
specifications have been done.

Cheers

Magnus Westerlund

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On 2014-03-06 14:34, Simon Perreault wrote:
> Le 2014-03-06 13:23, Justin Uberti a Ã©crit :
>> can we agree that TURN TCP candidates are a SHOULD NOT?
> 
> Not a SHOULD, sure. SHOULD NOT, no.
> 

I would guess that the simplest is to remove discussion of TURN TCP
altogether from Transports. That would not recommend it nor make it
disallowed. If we want to be explicit, then simply motivate why we don't
believe it necessary.

Justin, what is the reason you wanted SHOULD NOT?

cheers

Magnus Westerlund

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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 10 March 2014 16:19, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> Can you please think through the linkage issue a bit first before
> changing this property.

We already came to the conclusion that if media was being forwarded by
a peer, then that is new media originated by that peer, not the same
media.  Thus, if looping occurs, it is at the application layer.  We
definitely already agreed that RTP forwarding is not a function that a
browser will support.

Obviously, if we add that feature, CNAMEs will be just one of the
manifold issues that will arise and we will have to reconsider.
Compromising the privacy of users doesn't seem like a good trade-off
against some minor advantages in terms of loop detection (unlikely)
and synchronization (easy to apply manually).


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Date: Mon, 10 Mar 2014 16:57:56 +0100
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To: Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] On ptimes and maxptimes
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Hi,

(As individual)

On 2014-03-05 17:26, Justin Uberti wrote:
> From the session this morning, I recall this consensus:
> 
>   * We will add recommendations on the frame sizes to use for PCMU/A to
>     the rtcweb audio codec I-D (20, 30, 60 ms frames).

The above applies to sending payloads

Yes, but that was given that a receiver will be capable of receiving any
valid number of frames, or samples within a reasonable packetization,
and I think that should be 120 ms.

>   * We will consider adding an a=maxptime attribute that represents the
>     "minimum maxptime" of all the offered codecs. That is, if both Opus
>     (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a
>     maxptime=60 SHOULD be included.
>       o Since maxptime spans all codecs, this means that some modes
>         (e.g. Opus 120ms) will be unavailable, unless only Opus is offered.

I think you base the assumption on maxptime based on the sending
recommendations, not what is capable of receiving. If one is capable of
receiving 200 ms of a-law and Opus, then you would need to say 120 ms as
Opus will limit you.

>   * We will consider adding an a=ptime attribute, set to the receiver's
>     preferred frame size to receive. Again, this value spans all codecs,
>     so the specified value may not be viable for all codecs (e.g. 30 ms
>     works for PCMU but not for Opus)
> 
> After thinking about this some, I'm not sure the maxptime/ptime values
> will lead to any different behavior than if they were not specified.
> Since there is a complexity cost (especially if the values need to
> change based on which codecs are offered), is there a clear upside here?

The maxptime values may in some cases be limited downwards for
interoperability, for example if one like to gateway AMR into a CS
network then the gateway might indicate a maxptime that is less, more
like 20 or 40 ms.

Please remember that these rules do apply to any codec, not only the MTI
ones.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

>>>>>> Before I comment on your proposal, let's take a few steps back=20
>>>>>> (maybe it can even solve your issue).
>>>>>>=20
>>>>>> BUNDLE currently mandates the usage of both the SDP rtcp-mux and=20
>>>>>> rtcp attributes. I remember we had discussions about that, but I=20
>>>>>> don't remember the justification at the moment.
>>>>>> But, I wonder whether that text is from before we made a decision=20
>>>>>> that, unless both endpoints support BUNDLE, no endpoint will use a=20
>>>>>> single address:port.
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> So, the question is: do we really need the SDP rtcp attribute?
>>>>>=20
>>>>> I think it is a question of how you want the fallback to work.
>>>>> In the case of bundle only lines, a non  bundle supporting peer=20
>>>>> will reject them, thus no issues with what is written around=20
>>>>> RTCP-mux and a=3Drtcp in those m=3D blocks. However, the first one if=
=20
>>>>> you want that to support fallback that makes sense then you will=20
>>>>> need both a=3Drtcp-mux and an a=3Drtcp (with a different port) to=20
>>>>> handle that with optimal backwards compatibility.
>>>>>=20
>>>>> Regarding the motivation for a=3Drtcp-mux and a=3Drtcp, I think it is=
=20
>>>>> reasonably straight forward. We want a=3Drtcp-mux to minimize port=20
>>>>> usage, i.e. ports=3D1. Not supporting a=3Drtcp-mux with bundle that=20
>>>>> would mean two ports (RTP and RTCP) for the bundle group.
>>>>> To get the backwards compatibility the usage of a=3Drtcp would be=20
>>>>> required.
>>>>=20
>>>> Keep in mind that the offerer must always be prepared that the=20
>>>> answerer does not support the rtcp-mux attribute (or the rtcp=20
>>>> attribute).
>>>=20
>>> Yes, that was what I meant with fallback in the above text. What you=20
>>> do in case the peer doesn't support bundle.
>>>=20
>>>> So, at least my understanding is: no matter whether BUNDLE is used=20
>>>> or not, if the answer does not contain a rtcp-mux (e.g. in case of=20
>>>> fallback) attribute, the offerer must use the default "rtp+1" port.
>>>=20
>>> I have a tendency to agree, which means that the first m=3D block in a=
=20
>>> bunde-only group will be required to contain a=3Drtcp and if RTCP mux=20
>>> is desired also that attribute.
>>>=20
>>> I think a=3Drtcp could be skipped for a bound-only m=3D block due to th=
e=20
>>> requirement on supporting a=3Drtcp-mux if you do bundle.
>>=20
>> For bundle-only, I actually think BOTH can be skipped, because=20
>> bundle-only will by default use whatever address:port is negotiated -=20
>> both for RTP and RTCP.
>
> No, I see a path of grief to not explicitly signal things when you use th=
em.=20

But, the whole idea of using port zero in bundle-only was that, if the answ=
erer supports BUNDLE, port zero will be replaced with the negotiated addres=
s:port.

When you send the offer, you don't yet know which address:port will be sele=
cted, so there is no idea to insert a=3Drtcp. Inserting a=3Drtcp-mux probab=
ly doesn't harm, though, so I won't argue about it :)

>I also thought the a=3Drtcp-mux is a MUST to implement, not a MUST use.

Currently the draft says MUST use. But, that may be a mistake, or a leftove=
r from previous procedures. I agree that one should be able to use BUNDLE a=
lso without rtcp-mux (i.e. using separate BUNDLE ports for the RTP traffic =
and the RTCP traffic).
=20
>> But, for non-bundle-only, I still ask whether we really need a=3Drtcp.
>> Isn't a=3Drtcp-mux enough? If the answerer supports it, you will use it,=
=20
>> otherwise you will use rtp+1.
>
>My understanding is that a=3Drtcp has been required in any non-private net=
work deployments due to >NATs for the last 10 years. Thus, I don't see how =
you can remove them and expect it to work, unless >you are doing a=3Drtcp-m=
ux. Thus, my view would be that you can remove a=3Drtcp if you know that th=
e >peer supprots a=3Drtcp-mux.

Are you saying that we should use a=3Drtcp ONLY to indicate the RTP port, i=
n case the answerer does not support a=3Drtcp-mux?
=20
>> Otherwise, assume that the answerer supports BOTH a=3Drtcp and=20
>> a=3Drtcp-mux. Which has higher "priority"? How does the answerer knows=20
>> whether the offerer wants to use rtcp-mux, or whether it wants to use=20
>> whatever port is indicated in a=3Drtcp?
>
> That is a=3Drtcp-mux per RFC 5761 that has higher priority, please see Se=
ction 5.1.1.

You are right - my mistake.

But, never the less, if the offerer sends both a=3Drtcp-mux and a=3Drtcp, u=
nless a=3Drtcp points to the rtp port, the offerer needs to be prepared to =
use 3 different ports for RTCP (rtp+1 port, rtp port and rtp+<a=3Drtcp valu=
e>), because it does not know which attribute (if any) the answerer support=
s, and which (if any) the answerer is willing to use.

Regards,

Christer


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At 12:16 AM 3/10/2014, Justin Uberti wrote:
>I think that the existing guidance should be sufficient in 
>non-multiplexing cases (i.e. BUNDLE). For consent checks, I think 
>the same DSCP markings as any other ICE connectivity checks should 
>be used; for ICE-for-SCTP checks, the same DSCP markings as media 
>packets (i.e. the SCTP traffic) should be used.
>
>If multiplexing is being performed, the connectivity checks probably 
>should use the highest DSCP value being used by the multiplexed media.

why is (seemingly) everyone's default position "use the highest DSCP 
available" for signaling packets?

You just need to make sure the packets aren't starved or dropped 
by/at congestion points.

James



>On Fri, Mar 7, 2014 at 4:35 AM, Muthu Arul Mozhi Perumal (mperumal) 
><<mailto:mperumal@cisco.com>mperumal@cisco.com> wrote:
>
>ICE says that an endpoint SHOULD apply the same DSCP marking as 
>media packets to connectivity checks. This looks insufficient for 
>RTCWEB, since it doesn't cover (at least) the following:
>
>- connectivity checks performed for a stream using different drop 
>pecedences (draft-dhesikan-tsvwg-rtcweb-qos).
>
>- connectivity checks performed for non-media/data-channel traffic.
>
>- DSCP markings for consent checks.
>
>
>
>Comments on where we need to specify them? The last one can probably 
>go into draft-ietf-rtcweb-stun-consent-freshness?
>
>
>
>Muthu
>
>_______________________________________________
>rtcweb mailing list
><mailto:rtcweb@ietf.org>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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> I would suggest that we revisit this question after the core set of
> specifications have been done.

+1
=3D> ECN for rtcweb could be great.

Cheers,
Michael


>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Bernard Aboba <bernard.aboba@gmail.com>
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+1 to what Magnus said.  ECN for RTP isn't mature enough to make this a
WebRTC requirement at this time.


On Mon, Mar 10, 2014 at 8:28 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-06 10:56, Huangyihong (Rachel) wrote:
> > Hi,
> >
> >
> >
> > draft-ietf-rtcweb-rtp-usage-12 doesn't have any description regarding t=
o
> > RTP ECN RFC6679. I'm wondering why. But from my point of view, it's
> > worth some discussion. The usage of ECN is a trend though it is not
> > universal in current network. Supporting it in browsers for RTP media
> > transport may be good for future.
> >
>
> Hi Rachel,
> (as individual/author)
>
> We did consider ECN for RTP and it was discussed a bit at the Stockholm
> interim meeting in June 2012. My recollection of this is that the
> complexity trade-off versus the potential gains and the risk for it
> causing negative impact resulted in that we did not included it at this
> stage.
>
> I do think adding ECN would be a good thing, but we do need to know that
> it is working reasonably well and provide a benefit sufficiently often.
> I think the importance of AQM will enable more deployment of ECN also.
> Thus enabling improved performance for real-time applications like
> WebRTC based ones.
>
> I would suggest that we revisit this question after the core set of
> specifications have been done.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">+1 to what Magnus said. &nbsp;ECN for RTP isn&#39;t mature=
 enough to make this a WebRTC requirement at this time.&nbsp;</div><div cla=
ss=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Mar 10, 2014 =
at 8:28 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnu=
s.westerlund@ericsson.com" target=3D"_blank">magnus.westerlund@ericsson.com=
</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"HOEnZb"><div class=3D"h5">On 2=
014-03-06 10:56, Huangyihong (Rachel) wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; draft-ietf-rtcweb-rtp-usage-12 doesn&rsquo;t have any description rega=
rding to<br>
&gt; RTP ECN RFC6679. I&rsquo;m wondering why. But from my point of view, i=
t&rsquo;s<br>
&gt; worth some discussion. The usage of ECN is a trend though it is not<br=
>
&gt; universal in current network. Supporting it in browsers for RTP media<=
br>
&gt; transport may be good for future.<br>
&gt;<br>
<br>
</div></div>Hi Rachel,<br>
(as individual/author)<br>
<br>
We did consider ECN for RTP and it was discussed a bit at the Stockholm<br>
interim meeting in June 2012. My recollection of this is that the<br>
complexity trade-off versus the potential gains and the risk for it<br>
causing negative impact resulted in that we did not included it at this<br>
stage.<br>
<br>
I do think adding ECN would be a good thing, but we do need to know that<br=
>
it is working reasonably well and provide a benefit sufficiently often.<br>
I think the importance of AQM will enable more deployment of ECN also.<br>
Thus enabling improved performance for real-time applications like<br>
WebRTC based ones.<br>
<br>
I would suggest that we revisit this question after the core set of<br>
specifications have been done.<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | Phone=
 &nbsp;<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7=
148287</a><br>
F=E4r=F6gatan 6 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | M=
obile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+46 73 09=
49079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--f46d0444eb67bcf47a04f44d3b7c--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Mon, 10 Mar 2014 21:43:35 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Christer said:

"BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp
attributes. I remember we had discussions about that, but I don't remember
the justification at the moment."

[BA] The justification was that an implementation interested in
multiplexing A/V would almost certainly also want to multiplex RTP and
RTCP, because the motivation for both is minimizing port usage.


On Mon, Mar 10, 2014 at 1:24 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> Before I comment on your proposal, let's take a few steps back (maybe it
> can even solve your issue).
>
>
>
> BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp
> attributes. I remember we had discussions about that, but I don't remember
> the justification at the moment. But, I wonder whether that text is from
> before we made a decision that, unless both endpoints support BUNDLE, no
> endpoint will use a single address:port.
>
>
>
> So, the question is: do we really need the SDP rtcp attribute?
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
>
>
>
>
>
>
> *From:* Justin Uberti [mailto:juberti@google.com]
> *Sent:* 10. maaliskuuta 2014 8:18
> *To:* rtcweb@ietf.org; Eric Rescorla; Christer Holmberg
> *Subject:* BUNDLE with implicit rtcp-mux
>
>
>
> In my JSEP preso during the RTCWEB session last Wednesday, I discussed
> adding a new BUNDLE policy to JSEP. This policy, which would be called
> something like "force-bundle", would force the use of the same ports for
> BUNDLEd m-sections, as well as RTCP mux. When one knows a priori that the
> other side supports BUNDLE, this would reduce candidate gathering even over
> the currently defined "max-bundle" policy, since there would be no need to
> gather RTCP ports at all, and also avoid the need for the BUNDLE fixup
> offer (where the ports are updated to be the same for all BUNDLEd
> m-sections).
>
> To explain the differences here, here are some example offers for an
> audio+video call:
>
> Default bundle policy - separate ports for audio, video, RTP and RTCP:
> m=audio 10000
> a=rtcp:10001
> a=rtcp-mux
> m=video 11000
> a=rtcp:11001
> a=rtcp-mux
>
> Existing "max-bundle" policy - video port is 0 because bundle-only, but
> both RTP and RTCP ports allocated for audio:
> m=audio 10000
> a=rtcp:10001
> a=rtcp-mux
> m=video 0
> a=bundle-only
> a=rtcp-mux
>
> The new "force-bundle" policy - only a single set of ports allocated -
> audio, video, RTP, RTCP share the same port.
> m=audio 10000
> a=rtcp:10000
> a=rtcp-mux
> m=video 10000
> a=rtcp:10000
> a=rtcp-mux
>
> During the discussion, the ability to avoid RTCP port gathering was seen
> as valuable, but not the ability to avoid the fixup offer. However, after
> thinking about it a bit, I don't think that there is a viable solution that
> avoids RTCP port gathering that *doesn't* use a single set of ports. The
> only other option would be to do something like this:
>
> m=audio 10000
> a=rtcp:10000 # same as m=audio port
> a=rtcp-mux
> m=video 0
> a=bundle-only
> a=rtcp-mux
>
> but this will almost surely will result in a malfunction when talking to a
> non-rtcp-mux client, due to the same port being used for RTP and RTCP, and
> still requires the fixup offer, and as such is net worse than the
> force-bundle approach indicated above.
>
> So I think we the force-bundle approach is the one we want to minimize
> port gathering. The only question is whether we should keep max-bundle as
> well. To be specific, we need to decide between:
>
> a) Keep max-bundle as it is (separate RTP and RTCP ports allocated), and
> add a new force-bundle policy that creates an offer with a single port
> shared between all m-sections (RTP and RTCP). max-bundle remains nominally
> backwards compatible with non-BUNDLE/non-rtcp-mux endpoints, in that it
> will still send a single audio stream. force-bundle is not backwards
> compatible with non-BUNDLE endpoints.
>
> b) Add force-bundle, but get rid of max-bundle.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Christer said:&nbsp;<div><br></div><div>&quot;<span style=
=3D"font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">BUN=
DLE currently mandates the usage of both the SDP rtcp-mux and rtcp attribut=
es. I remember we had discussions about that, but I don&rsquo;t remember th=
e justification at the moment.</span>&quot;</div>

<div><br></div><div>[BA] The justification was that an implementation inter=
ested in multiplexing A/V would almost certainly also want to multiplex RTP=
 and RTCP, because the motivation for both is minimizing port usage.&nbsp;<=
/div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon,=
 Mar 10, 2014 at 1:24 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Before I comment on your =
proposal, let&rsquo;s take a few steps back (maybe it can even solve your i=
ssue).<u></u><u></u></span></p>


<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">BUNDLE currently mandates=
 the usage of both the SDP rtcp-mux and rtcp attributes. I remember we had =
discussions about that, but I don&rsquo;t remember the justification
 at the moment. But, I wonder whether that text is from before we made a de=
cision that, unless both endpoints support BUNDLE, no endpoint will use a s=
ingle address:port.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">So, the question is: do w=
e really need the SDP rtcp attribute?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Justin U=
berti [mailto:<a href=3D"mailto:juberti@google.com" target=3D"_blank">juber=
ti@google.com</a>]
<br>
<b>Sent:</b> 10. maaliskuuta 2014 8:18<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a>; Eric Rescorla; Christer Holmberg<br>
<b>Subject:</b> BUNDLE with implicit rtcp-mux<u></u><u></u></span></p><div>=
<div class=3D"h5">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<div>
<p class=3D"MsoNormal">In my JSEP preso during the RTCWEB session last Wedn=
esday, I discussed adding a new BUNDLE policy to JSEP. This policy, which w=
ould be called something like &ldquo;force-bundle&rdquo;, would
 force the use of the same ports for BUNDLEd m-sections, as well as RTCP mu=
x. When one knows a priori that the other side supports BUNDLE, this would =
reduce candidate gathering even over the currently defined &ldquo;max-bundl=
e&rdquo; policy, since there would be no need
 to gather RTCP ports at all, and also avoid the need for the BUNDLE fixup =
offer (where the ports are updated to be the same for all BUNDLEd m-section=
s).<u></u><u></u></p>
<p class=3D"MsoNormal">To explain the differences here, here are some examp=
le offers for an audio+video call:<u></u><u></u></p>
<p class=3D"MsoNormal">Default bundle policy - separate ports for audio, vi=
deo, RTP and RTCP:<br>
m=3Daudio 10000<br>
a=3Drtcp:10001<br>
a=3Drtcp-mux<br>
m=3Dvideo 11000<br>
a=3Drtcp:11001<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">Existing &quot;max-bundle&quot; policy - video port =
is 0 because bundle-only, but both RTP and RTCP ports allocated for audio:<=
br>
m=3Daudio 10000<br>
a=3Drtcp:10001<br>
a=3Drtcp-mux<br>
m=3Dvideo 0<br>
a=3Dbundle-only<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">The new &ldquo;force-bundle&rdquo; policy - only a s=
ingle set of ports allocated - audio, video, RTP, RTCP share the same port.=
<br>
m=3Daudio 10000<br>
a=3Drtcp:10000<br>
a=3Drtcp-mux<br>
m=3Dvideo 10000<br>
a=3Drtcp:10000<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">During the discussion, the ability to avoid RTCP por=
t gathering was seen as valuable, but not the ability to avoid the fixup of=
fer. However, after thinking about it a bit, I don&rsquo;t
 think that there is a viable solution that avoids RTCP port gathering that=
 *doesn&rsquo;t* use a single set of ports. The only other option would be =
to do something like this:<u></u><u></u></p>
<p class=3D"MsoNormal">m=3Daudio 10000<br>
a=3Drtcp:10000 # same as m=3Daudio port<br>
a=3Drtcp-mux<br>
m=3Dvideo 0<br>
a=3Dbundle-only<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">but this will almost surely will result in a malfunc=
tion when talking to a non-rtcp-mux client, due to the same port being used=
 for RTP and RTCP, and still requires the fixup offer,
 and as such is net worse than the force-bundle approach indicated above.<u=
></u><u></u></p>
<p class=3D"MsoNormal">So I think we the force-bundle approach is the one w=
e want to minimize port gathering. The only question is whether we should k=
eep max-bundle as well. To be specific, we need to
 decide between:<u></u><u></u></p>
<p class=3D"MsoNormal">a) Keep max-bundle as it is (separate RTP and RTCP p=
orts allocated), and add a new force-bundle policy that creates an offer wi=
th a single port shared between all m-sections (RTP
 and RTCP). max-bundle remains nominally backwards compatible with non-BUND=
LE/non-rtcp-mux endpoints, in that it will still send a single audio stream=
. force-bundle is not backwards compatible with non-BUNDLE endpoints.<u></u=
><u></u></p>


<p class=3D"MsoNormal">b) Add force-bundle, but get rid of max-bundle.<u></=
u><u></u></p>
</div>
</div></div></div>
</div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Mon, 10 Mar 2014 21:47:28 -0700
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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Cullen said:

"I am opposed to mandating that Client-to-Mixer Audio Level RTP header is
REQUIRED to be encrypted. It means that we can not really build
conferencing system where the mixers do not have access to decrypt the
media."

[BA] I am also not enthused about mandating encryption of header
extensions. However, in the interest of fairness, it is worth pointing out
that header extensions are optional, so if a particular extension is
encrypted, the mixer can always ignore it (and in this case, thereby do
more work mixing muted audio streams).  So my take would be to say SHOULD
encrypt Client-to-MIxer Audio level, but leave this up to negotiation so
that if the mixer doesn't want it encrypted, it can turn that off.


On Wed, Mar 5, 2014 at 4:32 AM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> I am opposed to mandating that Client-to-Mixer Audio Level RTP header is
> REQUIRED to be encrypted. It means that we can not really build
> conferencing system where the mixers do not have access to decrypt the
> media.
>
> I would like to propose that the JS application can control  if the header
> is encrypted or not.
>
> I am aware of the paper that suggests these audio levels can reveal the
> contents of the conversation. There is some element of truth to this. There
> is also some element of truth to the point that if this was true, speech
> recognition system would work better than they do. Encrypting this or not
> is a risk that the application using the header extension needs to
> evaluate, and WebRTC system needs to provide the applications with a
> control surfaces that allows them to use this in the way the applications
> desires.
>
> I will note that an normal application that used this header could decide
> if it was encrypted or not, what is different in RTCWeb is that we are
> building a platform and my view is that this platform should allow the
> applications build on the platform to decide the tradeoffs of risk for
> encrypting this or not.
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Cullen said:=A0<div><br></div><div>&quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">I am opposed to mandating that C=
lient-to-Mixer Audio Level RTP header is REQUIRED to be encrypted. It means=
 that we can not really build conferencing system where the mixers do not h=
ave access to decrypt the media.&quot;</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div><div><span style=3D"font-family:arial,sans-serif;font-size:13px">[BA=
] I am also not enthused about mandating encryption of header extensions. H=
owever, in the interest of fairness, it is worth pointing out that header e=
xtensions are optional, so if a particular extension is encrypted, the mixe=
r can always ignore it (and in this case, thereby do more work mixing muted=
 audio streams). =A0So my take would be to say SHOULD encrypt Client-to-MIx=
er Audio level, but leave this up to negotiation so that if the mixer doesn=
&#39;t want it encrypted, it can turn that off.=A0</span></div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed,=
 Mar 5, 2014 at 4:32 AM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt;<a =
href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;=
</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
I am opposed to mandating that Client-to-Mixer Audio Level RTP header is RE=
QUIRED to be encrypted. It means that we can not really build conferencing =
system where the mixers do not have access to decrypt the media.<br>
<br>
I would like to propose that the JS application can control =A0if the heade=
r is encrypted or not.<br>
<br>
I am aware of the paper that suggests these audio levels can reveal the con=
tents of the conversation. There is some element of truth to this. There is=
 also some element of truth to the point that if this was true, speech reco=
gnition system would work better than they do. Encrypting this or not is a =
risk that the application using the header extension needs to evaluate, and=
 WebRTC system needs to provide the applications with a control surfaces th=
at allows them to use this in the way the applications desires.<br>


<br>
I will note that an normal application that used this header could decide i=
f it was encrypted or not, what is different in RTCWeb is that we are build=
ing a platform and my view is that this platform should allow the applicati=
ons build on the platform to decide the tradeoffs of risk for encrypting th=
is or not.<br>


<br>
<br>
<br>
<br>
<br>
<br>
_______________________________________________<br>
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<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

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From: "Stach, Thomas" <thomas.stach@unify.com>
To: Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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From nobody Tue Mar 11 00:51:16 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Date: Tue, 11 Mar 2014 07:51:01 +0000
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

"BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp attr=
ibutes. I remember we had discussions about that, but I don=92t remember th=
e justification at the moment."

[BA] The justification was that an implementation interested in multiplexin=
g A/V would almost certainly also want to multiplex RTP and RTCP, because t=
he motivation for both is minimizing port usage.

[Christer] So, again, the idea is to use a=3Drtcp to indicate the *same* po=
rt as a=3Drtcp-mux indicates, and hope that the receiver supports at least =
either a=3Drtcp or a=3Drtcp-mux?

Regards,

Christer


On Mon, Mar 10, 2014 at 1:24 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

Before I comment on your proposal, let=92s take a few steps back (maybe it =
can even solve your issue).

BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp attri=
butes. I remember we had discussions about that, but I don=92t remember the=
 justification at the moment. But, I wonder whether that text is from befor=
e we made a decision that, unless both endpoints support BUNDLE, no endpoin=
t will use a single address:port.

So, the question is: do we really need the SDP rtcp attribute?

Regards,

Christer





From: Justin Uberti [mailto:juberti@google.com<mailto:juberti@google.com>]
Sent: 10. maaliskuuta 2014 8:18
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; Eric Rescorla; Christer Holmbe=
rg
Subject: BUNDLE with implicit rtcp-mux

In my JSEP preso during the RTCWEB session last Wednesday, I discussed addi=
ng a new BUNDLE policy to JSEP. This policy, which would be called somethin=
g like =93force-bundle=94, would force the use of the same ports for BUNDLE=
d m-sections, as well as RTCP mux. When one knows a priori that the other s=
ide supports BUNDLE, this would reduce candidate gathering even over the cu=
rrently defined =93max-bundle=94 policy, since there would be no need to ga=
ther RTCP ports at all, and also avoid the need for the BUNDLE fixup offer =
(where the ports are updated to be the same for all BUNDLEd m-sections).
To explain the differences here, here are some example offers for an audio+=
video call:
Default bundle policy - separate ports for audio, video, RTP and RTCP:
m=3Daudio 10000
a=3Drtcp:10001
a=3Drtcp-mux
m=3Dvideo 11000
a=3Drtcp:11001
a=3Drtcp-mux
Existing "max-bundle" policy - video port is 0 because bundle-only, but bot=
h RTP and RTCP ports allocated for audio:
m=3Daudio 10000
a=3Drtcp:10001
a=3Drtcp-mux
m=3Dvideo 0
a=3Dbundle-only
a=3Drtcp-mux
The new =93force-bundle=94 policy - only a single set of ports allocated - =
audio, video, RTP, RTCP share the same port.
m=3Daudio 10000
a=3Drtcp:10000
a=3Drtcp-mux
m=3Dvideo 10000
a=3Drtcp:10000
a=3Drtcp-mux
During the discussion, the ability to avoid RTCP port gathering was seen as=
 valuable, but not the ability to avoid the fixup offer. However, after thi=
nking about it a bit, I don=92t think that there is a viable solution that =
avoids RTCP port gathering that *doesn=92t* use a single set of ports. The =
only other option would be to do something like this:
m=3Daudio 10000
a=3Drtcp:10000 # same as m=3Daudio port
a=3Drtcp-mux
m=3Dvideo 0
a=3Dbundle-only
a=3Drtcp-mux
but this will almost surely will result in a malfunction when talking to a =
non-rtcp-mux client, due to the same port being used for RTP and RTCP, and =
still requires the fixup offer, and as such is net worse than the force-bun=
dle approach indicated above.
So I think we the force-bundle approach is the one we want to minimize port=
 gathering. The only question is whether we should keep max-bundle as well.=
 To be specific, we need to decide between:
a) Keep max-bundle as it is (separate RTP and RTCP ports allocated), and ad=
d a new force-bundle policy that creates an offer with a single port shared=
 between all m-sections (RTP and RTCP). max-bundle remains nominally backwa=
rds compatible with non-BUNDLE/non-rtcp-mux endpoints, in that it will stil=
l send a single audio stream. force-bundle is not backwards compatible with=
 non-BUNDLE endpoints.
b) Add force-bundle, but get rid of max-bundle.

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb



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<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">Hi,<br>
<br>
<div style=3D"font-family: Times New Roman; color: #000000; font-size: 16px=
">
<div>
<div dir=3D"ltr">
<div>&quot;<span style=3D"font-size:11pt; font-family:Calibri,sans-serif; c=
olor:rgb(31,73,125)">BUNDLE currently mandates the usage of both the SDP rt=
cp-mux and rtcp attributes. I remember we had discussions about that, but I=
 don=92t remember the justification at the
 moment.</span>&quot;</div>
<div><br>
</div>
<div>[BA] The justification was that an implementation interested in multip=
lexing A/V would almost certainly also want to multiplex RTP and RTCP, beca=
use the motivation for both is minimizing port usage.&nbsp;</div>
</div>
<div class=3D"gmail_extra"><br>
[Christer] So, again, the idea is to use a=3Drtcp to indicate the *same* po=
rt as a=3Drtcp-mux indicates, and hope that the receiver supports at least =
either a=3Drtcp or a=3Drtcp-mux?<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
<div class=3D"gmail_quote">On Mon, Mar 10, 2014 at 1:24 AM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div lang=3D"EN-US">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d">Hi,<u></u><u></u></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d">Before I comment on you=
r proposal, let=92s take a few steps back (maybe it can even solve your iss=
ue).<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d">BUNDLE currently mandat=
es the usage of both the SDP rtcp-mux and rtcp attributes. I remember we ha=
d discussions about that, but I don=92t remember the justification
 at the moment. But, I wonder whether that text is from before we made a de=
cision that, unless both endpoints support BUNDLE, no endpoint will use a s=
ingle address:port.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d">So, the question is: do=
 we really need the SDP rtcp attribute?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d">Regards,<u></u><u></u><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d">Christer<u></u><u></u><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1f497d"><u></u>&nbsp;<u></u></s=
pan></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt; font-family:&quo=
t;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-=
size:10.0pt; font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Justin=
 Uberti [mailto:<a href=3D"mailto:juberti@google.com" target=3D"_blank">jub=
erti@google.com</a>]
<br>
<b>Sent:</b> 10. maaliskuuta 2014 8:18<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a>; Eric Rescorla; Christer Holmberg<br>
<b>Subject:</b> BUNDLE with implicit rtcp-mux<u></u><u></u></span></p>
<div>
<div class=3D"h5">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<div>
<p class=3D"MsoNormal">In my JSEP preso during the RTCWEB session last Wedn=
esday, I discussed adding a new BUNDLE policy to JSEP. This policy, which w=
ould be called something like =93force-bundle=94, would force the use of th=
e same ports for BUNDLEd m-sections, as
 well as RTCP mux. When one knows a priori that the other side supports BUN=
DLE, this would reduce candidate gathering even over the currently defined =
=93max-bundle=94 policy, since there would be no need to gather RTCP ports =
at all, and also avoid the need for
 the BUNDLE fixup offer (where the ports are updated to be the same for all=
 BUNDLEd m-sections).<u></u><u></u></p>
<p class=3D"MsoNormal">To explain the differences here, here are some examp=
le offers for an audio&#43;video call:<u></u><u></u></p>
<p class=3D"MsoNormal">Default bundle policy - separate ports for audio, vi=
deo, RTP and RTCP:<br>
m=3Daudio 10000<br>
a=3Drtcp:10001<br>
a=3Drtcp-mux<br>
m=3Dvideo 11000<br>
a=3Drtcp:11001<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">Existing &quot;max-bundle&quot; policy - video port =
is 0 because bundle-only, but both RTP and RTCP ports allocated for audio:<=
br>
m=3Daudio 10000<br>
a=3Drtcp:10001<br>
a=3Drtcp-mux<br>
m=3Dvideo 0<br>
a=3Dbundle-only<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">The new =93force-bundle=94 policy - only a single se=
t of ports allocated - audio, video, RTP, RTCP share the same port.<br>
m=3Daudio 10000<br>
a=3Drtcp:10000<br>
a=3Drtcp-mux<br>
m=3Dvideo 10000<br>
a=3Drtcp:10000<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">During the discussion, the ability to avoid RTCP por=
t gathering was seen as valuable, but not the ability to avoid the fixup of=
fer. However, after thinking about it a bit, I don=92t think that there is =
a viable solution that avoids RTCP port
 gathering that *doesn=92t* use a single set of ports. The only other optio=
n would be to do something like this:<u></u><u></u></p>
<p class=3D"MsoNormal">m=3Daudio 10000<br>
a=3Drtcp:10000 # same as m=3Daudio port<br>
a=3Drtcp-mux<br>
m=3Dvideo 0<br>
a=3Dbundle-only<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">but this will almost surely will result in a malfunc=
tion when talking to a non-rtcp-mux client, due to the same port being used=
 for RTP and RTCP, and still requires the fixup offer, and as such is net w=
orse than the force-bundle approach
 indicated above.<u></u><u></u></p>
<p class=3D"MsoNormal">So I think we the force-bundle approach is the one w=
e want to minimize port gathering. The only question is whether we should k=
eep max-bundle as well. To be specific, we need to decide between:<u></u><u=
></u></p>
<p class=3D"MsoNormal">a) Keep max-bundle as it is (separate RTP and RTCP p=
orts allocated), and add a new force-bundle policy that creates an offer wi=
th a single port shared between all m-sections (RTP and RTCP). max-bundle r=
emains nominally backwards compatible
 with non-BUNDLE/non-rtcp-mux endpoints, in that it will still send a singl=
e audio stream. force-bundle is not backwards compatible with non-BUNDLE en=
dpoints.<u></u><u></u></p>
<p class=3D"MsoNormal">b) Add force-bundle, but get rid of max-bundle.<u></=
u><u></u></p>
</div>
</div>
</div>
</div>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</body>
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From nobody Tue Mar 11 07:02:30 2014
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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Le 2014-03-10 16:57, James Polk a écrit :
> At 12:16 AM 3/10/2014, Justin Uberti wrote:
>> I think that the existing guidance should be sufficient in
>> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think the
>> same DSCP markings as any other ICE connectivity checks should be
>> used; for ICE-for-SCTP checks, the same DSCP markings as media packets
>> (i.e. the SCTP traffic) should be used.
>>
>> If multiplexing is being performed, the connectivity checks probably
>> should use the highest DSCP value being used by the multiplexed media.
> 
> why is (seemingly) everyone's default position "use the highest DSCP
> available" for signaling packets?
> 
> You just need to make sure the packets aren't starved or dropped by/at
> congestion points.

The underlying principle is that connectivity checks need to *check
connectivity* (duh!). That's why you use the same DSCP as the media.
Connectivity is affected by the DSCP. For example some DSCPs could be
filtered, or could be placed in a low-priority queue and get dropped
such that connectivity fails.

>From this principle follows this conclusion: on a given 5-tuple, if your
media uses different DSCPs, you need to check connectivity for all those
DSCPs. That is, you would send multiple connectivity checks, one for
each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we might
need an alternative heuristic: try the lowest-priority DSCP, and assume
that if that one works, the higher-priority ones should work as well.
(Note that Justin suggested the contrary.)

What you want to avoid is STUN packets getting through but not the
corresponding media. That is an unworkable situation. The reverse is not
ideal either (you think you have no connectivity when in fact you do),
but at least ICE can work around it (by picking a different candidate).

Simon
-- 
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Magnus said:

"I would guess that the simplest is to remove discussion of TURN TCP
altogether from Transports. That would not recommend it nor make it
disallowed. If we want to be explicit, then simply motivate why we don't
believe it necessary."

[BA] Removing a recommendation from the Transport document is OK with  me.
However, since we've had quite a bit of discussion on this topic, it would
be good to state why we don't believe it to be necessary (so as to avoid
rehashing this down the line).


On Mon, Mar 10, 2014 at 8:33 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-06 14:34, Simon Perreault wrote:
> > Le 2014-03-06 13:23, Justin Uberti a =E9crit :
> >> can we agree that TURN TCP candidates are a SHOULD NOT?
> >
> > Not a SHOULD, sure. SHOULD NOT, no.
> >
>
> I would guess that the simplest is to remove discussion of TURN TCP
> altogether from Transports. That would not recommend it nor make it
> disallowed. If we want to be explicit, then simply motivate why we don't
> believe it necessary.
>
> Justin, what is the reason you wanted SHOULD NOT?
>
> cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
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<div dir=3D"ltr"><div>Magnus said: </div><div><br></div><div>&quot;I would =
guess that the simplest is to remove discussion of TURN TCP<br> altogether =
from Transports. That would not recommend it nor make it<br> disallowed. If=
 we want to be explicit, then simply motivate why we don&#39;t<br>

 believe it necessary.&quot;</div><div><br></div><div>[BA] Removing a recom=
mendation from the Transport document=A0is OK with=A0 me.=A0 However, since=
 we&#39;ve had quite a bit of discussion on this topic, it would be good to=
 state why we don&#39;t believe it to be necessary (so as to avoid rehashin=
g this down the line). </div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon,=
 Mar 10, 2014 at 8:33 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=
=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerl=
und@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">On 2014-03-06 14:34, Simon Perreault wrote:<=
br>
&gt; Le 2014-03-06 13:23, Justin Uberti a =E9crit :<br>
&gt;&gt; can we agree that TURN TCP candidates are a SHOULD NOT?<br>
&gt;<br>
&gt; Not a SHOULD, sure. SHOULD NOT, no.<br>
&gt;<br>
<br>
I would guess that the simplest is to remove discussion of TURN TCP<br>
altogether from Transports. That would not recommend it nor make it<br>
disallowed. If we want to be explicit, then simply motivate why we don&#39;=
t<br>
believe it necessary.<br>
<br>
Justin, what is the reason you wanted SHOULD NOT?<br>
<br>
cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Phone =A0<a href=3D"tel:%2B46=
%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | Mobile <a href=3D"tel:%2B=
46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On 03/11/2014 03:02 PM, Simon Perreault wrote:
> Le 2014-03-10 16:57, James Polk a écrit :
>> At 12:16 AM 3/10/2014, Justin Uberti wrote:
>>> I think that the existing guidance should be sufficient in
>>> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think the
>>> same DSCP markings as any other ICE connectivity checks should be
>>> used; for ICE-for-SCTP checks, the same DSCP markings as media packets
>>> (i.e. the SCTP traffic) should be used.
>>>
>>> If multiplexing is being performed, the connectivity checks probably
>>> should use the highest DSCP value being used by the multiplexed media.
>> why is (seemingly) everyone's default position "use the highest DSCP
>> available" for signaling packets?
>>
>> You just need to make sure the packets aren't starved or dropped by/at
>> congestion points.
> The underlying principle is that connectivity checks need to *check
> connectivity* (duh!). That's why you use the same DSCP as the media.
> Connectivity is affected by the DSCP. For example some DSCPs could be
> filtered, or could be placed in a low-priority queue and get dropped
> such that connectivity fails.
>
> >From this principle follows this conclusion: on a given 5-tuple, if your
> media uses different DSCPs, you need to check connectivity for all those
> DSCPs. That is, you would send multiple connectivity checks, one for
> each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we might
> need an alternative heuristic: try the lowest-priority DSCP, and assume
> that if that one works, the higher-priority ones should work as well.
> (Note that Justin suggested the contrary.)
>
> What you want to avoid is STUN packets getting through but not the
> corresponding media. That is an unworkable situation. The reverse is not
> ideal either (you think you have no connectivity when in fact you do),
> but at least ICE can work around it (by picking a different candidate).

This explanation makes sense for (ICE) connectivity checks.

For other control information, especially information about delay, you 
want to have it delivered as fast as possible, and with as little loss 
as possible, because jitter in the return path will make the delay 
signal more noisy, and lost packets may cause drastic actions like those 
the circuit-breakers draft recommends.

Whether that always translates to "the highest DSCP code point" is .... 
a good question.




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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On Mar 11, 2014 7:44 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote:
>
> On 03/11/2014 03:02 PM, Simon Perreault wrote:
>>
>> Le 2014-03-10 16:57, James Polk a =E9crit :
>>>
>>> At 12:16 AM 3/10/2014, Justin Uberti wrote:
>>>>
>>>> I think that the existing guidance should be sufficient in
>>>> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think the
>>>> same DSCP markings as any other ICE connectivity checks should be
>>>> used; for ICE-for-SCTP checks, the same DSCP markings as media packets
>>>> (i.e. the SCTP traffic) should be used.
>>>>
>>>> If multiplexing is being performed, the connectivity checks probably
>>>> should use the highest DSCP value being used by the multiplexed media.
>>>
>>> why is (seemingly) everyone's default position "use the highest DSCP
>>> available" for signaling packets?
>>>
>>> You just need to make sure the packets aren't starved or dropped by/at
>>> congestion points.
>>
>> The underlying principle is that connectivity checks need to *check
>> connectivity* (duh!). That's why you use the same DSCP as the media.
>> Connectivity is affected by the DSCP. For example some DSCPs could be
>> filtered, or could be placed in a low-priority queue and get dropped
>> such that connectivity fails.
>>
>> >From this principle follows this conclusion: on a given 5-tuple, if you=
r
>> media uses different DSCPs, you need to check connectivity for all those
>> DSCPs. That is, you would send multiple connectivity checks, one for
>> each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we might
>> need an alternative heuristic: try the lowest-priority DSCP, and assume
>> that if that one works, the higher-priority ones should work as well.
>> (Note that Justin suggested the contrary.)
>>
>> What you want to avoid is STUN packets getting through but not the
>> corresponding media. That is an unworkable situation. The reverse is not
>> ideal either (you think you have no connectivity when in fact you do),
>> but at least ICE can work around it (by picking a different candidate).
>
>
> This explanation makes sense for (ICE) connectivity checks.
>
> For other control information, especially information about delay, you
want to have it delivered as fast as possible, and with as little loss as
possible, because jitter in the return path will make the delay signal more
noisy, and lost packets may cause drastic actions like those the
circuit-breakers draft recommends.
>
> Whether that always translates to "the highest DSCP code point" is .... a
good question.

It doesn't.  Keith added af42 support to mosh last year and had several
reports of packets being dropped with that marking. Worse the drops
happened after 10 seconds of connectivity.

Ecn markings on the other hand have thus far survived (if occasionally
stomped on) across the open internet in that protocol.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<p dir=3D"ltr"><br>
On Mar 11, 2014 7:44 AM, &quot;Harald Alvestrand&quot; &lt;<a href=3D"mailt=
o:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
&gt;<br>
&gt; On 03/11/2014 03:02 PM, Simon Perreault wrote:<br>
&gt;&gt;<br>
&gt;&gt; Le 2014-03-10 16:57, James Polk a =E9crit :<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; At 12:16 AM 3/10/2014, Justin Uberti wrote:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I think that the existing guidance should be sufficient in=
<br>
&gt;&gt;&gt;&gt; non-multiplexing cases (i.e. BUNDLE). For consent checks, =
I think the<br>
&gt;&gt;&gt;&gt; same DSCP markings as any other ICE connectivity checks sh=
ould be<br>
&gt;&gt;&gt;&gt; used; for ICE-for-SCTP checks, the same DSCP markings as m=
edia packets<br>
&gt;&gt;&gt;&gt; (i.e. the SCTP traffic) should be used.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; If multiplexing is being performed, the connectivity check=
s probably<br>
&gt;&gt;&gt;&gt; should use the highest DSCP value being used by the multip=
lexed media.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; why is (seemingly) everyone&#39;s default position &quot;use t=
he highest DSCP<br>
&gt;&gt;&gt; available&quot; for signaling packets?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; You just need to make sure the packets aren&#39;t starved or d=
ropped by/at<br>
&gt;&gt;&gt; congestion points.<br>
&gt;&gt;<br>
&gt;&gt; The underlying principle is that connectivity checks need to *chec=
k<br>
&gt;&gt; connectivity* (duh!). That&#39;s why you use the same DSCP as the =
media.<br>
&gt;&gt; Connectivity is affected by the DSCP. For example some DSCPs could=
 be<br>
&gt;&gt; filtered, or could be placed in a low-priority queue and get dropp=
ed<br>
&gt;&gt; such that connectivity fails.<br>
&gt;&gt;<br>
&gt;&gt; &gt;From this principle follows this conclusion: on a given 5-tupl=
e, if your<br>
&gt;&gt; media uses different DSCPs, you need to check connectivity for all=
 those<br>
&gt;&gt; DSCPs. That is, you would send multiple connectivity checks, one f=
or<br>
&gt;&gt; each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we =
might<br>
&gt;&gt; need an alternative heuristic: try the lowest-priority DSCP, and a=
ssume<br>
&gt;&gt; that if that one works, the higher-priority ones should work as we=
ll.<br>
&gt;&gt; (Note that Justin suggested the contrary.)<br>
&gt;&gt;<br>
&gt;&gt; What you want to avoid is STUN packets getting through but not the=
<br>
&gt;&gt; corresponding media. That is an unworkable situation. The reverse =
is not<br>
&gt;&gt; ideal either (you think you have no connectivity when in fact you =
do),<br>
&gt;&gt; but at least ICE can work around it (by picking a different candid=
ate).<br>
&gt;<br>
&gt;<br>
&gt; This explanation makes sense for (ICE) connectivity checks.<br>
&gt;<br>
&gt; For other control information, especially information about delay, you=
 want to have it delivered as fast as possible, and with as little loss as =
possible, because jitter in the return path will make the delay signal more=
 noisy, and lost packets may cause drastic actions like those the circuit-b=
reakers draft recommends.<br>

&gt;<br>
&gt; Whether that always translates to &quot;the highest DSCP code point&qu=
ot; is .... a good question.</p>
<p dir=3D"ltr">It doesn&#39;t.=A0 Keith added af42 support to mosh last yea=
r and had several reports of packets being dropped with that marking. Worse=
 the drops happened after 10 seconds of connectivity.</p>
<p dir=3D"ltr">Ecn markings on the other hand have thus far survived (if oc=
casionally stomped on) across the open internet in that protocol.<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

--001a11c26bb86d04a004f457e631--


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Subject: Re: [rtcweb] Transports: RFC 6062 support
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On 03/11/2014 03:41 PM, Bernard Aboba wrote:
> Magnus said:
>
> "I would guess that the simplest is to remove discussion of TURN TCP
> altogether from Transports. That would not recommend it nor make it
> disallowed. If we want to be explicit, then simply motivate why we don't
> believe it necessary."
>
> [BA] Removing a recommendation from the Transport document is OK with  
> me.  However, since we've had quite a bit of discussion on this topic, 
> it would be good to state why we don't believe it to be necessary (so 
> as to avoid rehashing this down the line).

In RFC 2119 theory, MAY and not mentioning it at all have exactly the 
same weight.
MAY gives you a place to hang the discussion of why it's not anything 
stronger on.


>
>
> On Mon, Mar 10, 2014 at 8:33 AM, Magnus Westerlund 
> <magnus.westerlund@ericsson.com 
> <mailto:magnus.westerlund@ericsson.com>> wrote:
>
>     On 2014-03-06 14:34, Simon Perreault wrote:
>     > Le 2014-03-06 13:23, Justin Uberti a écrit :
>     >> can we agree that TURN TCP candidates are a SHOULD NOT?
>     >
>     > Not a SHOULD, sure. SHOULD NOT, no.
>     >
>
>     I would guess that the simplest is to remove discussion of TURN TCP
>     altogether from Transports. That would not recommend it nor make it
>     disallowed. If we want to be explicit, then simply motivate why we
>     don't
>     believe it necessary.
>
>     Justin, what is the reason you wanted SHOULD NOT?
>
>     cheers
>
>     Magnus Westerlund
>
>     ----------------------------------------------------------------------
>     Services, Media and Network features, Ericsson Research EAB/TXM
>     ----------------------------------------------------------------------
>     Ericsson AB                 | Phone +46 10 7148287
>     <tel:%2B46%2010%207148287>
>     Färögatan 6                 | Mobile +46 73 0949079
>     <tel:%2B46%2073%200949079>
>     SE-164 80 Stockholm, Sweden | mailto:
>     magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>
>     ----------------------------------------------------------------------
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html>
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    <div class="moz-cite-prefix">On 03/11/2014 03:41 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOW+2dtmLUWO-fKtNMMxn4ozywY2-dM3X0kEYwipRb9LcHgxQw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>Magnus said: </div>
        <div><br>
        </div>
        <div>"I would guess that the simplest is to remove discussion of
          TURN TCP<br>
          altogether from Transports. That would not recommend it nor
          make it<br>
          disallowed. If we want to be explicit, then simply motivate
          why we don't<br>
          believe it necessary."</div>
        <div><br>
        </div>
        <div>[BA] Removing a recommendation from the Transport
          document&nbsp;is OK with&nbsp; me.&nbsp; However, since we've had quite a bit
          of discussion on this topic, it would be good to state why we
          don't believe it to be necessary (so as to avoid rehashing
          this down the line). </div>
      </div>
    </blockquote>
    <br>
    In RFC 2119 theory, MAY and not mentioning it at all have exactly
    the same weight.<br>
    MAY gives you a place to hang the discussion of why it's not
    anything stronger on.<br>
    <br>
    <br>
    <blockquote
cite="mid:CAOW+2dtmLUWO-fKtNMMxn4ozywY2-dM3X0kEYwipRb9LcHgxQw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Mon, Mar 10, 2014 at 8:33 AM, Magnus
          Westerlund <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:magnus.westerlund@ericsson.com"
              target="_blank">magnus.westerlund@ericsson.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">On
            2014-03-06 14:34, Simon Perreault wrote:<br>
            &gt; Le 2014-03-06 13:23, Justin Uberti a &eacute;crit :<br>
            &gt;&gt; can we agree that TURN TCP candidates are a SHOULD
            NOT?<br>
            &gt;<br>
            &gt; Not a SHOULD, sure. SHOULD NOT, no.<br>
            &gt;<br>
            <br>
            I would guess that the simplest is to remove discussion of
            TURN TCP<br>
            altogether from Transports. That would not recommend it nor
            make it<br>
            disallowed. If we want to be explicit, then simply motivate
            why we don't<br>
            believe it necessary.<br>
            <br>
            Justin, what is the reason you wanted SHOULD NOT?<br>
            <br>
            cheers<br>
            <br>
            Magnus Westerlund<br>
            <br>
----------------------------------------------------------------------<br>
            Services, Media and Network features, Ericsson Research
            EAB/TXM<br>
----------------------------------------------------------------------<br>
            Ericsson AB &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | Phone &nbsp;<a
              moz-do-not-send="true" href="tel:%2B46%2010%207148287"
              value="+46107148287">+46 10 7148287</a><br>
            F&auml;r&ouml;gatan 6 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | Mobile <a
              moz-do-not-send="true" href="tel:%2B46%2073%200949079"
              value="+46730949079">+46 73 0949079</a><br>
            SE-164 80 Stockholm, Sweden | mailto: <a
              moz-do-not-send="true"
              href="mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb"
              target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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* Harald Alvestrand wrote:
>In RFC 2119 theory, MAY and not mentioning it at all have exactly the 
>same weight.

(That depends on whether the default is "allowed" or "prohibited" and
there are plenty of specifications where explicit MAY overrides a de-
fault of "prohibited".)
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 


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On 03/11/2014 09:26 PM, Bjoern Hoehrmann wrote:
> * Harald Alvestrand wrote:
>> In RFC 2119 theory, MAY and not mentioning it at all have exactly the 
>> same weight.
> (That depends on whether the default is "allowed" or "prohibited" and
> there are plenty of specifications where explicit MAY overrides a de-
> fault of "prohibited".)
Probably true. But I can't think of any at the moment - do you have
something specific in mind?

Anyway, in this case, I don't think we have any default.

My expectation of "normal" is that any implementation is free to add any
extension (past, present or future), as long as it's possible to
practice it; it is very rare that we do something explicit like the
RTCWEB SDES "MUST NOT".

-- 
Surveillance is pervasive. Go Dark.


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From: Bjoern Hoehrmann <derhoermi@gmx.net>
To: Harald Alvestrand <harald@alvestrand.no>
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* Harald Alvestrand wrote:
>On 03/11/2014 09:26 PM, Bjoern Hoehrmann wrote:
>> * Harald Alvestrand wrote:
>>> In RFC 2119 theory, MAY and not mentioning it at all have exactly the 
>>> same weight.
>> (That depends on whether the default is "allowed" or "prohibited" and
>> there are plenty of specifications where explicit MAY overrides a de-
>> fault of "prohibited".)
>
>Probably true. But I can't think of any at the moment - do you have
>something specific in mind?

(A typical example is when RFC 2119 keywords are used to describe how a
protocol element is constructed, say, "<foo> elements MAY carry bar=''
attributes"; you wouldn't assume you can use <foo bar=''> without that.)
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 


From nobody Tue Mar 11 14:39:28 2014
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On 03/11/2014 10:23 PM, Bjoern Hoehrmann wrote:
> * Harald Alvestrand wrote:
>> On 03/11/2014 09:26 PM, Bjoern Hoehrmann wrote:
>>> * Harald Alvestrand wrote:
>>>> In RFC 2119 theory, MAY and not mentioning it at all have exactly the 
>>>> same weight.
>>> (That depends on whether the default is "allowed" or "prohibited" and
>>> there are plenty of specifications where explicit MAY overrides a de-
>>> fault of "prohibited".)
>> Probably true. But I can't think of any at the moment - do you have
>> something specific in mind?
> (A typical example is when RFC 2119 keywords are used to describe how a
> protocol element is constructed, say, "<foo> elements MAY carry bar=''
> attributes"; you wouldn't assume you can use <foo bar=''> without that.)
Any specific examples?

Without context, this seems a strange way to specify protocol elements;
more common is to use ABNF, and not 2119 keywords.

And ABNF even allows "free" extensibility by having other specs use the
/= operator....


-- 
Surveillance is pervasive. Go Dark.


From nobody Tue Mar 11 14:45:07 2014
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* Harald Alvestrand wrote:
>On 03/11/2014 10:23 PM, Bjoern Hoehrmann wrote:
>> (A typical example is when RFC 2119 keywords are used to describe how a
>> protocol element is constructed, say, "<foo> elements MAY carry bar=''
>> attributes"; you wouldn't assume you can use <foo bar=''> without that.)
>
>Any specific examples?

I think this is really very common, https://tools.ietf.org/html/rfc4287
is an instance of my example above.
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 


From nobody Tue Mar 11 17:59:19 2014
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Subject: Re: [rtcweb] Transports: RFC 6062 support
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The reason I suggested SHOULD NOT is because I think this technique is
unnecessary for WebRTC, and typically net harmful for the reasons that
Matthew mentions.

That said, I would be fine with saying MAY and including some explanatory
text; my main concerns was to avoid the ambiguity that would arise from
leaving this out altogether.


On Mon, Mar 10, 2014 at 8:33 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-06 14:34, Simon Perreault wrote:
> > Le 2014-03-06 13:23, Justin Uberti a =C3=A9crit :
> >> can we agree that TURN TCP candidates are a SHOULD NOT?
> >
> > Not a SHOULD, sure. SHOULD NOT, no.
> >
>
> I would guess that the simplest is to remove discussion of TURN TCP
> altogether from Transports. That would not recommend it nor make it
> disallowed. If we want to be explicit, then simply motivate why we don't
> believe it necessary.
>
> Justin, what is the reason you wanted SHOULD NOT?
>
> cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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<div dir=3D"ltr">The reason I suggested SHOULD NOT is because I think this =
technique is unnecessary for WebRTC, and typically net harmful for the reas=
ons that Matthew mentions.=C2=A0<div><br></div><div>That said, I would be f=
ine with saying MAY and including some explanatory text; my main concerns w=
as to avoid the ambiguity that would arise from leaving this out altogether=
.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon,=
 Mar 10, 2014 at 8:33 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=
=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerl=
und@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 2014-03-06 14:34, Simon P=
erreault wrote:<br>
&gt; Le 2014-03-06 13:23, Justin Uberti a =C3=A9crit :<br>
&gt;&gt; can we agree that TURN TCP candidates are a SHOULD NOT?<br>
&gt;<br>
&gt; Not a SHOULD, sure. SHOULD NOT, no.<br>
&gt;<br>
<br>
</div>I would guess that the simplest is to remove discussion of TURN TCP<b=
r>
altogether from Transports. That would not recommend it nor make it<br>
disallowed. If we want to be explicit, then simply motivate why we don&#39;=
t<br>
believe it necessary.<br>
<br>
Justin, what is the reason you wanted SHOULD NOT?<br>
<br>
cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7=
148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+4=
6 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</blockquote></div><br></div>

--001a1135f378ff9d1e04f45e57e2--


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From: Justin Uberti <juberti@google.com>
Date: Tue, 11 Mar 2014 20:53:11 -0700
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On Tue, Mar 11, 2014 at 7:02 AM, Simon Perreault <
simon.perreault@viagenie.ca> wrote:

> Le 2014-03-10 16:57, James Polk a =C3=A9crit :
> > At 12:16 AM 3/10/2014, Justin Uberti wrote:
> >> I think that the existing guidance should be sufficient in
> >> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think the
> >> same DSCP markings as any other ICE connectivity checks should be
> >> used; for ICE-for-SCTP checks, the same DSCP markings as media packets
> >> (i.e. the SCTP traffic) should be used.
> >>
> >> If multiplexing is being performed, the connectivity checks probably
> >> should use the highest DSCP value being used by the multiplexed media.
> >
> > why is (seemingly) everyone's default position "use the highest DSCP
> > available" for signaling packets?
> >
> > You just need to make sure the packets aren't starved or dropped by/at
> > congestion points.
>
> The underlying principle is that connectivity checks need to *check
> connectivity* (duh!). That's why you use the same DSCP as the media.
> Connectivity is affected by the DSCP. For example some DSCPs could be
> filtered, or could be placed in a low-priority queue and get dropped
> such that connectivity fails.
>
> >From this principle follows this conclusion: on a given 5-tuple, if your
> media uses different DSCPs, you need to check connectivity for all those
> DSCPs. That is, you would send multiple connectivity checks, one for
> each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we might
> need an alternative heuristic: try the lowest-priority DSCP, and assume
> that if that one works, the higher-priority ones should work as well.
> (Note that Justin suggested the contrary.)
>
> What you want to avoid is STUN packets getting through but not the
> corresponding media. That is an unworkable situation. The reverse is not
> ideal either (you think you have no connectivity when in fact you do),
> but at least ICE can work around it (by picking a different candidate).
>

We also need to consider the scenario of STUN consent checks, which will
need to get through reliably even when media marked with DSCP is already
flowing. In this case I think that using the highest DSCP value for STUN is
unavoidable.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Mar 11, 2014 at 7:02 AM, Simon Perreault <span dir=3D"ltr">=
&lt;<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D"_blank">simon.=
perreault@viagenie.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2014-03-10 16:57, James Polk a =C3=A9crit=
 :<br>
<div class=3D"">&gt; At 12:16 AM 3/10/2014, Justin Uberti wrote:<br>
&gt;&gt; I think that the existing guidance should be sufficient in<br>
&gt;&gt; non-multiplexing cases (i.e. BUNDLE). For consent checks, I think =
the<br>
&gt;&gt; same DSCP markings as any other ICE connectivity checks should be<=
br>
&gt;&gt; used; for ICE-for-SCTP checks, the same DSCP markings as media pac=
kets<br>
&gt;&gt; (i.e. the SCTP traffic) should be used.<br>
&gt;&gt;<br>
&gt;&gt; If multiplexing is being performed, the connectivity checks probab=
ly<br>
&gt;&gt; should use the highest DSCP value being used by the multiplexed me=
dia.<br>
&gt;<br>
&gt; why is (seemingly) everyone&#39;s default position &quot;use the highe=
st DSCP<br>
&gt; available&quot; for signaling packets?<br>
&gt;<br>
&gt; You just need to make sure the packets aren&#39;t starved or dropped b=
y/at<br>
&gt; congestion points.<br>
<br>
</div>The underlying principle is that connectivity checks need to *check<b=
r>
connectivity* (duh!). That&#39;s why you use the same DSCP as the media.<br=
>
Connectivity is affected by the DSCP. For example some DSCPs could be<br>
filtered, or could be placed in a low-priority queue and get dropped<br>
such that connectivity fails.<br>
<br>
&gt;From this principle follows this conclusion: on a given 5-tuple, if you=
r<br>
media uses different DSCPs, you need to check connectivity for all those<br=
>
DSCPs. That is, you would send multiple connectivity checks, one for<br>
each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we might<br>
need an alternative heuristic: try the lowest-priority DSCP, and assume<br>
that if that one works, the higher-priority ones should work as well.<br>
(Note that Justin suggested the contrary.)<br>
<br>
What you want to avoid is STUN packets getting through but not the<br>
corresponding media. That is an unworkable situation. The reverse is not<br=
>
ideal either (you think you have no connectivity when in fact you do),<br>
but at least ICE can work around it (by picking a different candidate).<br>=
</blockquote><div><br></div><div>We also need to consider the scenario of S=
TUN consent checks, which will need to get through reliably even when media=
 marked with DSCP is already flowing. In this case I think that using the h=
ighest DSCP value for STUN is unavoidable.</div>

</div></div></div>

--20cf307f313cb029e004f460c7fd--


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From: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On Tue, Mar 11, 2014 at 10:17 AM, Dave Taht <dave.taht@gmail.com> wrote:

>
> On Mar 11, 2014 7:44 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote=
:
> >
> > On 03/11/2014 03:02 PM, Simon Perreault wrote:
> >>
> >> Le 2014-03-10 16:57, James Polk a =C3=A9crit :
> >>>
> >>> At 12:16 AM 3/10/2014, Justin Uberti wrote:
> >>>>
> >>>> I think that the existing guidance should be sufficient in
> >>>> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think th=
e
> >>>> same DSCP markings as any other ICE connectivity checks should be
> >>>> used; for ICE-for-SCTP checks, the same DSCP markings as media packe=
ts
> >>>> (i.e. the SCTP traffic) should be used.
> >>>>
> >>>> If multiplexing is being performed, the connectivity checks probably
> >>>> should use the highest DSCP value being used by the multiplexed medi=
a.
> >>>
> >>> why is (seemingly) everyone's default position "use the highest DSCP
> >>> available" for signaling packets?
> >>>
> >>> You just need to make sure the packets aren't starved or dropped by/a=
t
> >>> congestion points.
> >>
> >> The underlying principle is that connectivity checks need to *check
> >> connectivity* (duh!). That's why you use the same DSCP as the media.
> >> Connectivity is affected by the DSCP. For example some DSCPs could be
> >> filtered, or could be placed in a low-priority queue and get dropped
> >> such that connectivity fails.
> >>
> >> >From this principle follows this conclusion: on a given 5-tuple, if
> your
> >> media uses different DSCPs, you need to check connectivity for all tho=
se
> >> DSCPs. That is, you would send multiple connectivity checks, one for
> >> each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we migh=
t
> >> need an alternative heuristic: try the lowest-priority DSCP, and assum=
e
> >> that if that one works, the higher-priority ones should work as well.
> >> (Note that Justin suggested the contrary.)
> >>
> >> What you want to avoid is STUN packets getting through but not the
> >> corresponding media. That is an unworkable situation. The reverse is n=
ot
> >> ideal either (you think you have no connectivity when in fact you do),
> >> but at least ICE can work around it (by picking a different candidate)=
.
> >
> >
> > This explanation makes sense for (ICE) connectivity checks.
> >
> > For other control information, especially information about delay, you
> want to have it delivered as fast as possible, and with as little loss as
> possible, because jitter in the return path will make the delay signal mo=
re
> noisy, and lost packets may cause drastic actions like those the
> circuit-breakers draft recommends.
> >
> > Whether that always translates to "the highest DSCP code point" is ....
> a good question.
>
> It doesn't.  Keith added af42 support to mosh last year and had several
> reports of packets being dropped with that marking. Worse the drops
> happened after 10 seconds of connectivity.
>
> Ecn markings on the other hand have thus far survived (if occasionally
> stomped on) across the open internet in that protocol.
>

This is a concern of mine as well, which is why we haven't yet flipped the
switch to turn DSCP on by default in Chrome. I think we'll need to do some
experimentation to see how often DSCP makes things worse.

--20cf3071c8ac65e9e204f460cfab
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Mar 11, 2014 at 10:17 AM, Dave Taht <span dir=3D"ltr">&lt;<=
a href=3D"mailto:dave.taht@gmail.com" target=3D"_blank">dave.taht@gmail.com=
</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"HOEnZb"><div class=3D"h5"><p d=
ir=3D"ltr"><br>
On Mar 11, 2014 7:44 AM, &quot;Harald Alvestrand&quot; &lt;<a href=3D"mailt=
o:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt; wrot=
e:<br>
&gt;<br>
&gt; On 03/11/2014 03:02 PM, Simon Perreault wrote:<br>
&gt;&gt;<br>
&gt;&gt; Le 2014-03-10 16:57, James Polk a =C3=A9crit :<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; At 12:16 AM 3/10/2014, Justin Uberti wrote:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I think that the existing guidance should be sufficient in=
<br>
&gt;&gt;&gt;&gt; non-multiplexing cases (i.e. BUNDLE). For consent checks, =
I think the<br>
&gt;&gt;&gt;&gt; same DSCP markings as any other ICE connectivity checks sh=
ould be<br>
&gt;&gt;&gt;&gt; used; for ICE-for-SCTP checks, the same DSCP markings as m=
edia packets<br>
&gt;&gt;&gt;&gt; (i.e. the SCTP traffic) should be used.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; If multiplexing is being performed, the connectivity check=
s probably<br>
&gt;&gt;&gt;&gt; should use the highest DSCP value being used by the multip=
lexed media.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; why is (seemingly) everyone&#39;s default position &quot;use t=
he highest DSCP<br>
&gt;&gt;&gt; available&quot; for signaling packets?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; You just need to make sure the packets aren&#39;t starved or d=
ropped by/at<br>
&gt;&gt;&gt; congestion points.<br>
&gt;&gt;<br>
&gt;&gt; The underlying principle is that connectivity checks need to *chec=
k<br>
&gt;&gt; connectivity* (duh!). That&#39;s why you use the same DSCP as the =
media.<br>
&gt;&gt; Connectivity is affected by the DSCP. For example some DSCPs could=
 be<br>
&gt;&gt; filtered, or could be placed in a low-priority queue and get dropp=
ed<br>
&gt;&gt; such that connectivity fails.<br>
&gt;&gt;<br>
&gt;&gt; &gt;From this principle follows this conclusion: on a given 5-tupl=
e, if your<br>
&gt;&gt; media uses different DSCPs, you need to check connectivity for all=
 those<br>
&gt;&gt; DSCPs. That is, you would send multiple connectivity checks, one f=
or<br>
&gt;&gt; each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we =
might<br>
&gt;&gt; need an alternative heuristic: try the lowest-priority DSCP, and a=
ssume<br>
&gt;&gt; that if that one works, the higher-priority ones should work as we=
ll.<br>
&gt;&gt; (Note that Justin suggested the contrary.)<br>
&gt;&gt;<br>
&gt;&gt; What you want to avoid is STUN packets getting through but not the=
<br>
&gt;&gt; corresponding media. That is an unworkable situation. The reverse =
is not<br>
&gt;&gt; ideal either (you think you have no connectivity when in fact you =
do),<br>
&gt;&gt; but at least ICE can work around it (by picking a different candid=
ate).<br>
&gt;<br>
&gt;<br>
&gt; This explanation makes sense for (ICE) connectivity checks.<br>
&gt;<br>
&gt; For other control information, especially information about delay, you=
 want to have it delivered as fast as possible, and with as little loss as =
possible, because jitter in the return path will make the delay signal more=
 noisy, and lost packets may cause drastic actions like those the circuit-b=
reakers draft recommends.<br>



&gt;<br>
&gt; Whether that always translates to &quot;the highest DSCP code point&qu=
ot; is .... a good question.</p>
</div></div><p dir=3D"ltr">It doesn&#39;t.=C2=A0 Keith added af42 support t=
o mosh last year and had several reports of packets being dropped with that=
 marking. Worse the drops happened after 10 seconds of connectivity.</p>
<p dir=3D"ltr">Ecn markings on the other hand have thus far survived (if oc=
casionally stomped on) across the open internet in that protocol.</p></bloc=
kquote><div><br></div><div>This is a concern of mine as well, which is why =
we haven&#39;t yet flipped the switch to turn DSCP on by default in Chrome.=
 I think we&#39;ll need to do some experimentation to see how often DSCP ma=
kes things worse.</div>

</div></div></div>

--20cf3071c8ac65e9e204f460cfab--


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Date: Tue, 11 Mar 2014 21:21:24 -0700
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Subject: Re: [rtcweb] draft-ietf-rtcweb-rtp-usage-12 Client-to-Mixer Audio Level
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--e89a8ff1c2469ce62d04f4612cde
Content-Type: text/plain; charset=UTF-8

I agree we need to mandate implementation of encryption for the c2m audio
level headers. For actual use, what controls do we want - the ability to
offer one of (encrypted, encrypted + unencrypted, none)?

On Mon, Mar 10, 2014 at 9:47 PM, Bernard Aboba <bernard.aboba@gmail.com>wrote:

> Cullen said:
>
> "I am opposed to mandating that Client-to-Mixer Audio Level RTP header is
> REQUIRED to be encrypted. It means that we can not really build
> conferencing system where the mixers do not have access to decrypt the
> media."
>
> [BA] I am also not enthused about mandating encryption of header
> extensions. However, in the interest of fairness, it is worth pointing out
> that header extensions are optional, so if a particular extension is
> encrypted, the mixer can always ignore it (and in this case, thereby do
> more work mixing muted audio streams).  So my take would be to say SHOULD
> encrypt Client-to-MIxer Audio level, but leave this up to negotiation so
> that if the mixer doesn't want it encrypted, it can turn that off.
>
>
> On Wed, Mar 5, 2014 at 4:32 AM, Cullen Jennings (fluffy) <fluffy@cisco.com
> > wrote:
>
>>
>> I am opposed to mandating that Client-to-Mixer Audio Level RTP header is
>> REQUIRED to be encrypted. It means that we can not really build
>> conferencing system where the mixers do not have access to decrypt the
>> media.
>>
>> I would like to propose that the JS application can control  if the
>> header is encrypted or not.
>>
>> I am aware of the paper that suggests these audio levels can reveal the
>> contents of the conversation. There is some element of truth to this. There
>> is also some element of truth to the point that if this was true, speech
>> recognition system would work better than they do. Encrypting this or not
>> is a risk that the application using the header extension needs to
>> evaluate, and WebRTC system needs to provide the applications with a
>> control surfaces that allows them to use this in the way the applications
>> desires.
>>
>> I will note that an normal application that used this header could decide
>> if it was encrypted or not, what is different in RTCWeb is that we are
>> building a platform and my view is that this platform should allow the
>> applications build on the platform to decide the tradeoffs of risk for
>> encrypting this or not.
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--e89a8ff1c2469ce62d04f4612cde
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div style=3D"font-family:arial,sans-serif;font-size:=
13px"><div>I agree we need to mandate implementation of encryption for the =
c2m audio level headers. For actual use, what controls do we want - the abi=
lity to offer one of (encrypted, encrypted + unencrypted, none)?=C2=A0</div=
>

</div></div><div style=3D"font-family:arial,sans-serif;font-size:13px"><br>=
</div><div class=3D"gmail_extra"><div class=3D"gmail_quote">On Mon, Mar 10,=
 2014 at 9:47 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:ber=
nard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;</sp=
an> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"">Cullen said:=C2=A0<div><b=
r></div>
<div>
&quot;<span style=3D"font-family:arial,sans-serif;font-size:13px">I am oppo=
sed to mandating that Client-to-Mixer Audio Level RTP header is REQUIRED to=
 be encrypted. It means that we can not really build conferencing system wh=
ere the mixers do not have access to decrypt the media.&quot;</span></div>



<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div></div><div><span style=3D"font-family:arial,sans-serif;font-size:13p=
x">[BA] I am also not enthused about mandating encryption of header extensi=
ons. However, in the interest of fairness, it is worth pointing out that he=
ader extensions are optional, so if a particular extension is encrypted, th=
e mixer can always ignore it (and in this case, thereby do more work mixing=
 muted audio streams). =C2=A0So my take would be to say SHOULD encrypt Clie=
nt-to-MIxer Audio level, but leave this up to negotiation so that if the mi=
xer doesn&#39;t want it encrypted, it can turn that off.=C2=A0</span></div>



</div><div class=3D""><div class=3D"h5"><div class=3D"gmail_extra"><br><br>=
<div class=3D"gmail_quote">On Wed, Mar 5, 2014 at 4:32 AM, Cullen Jennings =
(fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=
=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><br>
I am opposed to mandating that Client-to-Mixer Audio Level RTP header is RE=
QUIRED to be encrypted. It means that we can not really build conferencing =
system where the mixers do not have access to decrypt the media.<br>
<br>
I would like to propose that the JS application can control =C2=A0if the he=
ader is encrypted or not.<br>
<br>
I am aware of the paper that suggests these audio levels can reveal the con=
tents of the conversation. There is some element of truth to this. There is=
 also some element of truth to the point that if this was true, speech reco=
gnition system would work better than they do. Encrypting this or not is a =
risk that the application using the header extension needs to evaluate, and=
 WebRTC system needs to provide the applications with a control surfaces th=
at allows them to use this in the way the applications desires.<br>




<br>
I will note that an normal application that used this header could decide i=
f it was encrypted or not, what is different in RTCWeb is that we are build=
ing a platform and my view is that this platform should allow the applicati=
ons build on the platform to decide the tradeoffs of risk for encrypting th=
is or not.<br>




<br>
<br>
<br>
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--e89a8ff1c2469ce62d04f4612cde--


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From: Justin Uberti <juberti@google.com>
Date: Tue, 11 Mar 2014 21:37:33 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Sort of. The idea is whether JSEP should have a mode which _gathers_ only a
single set of ports, in effect assuming the other side supports rtcp-mux.
The use of a=3Drtcp is incidental here, and only used to point out that the
RTP and RTCP ports that are offered will be identical.


On Tue, Mar 11, 2014 at 12:51 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>   "BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp
> attributes. I remember we had discussions about that, but I don=E2=80=99t=
 remember
> the justification at the moment."
>
>  [BA] The justification was that an implementation interested in
> multiplexing A/V would almost certainly also want to multiplex RTP and
> RTCP, because the motivation for both is minimizing port usage.
>
> [Christer] So, again, the idea is to use a=3Drtcp to indicate the *same*
> port as a=3Drtcp-mux indicates, and hope that the receiver supports at le=
ast
> either a=3Drtcp or a=3Drtcp-mux?
>
> Regards,
>
> Christer
>
>
>
> On Mon, Mar 10, 2014 at 1:24 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>>  Hi,
>>
>>
>>
>> Before I comment on your proposal, let=E2=80=99s take a few steps back (=
maybe it
>> can even solve your issue).
>>
>>
>>
>> BUNDLE currently mandates the usage of both the SDP rtcp-mux and rtcp
>> attributes. I remember we had discussions about that, but I don=E2=80=99=
t remember
>> the justification at the moment. But, I wonder whether that text is from
>> before we made a decision that, unless both endpoints support BUNDLE, no
>> endpoint will use a single address:port.
>>
>>
>>
>> So, the question is: do we really need the SDP rtcp attribute?
>>
>>
>>
>> Regards,
>>
>>
>>
>> Christer
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *From:* Justin Uberti [mailto:juberti@google.com]
>> *Sent:* 10. maaliskuuta 2014 8:18
>> *To:* rtcweb@ietf.org; Eric Rescorla; Christer Holmberg
>> *Subject:* BUNDLE with implicit rtcp-mux
>>
>>
>>
>> In my JSEP preso during the RTCWEB session last Wednesday, I discussed
>> adding a new BUNDLE policy to JSEP. This policy, which would be called
>> something like =E2=80=9Cforce-bundle=E2=80=9D, would force the use of th=
e same ports for
>> BUNDLEd m-sections, as well as RTCP mux. When one knows a priori that th=
e
>> other side supports BUNDLE, this would reduce candidate gathering even o=
ver
>> the currently defined =E2=80=9Cmax-bundle=E2=80=9D policy, since there w=
ould be no need to
>> gather RTCP ports at all, and also avoid the need for the BUNDLE fixup
>> offer (where the ports are updated to be the same for all BUNDLEd
>> m-sections).
>>
>> To explain the differences here, here are some example offers for an
>> audio+video call:
>>
>> Default bundle policy - separate ports for audio, video, RTP and RTCP:
>> m=3Daudio 10000
>> a=3Drtcp:10001
>> a=3Drtcp-mux
>> m=3Dvideo 11000
>> a=3Drtcp:11001
>> a=3Drtcp-mux
>>
>> Existing "max-bundle" policy - video port is 0 because bundle-only, but
>> both RTP and RTCP ports allocated for audio:
>> m=3Daudio 10000
>> a=3Drtcp:10001
>> a=3Drtcp-mux
>> m=3Dvideo 0
>> a=3Dbundle-only
>> a=3Drtcp-mux
>>
>> The new =E2=80=9Cforce-bundle=E2=80=9D policy - only a single set of por=
ts allocated -
>> audio, video, RTP, RTCP share the same port.
>> m=3Daudio 10000
>> a=3Drtcp:10000
>> a=3Drtcp-mux
>> m=3Dvideo 10000
>> a=3Drtcp:10000
>> a=3Drtcp-mux
>>
>> During the discussion, the ability to avoid RTCP port gathering was seen
>> as valuable, but not the ability to avoid the fixup offer. However, afte=
r
>> thinking about it a bit, I don=E2=80=99t think that there is a viable so=
lution that
>> avoids RTCP port gathering that *doesn=E2=80=99t* use a single set of po=
rts. The
>> only other option would be to do something like this:
>>
>> m=3Daudio 10000
>> a=3Drtcp:10000 # same as m=3Daudio port
>> a=3Drtcp-mux
>> m=3Dvideo 0
>> a=3Dbundle-only
>> a=3Drtcp-mux
>>
>> but this will almost surely will result in a malfunction when talking to
>> a non-rtcp-mux client, due to the same port being used for RTP and RTCP,
>> and still requires the fixup offer, and as such is net worse than the
>> force-bundle approach indicated above.
>>
>> So I think we the force-bundle approach is the one we want to minimize
>> port gathering. The only question is whether we should keep max-bundle a=
s
>> well. To be specific, we need to decide between:
>>
>> a) Keep max-bundle as it is (separate RTP and RTCP ports allocated), and
>> add a new force-bundle policy that creates an offer with a single port
>> shared between all m-sections (RTP and RTCP). max-bundle remains nominal=
ly
>> backwards compatible with non-BUNDLE/non-rtcp-mux endpoints, in that it
>> will still send a single audio stream. force-bundle is not backwards
>> compatible with non-BUNDLE endpoints.
>>
>> b) Add force-bundle, but get rid of max-bundle.
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

--089e013cb9965c47e604f46166d1
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<div dir=3D"ltr">Sort of. The idea is whether JSEP should have a mode which=
 _gathers_ only a single set of ports, in effect assuming the other side su=
pports rtcp-mux. The use of a=3Drtcp is incidental here, and only used to p=
oint out that the RTP and RTCP ports that are offered will be identical.</d=
iv>

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue, Mar 1=
1, 2014 at 12:51 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@eri=
csson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">




<div>
<div style=3D"direction:ltr;font-size:10pt;font-family:Tahoma">Hi,<br>
<br>
<div style=3D"font-size:16px;font-family:Times New Roman">
<div><div class=3D"">
<div dir=3D"ltr">
<div>&quot;<span style=3D"font-size:11pt;font-family:Calibri,sans-serif;col=
or:rgb(31,73,125)">BUNDLE currently mandates the usage of both the SDP rtcp=
-mux and rtcp attributes. I remember we had discussions about that, but I d=
on=E2=80=99t remember the justification at the
 moment.</span>&quot;</div>
<div><br>
</div>
<div>[BA] The justification was that an implementation interested in multip=
lexing A/V would almost certainly also want to multiplex RTP and RTCP, beca=
use the motivation for both is minimizing port usage.=C2=A0</div>
</div>
</div><div class=3D"gmail_extra"><br>
[Christer] So, again, the idea is to use a=3Drtcp to indicate the *same* po=
rt as a=3Drtcp-mux indicates, and hope that the receiver supports at least =
either a=3Drtcp or a=3Drtcp-mux?<br>
<br>
Regards,<br>
<br>
Christer<div><div class=3D"h5"><br>
<br>
<br>
<div class=3D"gmail_quote">On Mon, Mar 10, 2014 at 1:24 AM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div lang=3D"EN-US">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Before I comment on your =
proposal, let=E2=80=99s take a few steps back (maybe it can even solve your=
 issue).<u></u><u></u></span></p>


<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">BUNDLE currently mandates=
 the usage of both the SDP rtcp-mux and rtcp attributes. I remember we had =
discussions about that, but I don=E2=80=99t remember the justification
 at the moment. But, I wonder whether that text is from before we made a de=
cision that, unless both endpoints support BUNDLE, no endpoint will use a s=
ingle address:port.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">So, the question is: do w=
e really need the SDP rtcp attribute?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Justin U=
berti [mailto:<a href=3D"mailto:juberti@google.com" target=3D"_blank">juber=
ti@google.com</a>]
<br>
<b>Sent:</b> 10. maaliskuuta 2014 8:18<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a>; Eric Rescorla; Christer Holmberg<br>
<b>Subject:</b> BUNDLE with implicit rtcp-mux<u></u><u></u></span></p>
<div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal">In my JSEP preso during the RTCWEB session last Wedn=
esday, I discussed adding a new BUNDLE policy to JSEP. This policy, which w=
ould be called something like =E2=80=9Cforce-bundle=E2=80=9D, would force t=
he use of the same ports for BUNDLEd m-sections, as
 well as RTCP mux. When one knows a priori that the other side supports BUN=
DLE, this would reduce candidate gathering even over the currently defined =
=E2=80=9Cmax-bundle=E2=80=9D policy, since there would be no need to gather=
 RTCP ports at all, and also avoid the need for
 the BUNDLE fixup offer (where the ports are updated to be the same for all=
 BUNDLEd m-sections).<u></u><u></u></p>
<p class=3D"MsoNormal">To explain the differences here, here are some examp=
le offers for an audio+video call:<u></u><u></u></p>
<p class=3D"MsoNormal">Default bundle policy - separate ports for audio, vi=
deo, RTP and RTCP:<br>
m=3Daudio 10000<br>
a=3Drtcp:10001<br>
a=3Drtcp-mux<br>
m=3Dvideo 11000<br>
a=3Drtcp:11001<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">Existing &quot;max-bundle&quot; policy - video port =
is 0 because bundle-only, but both RTP and RTCP ports allocated for audio:<=
br>
m=3Daudio 10000<br>
a=3Drtcp:10001<br>
a=3Drtcp-mux<br>
m=3Dvideo 0<br>
a=3Dbundle-only<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">The new =E2=80=9Cforce-bundle=E2=80=9D policy - only=
 a single set of ports allocated - audio, video, RTP, RTCP share the same p=
ort.<br>
m=3Daudio 10000<br>
a=3Drtcp:10000<br>
a=3Drtcp-mux<br>
m=3Dvideo 10000<br>
a=3Drtcp:10000<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">During the discussion, the ability to avoid RTCP por=
t gathering was seen as valuable, but not the ability to avoid the fixup of=
fer. However, after thinking about it a bit, I don=E2=80=99t think that the=
re is a viable solution that avoids RTCP port
 gathering that *doesn=E2=80=99t* use a single set of ports. The only other=
 option would be to do something like this:<u></u><u></u></p>
<p class=3D"MsoNormal">m=3Daudio 10000<br>
a=3Drtcp:10000 # same as m=3Daudio port<br>
a=3Drtcp-mux<br>
m=3Dvideo 0<br>
a=3Dbundle-only<br>
a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal">but this will almost surely will result in a malfunc=
tion when talking to a non-rtcp-mux client, due to the same port being used=
 for RTP and RTCP, and still requires the fixup offer, and as such is net w=
orse than the force-bundle approach
 indicated above.<u></u><u></u></p>
<p class=3D"MsoNormal">So I think we the force-bundle approach is the one w=
e want to minimize port gathering. The only question is whether we should k=
eep max-bundle as well. To be specific, we need to decide between:<u></u><u=
></u></p>


<p class=3D"MsoNormal">a) Keep max-bundle as it is (separate RTP and RTCP p=
orts allocated), and add a new force-bundle policy that creates an offer wi=
th a single port shared between all m-sections (RTP and RTCP). max-bundle r=
emains nominally backwards compatible
 with non-BUNDLE/non-rtcp-mux endpoints, in that it will still send a singl=
e audio stream. force-bundle is not backwards compatible with non-BUNDLE en=
dpoints.<u></u><u></u></p>
<p class=3D"MsoNormal">b) Add force-bundle, but get rid of max-bundle.<u></=
u><u></u></p>
</div>
</div>
</div>
</div>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div></div></div>
</div>
</div>
</div>
</div>

</blockquote></div><br></div>

--089e013cb9965c47e604f46166d1--


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Future versions of the API should have the knobs needed to control the
CNAMEs more exactly. For WebRTC 1.0 though, scoping CNAMEs to a
PeerConnection feels like a prudent simplification.


On Mon, Mar 10, 2014 at 8:49 AM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 10 March 2014 16:19, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
> > Can you please think through the linkage issue a bit first before
> > changing this property.
>
> We already came to the conclusion that if media was being forwarded by
> a peer, then that is new media originated by that peer, not the same
> media.  Thus, if looping occurs, it is at the application layer.  We
> definitely already agreed that RTP forwarding is not a function that a
> browser will support.
>
> Obviously, if we add that feature, CNAMEs will be just one of the
> manifold issues that will arise and we will have to reconsider.
> Compromising the privacy of users doesn't seem like a good trade-off
> against some minor advantages in terms of loop detection (unlikely)
> and synchronization (easy to apply manually).
>

--047d7b33c70631aa3a04f461705a
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<div dir=3D"ltr">Future versions of the API should have the knobs needed to=
 control the CNAMEs more exactly. For WebRTC 1.0 though, scoping CNAMEs to =
a PeerConnection feels like a prudent simplification.</div><div class=3D"gm=
ail_extra">

<br><br><div class=3D"gmail_quote">On Mon, Mar 10, 2014 at 8:49 AM, Martin =
Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" t=
arget=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">

On 10 March 2014 16:19, Magnus Westerlund<br>
<div class=3D"">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnu=
s.westerlund@ericsson.com</a>&gt; wrote:<br>
&gt; Can you please think through the linkage issue a bit first before<br>
&gt; changing this property.<br>
<br>
</div>We already came to the conclusion that if media was being forwarded b=
y<br>
a peer, then that is new media originated by that peer, not the same<br>
media. =C2=A0Thus, if looping occurs, it is at the application layer. =C2=
=A0We<br>
definitely already agreed that RTP forwarding is not a function that a<br>
browser will support.<br>
<br>
Obviously, if we add that feature, CNAMEs will be just one of the<br>
manifold issues that will arise and we will have to reconsider.<br>
Compromising the privacy of users doesn&#39;t seem like a good trade-off<br=
>
against some minor advantages in terms of loop detection (unlikely)<br>
and synchronization (easy to apply manually).<br>
</blockquote></div><br></div>

--047d7b33c70631aa3a04f461705a--


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Dear all,
I've jotted down the following notes when reading the drafts from here and
the W3C as potential problem areas. Maybe there are things I've missed in
the drafts that address them, but I think they are still worth thinking
about. Some of these are more W3C, but we seem to be in charge of the
security. These issues vary widely in seriousness. One of them is a
demonstrated break of confidentiality, while several are open questions
about how we communicate to users.

Problem 1: Which John Smith?

Currently IdPs link attributes to identities. But the way in which
identities are presented to the user can be misleading. Names are not
unique. Do we need a way for an IdP to indicate more about the identity?
Should IdPs be allowed to use social graph information, e.g. prevent me
from calling people who I don't "friend" on a social network, and thus
avoid the wrong John Smith? How does this information get handled? Who
decides what policy is being used?

Problem 2: IdP pointy bits

An IdP has to cryptographically bind a username to some data. It would be
nice if we could come up with a secure method to do this, and have the
browser take care of it for the identity provider as much as possible,
instead of everyone having to do it themselves in likely wrong ways. The
webcrypto API has been roundly critiqued as being too low-level.

Problem 3: Renegotiation

The DTLS handshake being used is specified in RFC 4347. In particular, I
don't see anything about the renegotiation bug being fixed. This enables an
unknown key-share attack in which A thinks they are transmitting video to
B, but really are transmitting it to C who knows that they are talking to A.

Problem 4: HMAC-SHA1 in SRTP

If I've chased the chain of references correctly this is the sole MAC
provided. Is it okay? I have no idea: SHA-1 has been significantly weakened
in recent years.

Problem 5: What you see is not what I sent

This is an area that is somewhat underspecified, depending as it does on
the media API's capabilities. The video frame could be silently replaced by
a different video frame with rather different contents. Any authentication
of the contents of the video frame would not carry over, but if the
contents were sufficiently shocking/subtle this might not be noticed.
Authentication independent of the website offering the video chat makes
this attack worse, because users will trust what they see more. Isolation
doesn't help because this is wholesale replacement, not editing.

Most uses of this attack are for puerile purposes. I don't think it's a big
concern. Then again I could be very, very wrong.

Problem 6: General UI issues

Currently quite a bit of the security model depends on UI presentations of
subtly different states. I don't know how well users will understand this,
or how well the UI will work. Past experience suggests not well. Luckily we
can mostly change this later/it isn't part of standardisation.

Problem 7: VBR privacy leaks

I can do no better then to refer to the paper.
http://www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/reports/yes-we-can.pdfSpoken
phrases could be identified from encrypted data alone, using a
Hidden Markov Model when the length of packets was preserved by the
encryption.

Conclusion:
Some of these issues result from well-known problems in older protocols.
Others are longstanding difficult problems that have to be solved. Some are
quite dangerous, while others will only permit the occasional prank.
However, it is not enough to fix these specifically. Large areas of the
proposal have not been explored by me. If I had to rate these by
seriousness, problem 3 and 7 are the worst. They are luckily the easiest to
fix.

Sincerely,
Watson Ladd

--20cf303a3183337dca04f461deac
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Dear all,<br>I&#39;ve jotted down the following notes when=
 reading the drafts from here and the W3C as potential problem areas. Maybe=
 there are things I&#39;ve missed in the drafts that address them, but I th=
ink they are still worth thinking about. Some of these are more W3C, but we=
 seem to be in charge of the security. These issues vary widely in seriousn=
ess. One of them is a demonstrated break of confidentiality, while several =
are open questions about how we communicate to users.<br>
<br>Problem 1: Which John Smith?<br><br>Currently IdPs link attributes to i=
dentities. But the way in which identities are presented to the user can be=
 misleading. Names are not unique. Do we need a way for an IdP to indicate =
more about the identity? Should IdPs be allowed to use social graph informa=
tion, e.g. prevent me from calling people who I don&#39;t &quot;friend&quot=
; on a social network, and thus avoid the wrong John Smith? How does this i=
nformation get handled? Who decides what policy is being used?<br>
<br>Problem 2: IdP pointy bits<br><br>An IdP has to cryptographically bind =
a username to some data. It would be nice if we could come up with a secure=
 method to do this, and have the browser take care of it for the identity p=
rovider as much as possible, instead of everyone having to do it themselves=
 in likely wrong ways. The webcrypto API has been roundly critiqued as bein=
g too low-level.<br>
<br><div>Problem 3: Renegotiation</div><div><br></div><div>The DTLS handsha=
ke being used is specified in RFC 4347. In particular, I don&#39;t see anyt=
hing about the renegotiation bug being fixed. This enables an unknown key-s=
hare attack in which A thinks they are transmitting video to B, but really =
are transmitting it to C who knows that they are talking to A.</div>
<div><br></div><div>Problem 4: HMAC-SHA1 in SRTP</div><div><br></div><div>I=
f I&#39;ve chased the chain of references correctly this is the sole MAC pr=
ovided. Is it okay? I have no idea: SHA-1 has been significantly weakened i=
n recent years.</div>
<div><br></div><div>Problem 5: What you see is not what I sent</div><div><b=
r></div><div>This is an area that is somewhat underspecified, depending as =
it does on the media API&#39;s capabilities. The video frame could be silen=
tly replaced by a different video frame with rather different contents. Any=
 authentication of the contents of the video frame would not carry over, bu=
t if the contents were sufficiently shocking/subtle this might not be notic=
ed. Authentication independent of the website offering the video chat makes=
 this attack worse, because users will trust what they see more. Isolation =
doesn&#39;t help because this is wholesale replacement, not editing.</div>
<div><br></div><div>Most uses of this attack are for puerile purposes. I do=
n&#39;t think it&#39;s a big concern. Then again I could be very, very wron=
g.</div><div><br></div><div>Problem 6: General UI issues</div><div><br>
</div><div>Currently quite a bit of the security model depends on UI presen=
tations of subtly different states. I don&#39;t know how well users will un=
derstand this, or how well the UI will work. Past experience suggests not w=
ell. Luckily we can mostly change this later/it isn&#39;t part of standardi=
sation.<br>
</div><div><br></div><div>Problem 7: VBR privacy leaks</div><div><br></div>=
<div>I can do no better then to refer to the paper.=C2=A0<a href=3D"http://=
www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/reports/yes-we-can.pdf"=
>http://www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/reports/yes-we-=
can.pdf</a> Spoken phrases could be identified from encrypted data alone, u=
sing a Hidden Markov Model when the length of packets was preserved by the =
encryption.</div>
<div><br></div><div>Conclusion:</div><div>Some of these issues result from =
well-known problems in older protocols. Others are longstanding difficult p=
roblems that have to be solved. Some are quite dangerous, while others will=
 only permit the occasional prank. However, it is not enough to fix these s=
pecifically. Large areas of the proposal have not been explored by me. If I=
 had to rate these by seriousness, problem 3 and 7 are the worst. They are =
luckily the easiest to fix.</div>
<div><br></div><div>Sincerely,</div><div>Watson Ladd</div></div>

--20cf303a3183337dca04f461deac--


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On 2014-03-12 01:58, Justin Uberti wrote:
> The reason I suggested SHOULD NOT is because I think this technique is
> unnecessary for WebRTC, and typically net harmful for the reasons that
> Matthew mentions. 
> 
> That said, I would be fine with saying MAY and including some
> explanatory text; my main concerns was to avoid the ambiguity that would
> arise from leaving this out altogether.

Okay, as we have considerations, I would leave the RFC 2119 terms out. I
think a statement along the lines of the below might be most suitable.

Using TURN to provide TCP relay candidates [REF] was considered but not
seen as providing any significant benefit. First, use of TURN TCP would
only be relevant in cases which both peers requires to use TCP to
establish a PeerConnection. Secondly, that use case is anyway supported
by both side establishing UDP relay candidates using TURN over TCP
[REF]. Thirdly, using TCP only between the endpoint and its relay may
result in less issues with TCP in regards to real-time constraints, e.g.
due to head of line blocking.

Please provide feedback or word smith.

Cheers

Magnus
(As individual)

> 
> 
> On Mon, Mar 10, 2014 at 8:33 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     On 2014-03-06 14:34, Simon Perreault wrote:
>     > Le 2014-03-06 13:23, Justin Uberti a Ã©crit :
>     >> can we agree that TURN TCP candidates are a SHOULD NOT?
>     >
>     > Not a SHOULD, sure. SHOULD NOT, no.
>     >
> 
>     I would guess that the simplest is to remove discussion of TURN TCP
>     altogether from Transports. That would not recommend it nor make it
>     disallowed. If we want to be explicit, then simply motivate why we don't
>     believe it necessary.
> 
>     Justin, what is the reason you wanted SHOULD NOT?
> 
>     cheers
> 
>     Magnus Westerlund
> 
>     ----------------------------------------------------------------------
>     Services, Media and Network features, Ericsson Research EAB/TXM
>     ----------------------------------------------------------------------
>     Ericsson AB                 | Phone  +46 10 7148287
>     <tel:%2B46%2010%207148287>
>     FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
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>     SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>     <mailto:magnus.westerlund@ericsson.com>
>     ----------------------------------------------------------------------
> 
> 


-- 

Magnus Westerlund

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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On 2014-03-12 04:53, Justin Uberti wrote:
> 
> 
> 
> On Tue, Mar 11, 2014 at 7:02 AM, Simon Perreault
> <simon.perreault@viagenie.ca <mailto:simon.perreault@viagenie.ca>> wrote:
> 
>     Le 2014-03-10 16:57, James Polk a écrit :
>     > At 12:16 AM 3/10/2014, Justin Uberti wrote:
>     >> I think that the existing guidance should be sufficient in
>     >> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think the
>     >> same DSCP markings as any other ICE connectivity checks should be
>     >> used; for ICE-for-SCTP checks, the same DSCP markings as media
>     packets
>     >> (i.e. the SCTP traffic) should be used.
>     >>
>     >> If multiplexing is being performed, the connectivity checks probably
>     >> should use the highest DSCP value being used by the multiplexed
>     media.
>     >
>     > why is (seemingly) everyone's default position "use the highest DSCP
>     > available" for signaling packets?
>     >
>     > You just need to make sure the packets aren't starved or dropped by/at
>     > congestion points.
> 
>     The underlying principle is that connectivity checks need to *check
>     connectivity* (duh!). That's why you use the same DSCP as the media.
>     Connectivity is affected by the DSCP. For example some DSCPs could be
>     filtered, or could be placed in a low-priority queue and get dropped
>     such that connectivity fails.
> 
>     >From this principle follows this conclusion: on a given 5-tuple, if
>     your
>     media uses different DSCPs, you need to check connectivity for all those
>     DSCPs. That is, you would send multiple connectivity checks, one for
>     each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we might
>     need an alternative heuristic: try the lowest-priority DSCP, and assume
>     that if that one works, the higher-priority ones should work as well.
>     (Note that Justin suggested the contrary.)
> 
>     What you want to avoid is STUN packets getting through but not the
>     corresponding media. That is an unworkable situation. The reverse is not
>     ideal either (you think you have no connectivity when in fact you do),
>     but at least ICE can work around it (by picking a different candidate).
> 
> 
> We also need to consider the scenario of STUN consent checks, which will
> need to get through reliably even when media marked with DSCP is already
> flowing. In this case I think that using the highest DSCP value for STUN
> is unavoidable.
> 

(as individual)

Sorry, I don't understand your reasoning here. Is it to provide the
lowest possible drop probability, i.e. using the AFx class that has
least drop probability?

If there exist nodes that stomp on some DSCP markings, and not only
remark them to Best Effort (BE), and instead drop then. Then, it is not
obvious that that an AFx class is the most appropriate to use. In that
case using BE might be more suitable for the consent freshness.

I would also point out that highest DSCP value isn't clear. We are
talking about forwarding behavior and I don't see how Expedited
Forwarding (EF) can be compared with AF for example.

I think we have to consider what our goals are here. To determine if we
can get any packets through or determine if a particular 6-tuple (Source
Address, Destination Address, Source Port, Destination Port, Protocol,
and DSCP value) works? If it is the later then I think we need to have
checks running on all the 6-tuples in use. This clearly have overhead
concerns and helps worsening the ICE The Voice Hammer Attack (See
section 18.5.1. of RFC 5245).

If the goal is the first, isn't using BE a reasonable choice. Yes, it
might have higher drop probability than AF, but it is likely to be
robuster in general deployments and it determine if the peer is there or
not. Dealing with broken 6-tuples needs to be a separate issue.

Cheers

Magnus Westerlund

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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On 2014-03-10 21:09, Christer Holmberg wrote:
> Hi,
> 
>>>>>>> Before I comment on your proposal, let's take a few steps
>>>>>>> back (maybe it can even solve your issue).
>>>>>>> 
>>>>>>> BUNDLE currently mandates the usage of both the SDP
>>>>>>> rtcp-mux and rtcp attributes. I remember we had
>>>>>>> discussions about that, but I don't remember the
>>>>>>> justification at the moment. But, I wonder whether that
>>>>>>> text is from before we made a decision that, unless both
>>>>>>> endpoints support BUNDLE, no endpoint will use a single
>>>>>>> address:port.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> So, the question is: do we really need the SDP rtcp
>>>>>>> attribute?
>>>>>> 
>>>>>> I think it is a question of how you want the fallback to
>>>>>> work. In the case of bundle only lines, a non  bundle
>>>>>> supporting peer will reject them, thus no issues with what
>>>>>> is written around RTCP-mux and a=rtcp in those m= blocks.
>>>>>> However, the first one if you want that to support fallback
>>>>>> that makes sense then you will need both a=rtcp-mux and an
>>>>>> a=rtcp (with a different port) to handle that with optimal
>>>>>> backwards compatibility.
>>>>>> 
>>>>>> Regarding the motivation for a=rtcp-mux and a=rtcp, I think
>>>>>> it is reasonably straight forward. We want a=rtcp-mux to
>>>>>> minimize port usage, i.e. ports=1. Not supporting
>>>>>> a=rtcp-mux with bundle that would mean two ports (RTP and
>>>>>> RTCP) for the bundle group. To get the backwards
>>>>>> compatibility the usage of a=rtcp would be required.
>>>>> 
>>>>> Keep in mind that the offerer must always be prepared that
>>>>> the answerer does not support the rtcp-mux attribute (or the
>>>>> rtcp attribute).
>>>> 
>>>> Yes, that was what I meant with fallback in the above text.
>>>> What you do in case the peer doesn't support bundle.
>>>> 
>>>>> So, at least my understanding is: no matter whether BUNDLE is
>>>>> used or not, if the answer does not contain a rtcp-mux (e.g.
>>>>> in case of fallback) attribute, the offerer must use the
>>>>> default "rtp+1" port.
>>>> 
>>>> I have a tendency to agree, which means that the first m= block
>>>> in a bunde-only group will be required to contain a=rtcp and if
>>>> RTCP mux is desired also that attribute.
>>>> 
>>>> I think a=rtcp could be skipped for a bound-only m= block due
>>>> to the requirement on supporting a=rtcp-mux if you do bundle.
>>> 
>>> For bundle-only, I actually think BOTH can be skipped, because 
>>> bundle-only will by default use whatever address:port is
>>> negotiated - both for RTP and RTCP.
>> 
>> No, I see a path of grief to not explicitly signal things when you
>> use them.
> 
> But, the whole idea of using port zero in bundle-only was that, if
> the answerer supports BUNDLE, port zero will be replaced with the
> negotiated address:port.
> 
> When you send the offer, you don't yet know which address:port will
> be selected, so there is no idea to insert a=rtcp. Inserting
> a=rtcp-mux probably doesn't harm, though, so I won't argue about it
> :)

Well, if you can't insert the port, the same applies to a=rtcp as to the
port in the m= line.

Because in the bundle-only case we are assuming and that assumption
needs to apply also for a=rtcp. And for a bundle-only case I think the
appropriate thing is to assume that a=rtcp-mux will be honored. To have
it being honored you need explicit signal it. The a=rtcp can be
considered redundant in this case, but I am sensitive to removing it as
that would create a special case for a=bundle-only lines. I think it is
better to simply include it, even if it will say a=rtcp:0

> 
>> I also thought the a=rtcp-mux is a MUST to implement, not a MUST
>> use.
> 
> Currently the draft says MUST use. But, that may be a mistake, or a
> leftover from previous procedures. I agree that one should be able to
> use BUNDLE also without rtcp-mux (i.e. using separate BUNDLE ports
> for the RTP traffic and the RTCP traffic).

I hope if people disagree that they shout now.

> 
>>> But, for non-bundle-only, I still ask whether we really need
>>> a=rtcp. Isn't a=rtcp-mux enough? If the answerer supports it, you
>>> will use it, otherwise you will use rtp+1.
>> 
>> My understanding is that a=rtcp has been required in any
>> non-private network deployments due to >NATs for the last 10 years.
>> Thus, I don't see how you can remove them and expect it to work,
>> unless >you are doing a=rtcp-mux. Thus, my view would be that you
>> can remove a=rtcp if you know that the >peer supprots a=rtcp-mux.
> 
> Are you saying that we should use a=rtcp ONLY to indicate the RTP
> port, in case the answerer does not support a=rtcp-mux?

It has its normal usage, i.e. RTCP port in cases when a=rtcp-mux is not
agreed on. In a bundle-only offer, I think the only reasonable approach
is to handle it identically to the m= line port number.

> 
>>> Otherwise, assume that the answerer supports BOTH a=rtcp and 
>>> a=rtcp-mux. Which has higher "priority"? How does the answerer
>>> knows whether the offerer wants to use rtcp-mux, or whether it
>>> wants to use whatever port is indicated in a=rtcp?
>> 
>> That is a=rtcp-mux per RFC 5761 that has higher priority, please
>> see Section 5.1.1.
> 
> You are right - my mistake.
> 
> But, never the less, if the offerer sends both a=rtcp-mux and a=rtcp,
> unless a=rtcp points to the rtp port, the offerer needs to be
> prepared to use 3 different ports for RTCP (rtp+1 port, rtp port and
> rtp+<a=rtcp value>), because it does not know which attribute (if
> any) the answerer supports, and which (if any) the answerer is
> willing to use.

This is actually not necessary, if one can get the rtp+1 port for RTCP,
then include that in the a=rtcp line and everything is fine.

I was under the impression that almost all SIP SDP O/A was using a=rtcp.
But if that is a misunderstanding from my perspective, I still think the
rules will hold. If you don't include the a=rtcp attribute, then the
rtp+1 port is the equivalent of the a=rtcp value.

>From a bundle perspective, I think what needs to be considered is the
actual or implied values on the m= blocks for the rtcp port when using
bundle-only as well as the other case.

Cheers

Magnus Westerlund
(As individual)

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----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 2014-03-10 16:49, Martin Thomson wrote:
> On 10 March 2014 16:19, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> Can you please think through the linkage issue a bit first before
>> changing this property.
> 
> We already came to the conclusion that if media was being forwarded by
> a peer, then that is new media originated by that peer, not the same
> media.  Thus, if looping occurs, it is at the application layer.  We
> definitely already agreed that RTP forwarding is not a function that a
> browser will support.

I like to point out that the agreement and what is documented in
rtp-usage is that WebRTC endpoint will have to make forwarded streams
appear as locally originated. However, this as currently written does
not apply to RTP middleboxes that interconnects multiple PeerConnections
to form a multi-party session. This is deliberate to ensure that RTP
topologies like RTP mixer and SFM do work on the RTP/RTCP level.

> 
> Obviously, if we add that feature, CNAMEs will be just one of the
> manifold issues that will arise and we will have to reconsider.
> Compromising the privacy of users doesn't seem like a good trade-off
> against some minor advantages in terms of loop detection (unlikely)
> and synchronization (easy to apply manually).

I am especially interested to know how you will "easy to apply manually"
the synchronization. Can you please describe that. Because, that either
requires an API call to tell the media framework, please consider the
following CNAMES as equivalent, or some other method of telling the
media framework that these different MST are actually originating from a
common clock base.

I protested about this, not to lower the users privacy, but over the
concern that this was raised without providing a case where it was
obvious that the user's privacy was impacted. Martin, you said that you
would think about it, and the next statement was lets change it, without
providing any motivation why the concern was significant.

>From my perspective, if you can provide a good explanation how to
actually accomplish the synchronization, I think a change of scoping the
CNAME to a PeerConnection is okay. But, I would like to understand how
you believe it is accomplished.


Magnus Westerlund
(As RTP-usage author)

----------------------------------------------------------------------
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----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
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From nobody Wed Mar 12 01:35:41 2014
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On Wed, Mar 12, 2014 at 3:55 AM, Justin Uberti <juberti@google.com> wrote:
>
>
>
> On Tue, Mar 11, 2014 at 10:17 AM, Dave Taht <dave.taht@gmail.com> wrote:


>> >
>> > Whether that always translates to "the highest DSCP code point" is ...=
.
>> > a good question.
>>
>> It doesn't.  Keith added af42 support to mosh last year and had several
>> reports of packets being dropped with that marking. Worse the drops happ=
ened
>> after 10 seconds of connectivity.
>>
>> Ecn markings on the other hand have thus far survived (if occasionally
>> stomped on) across the open internet in that protocol.
>
>
> This is a concern of mine as well, which is why we haven't yet flipped th=
e
> switch to turn DSCP on by default in Chrome. I think we'll need to do som=
e
> experimentation to see how often DSCP makes things worse.

Well, while I'm at this, I note that how linux handles DSCP in WIFI 802.11e
is generally suboptimal. the CS6 and CS7 bit patterns map to the VO
queue which is not aggregatable in wireless-n, CS4 and CS5
map to the VI queue which has a lot of good properties for videoconference
and voice traffic (but limited aggregation), and CS1 and CS2, which map
to the background queue, which has good aggregation but limited txops.

I no longer have the bit patterns for AFxx memorized, but basically
all that was returned to the 802.11e classifier was dscp >> 5 up until very
recently. Lastly, there really is no interpretation whatsoever of the AF
classes equating to "drop probability" in wifi (at least in lnux), they mer=
ely
control queue selection and

it is generally better with wireless-n to aim for better aggregation in
one queue rather than using multiple queues as the cost of acquiring
the media dominates.



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html


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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

>> But, the whole idea of using port zero in bundle-only was that, if the=20
>> answerer supports BUNDLE, port zero will be replaced with the=20
>> negotiated address:port.
>>=20
>> When you send the offer, you don't yet know which address:port will be=20
>> selected, so there is no idea to insert a=3Drtcp. Inserting a=3Drtcp-mux=
=20
>> probably doesn't harm, though, so I won't argue about it
>> :)
>
> Well, if you can't insert the port, the same applies to a=3Drtcp as to th=
e port in the m=3D line.

Exactly.=20

The bundle-only m- lines will "inherit" the port that is selected for the w=
hole BUNDLE group.

> Because in the bundle-only case we are assuming and that assumption needs=
 to apply also for a=3Drtcp. And for a bundle-only case=20
> I think the appropriate thing is to assume that a=3Drtcp-mux will be hono=
red. To have it being honored you need explicit signal it.=20

Sure. But, the important thing is to include a=3Drtcp-mux in the m- line th=
at will be used for selecting the BUNDLE address. That will determine wheth=
er rtcp-mux will be used or not.

> The a=3Drtcp can be considered redundant in this case, but I am sensitive=
 to removing it as that would create a special case for a=3Dbundle-only lin=
es. I think it is better to simply include it, even if it will say a=3Drtcp=
:0

We can do that, if people think it's a good idea.


>>> I also thought the a=3Drtcp-mux is a MUST to implement, not a MUST use.
>>=20
>> Currently the draft says MUST use. But, that may be a mistake, or a=20
>> leftover from previous procedures. I agree that one should be able to=20
>> use BUNDLE also without rtcp-mux (i.e. using separate BUNDLE ports for=20
>> the RTP traffic and the RTCP traffic).
>
> I hope if people disagree that they shout now.

And shout loud :)


>>>>> But, for non-bundle-only, I still ask whether we really need a=3Drtcp=
.=20
>>>>> Isn't a=3Drtcp-mux enough? If the answerer supports it, you will use=
=20
>>>>> it, otherwise you will use rtp+1.
>>>>=20
>>> My understanding is that a=3Drtcp has been required in any non-private=
=20
>>> network deployments due to >NATs for the last 10 years.
>>> Thus, I don't see how you can remove them and expect it to work,=20
>>> unless >you are doing a=3Drtcp-mux. Thus, my view would be that you can=
=20
>>> remove a=3Drtcp if you know that the >peer supprots a=3Drtcp-mux.
>>=20
>> Are you saying that we should use a=3Drtcp ONLY to indicate the RTP=20
>> port, in case the answerer does not support a=3Drtcp-mux?
>
> It has its normal usage, i.e. RTCP port in cases when a=3Drtcp-mux is not=
 agreed on. In a bundle-only offer, I think the only reasonable approach is=
 to handle it identically to the m=3D line port number.

I don't know what you mean by "bundle-only offer". There must exist at leas=
t one non-bundle-only m- line in an offer, in order to indicate the port va=
lue. And, whatever a=3Drtcp value is used in that m- line will also be appl=
ied to the bundle-only m- lines is the BUNDLE group is accepted.
=20
>>>> Otherwise, assume that the answerer supports BOTH a=3Drtcp and=20
>>>> a=3Drtcp-mux. Which has higher "priority"? How does the answerer knows=
=20
>>>> whether the offerer wants to use rtcp-mux, or whether it wants to=20
>>>> use whatever port is indicated in a=3Drtcp?
>>>=20
>>> That is a=3Drtcp-mux per RFC 5761 that has higher priority, please see=
=20
>>> Section 5.1.1.
>>=20
>> You are right - my mistake.
>>=20
>> But, never the less, if the offerer sends both a=3Drtcp-mux and a=3Drtcp=
,=20
>> unless a=3Drtcp points to the rtp port, the offerer needs to be prepared=
=20
>> to use 3 different ports for RTCP (rtp+1 port, rtp port and
>> rtp+<a=3Drtcp value>), because it does not know which attribute (if
>> any) the answerer supports, and which (if any) the answerer is willing=20
>> to use.
>
> This is actually not necessary, if one can get the rtp+1 port for RTCP, t=
hen include that in the a=3Drtcp line and everything is fine.
>
> I was under the impression that almost all SIP SDP O/A was using a=3Drtcp=
.
> But if that is a misunderstanding from my perspective, I still think the =
rules will hold. If you don't include the a=3Drtcp attribute, then the
> rtp+1 port is the equivalent of the a=3Drtcp value.
>
> From a bundle perspective, I think what needs to be considered is the act=
ual or implied values on the m=3D blocks for the rtcp port when using bundl=
e-only as well as the other case.

It may be that we agree, but talk past each other :)

My understanding of bundle-only has been that it is a mechanism to ensure t=
hat the m- line is accepted only if the BUNDLE group is accepted, e.g. in o=
rder to avoid having to reserve extra ports. BUT, the bundle-only m- lines =
are NOT used to negotiate the actual BUNDLE address - they will inherit the=
 selected BUNDLE address, negotiated using non-bundle-only m- lines.

Regards,

Christer


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References: <CAOJ7v-063mGE_zndtAqAMN2fJw7kWvUHX2SgyDMryfw_-aS-9w@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D1F1892@ESESSMB209.ericsson.se> <531DC52B.6020500@ericsson.com> <7594FB04B1934943A5C02806D1A2204B1D1FA6BC@ESESSMB209.ericsson.se> <531DCBE9.70701@ericsson.com> <7594FB04B1934943A5C02806D1A2204B1D1FA7B2@ESESSMB209.ericsson.se> <531DD756.50900@ericsson.com> <7594FB04B1934943A5C02806D1A2204B1D200086@ESESSMB209.ericsson.se> <532017DD.1060500@ericsson.com> <7594FB04B1934943A5C02806D1A2204B1D20A684@ESESSMB209.ericsson.se>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On 2014-03-12 09:47, Christer Holmberg wrote:
> It may be that we agree, but talk past each other :)
> 
> My understanding of bundle-only has been that it is a mechanism to
> ensure that the m- line is accepted only if the BUNDLE group is
> accepted, e.g. in order to avoid having to reserve extra ports. BUT,
> the bundle-only m- lines are NOT used to negotiate the actual BUNDLE
> address - they will inherit the selected BUNDLE address, negotiated
> using non-bundle-only m- lines.
> 

I think we are agreeing. I think my main point is that the m= blocks
that contains bundle-only needs to contain the attribute a=rtcp-mux if
that is included in the first m= block for the bundle group that
contains the actual addresses. Similar with the a=rtcp attribute. Simply
to be explicit that they are configured similarly. This I think implies
that we need a "to be replaced" value also for a=rtcp in the bundle only
m= blocks.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Justin Uberti <juberti@google.com>, Simon Perreault <simon.perreault@viagenie.ca>
Thread-Topic: [rtcweb] DSCP marking for STUN packets
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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I think we are discussing two different problems:
1. Determining what DSCP actually works over the Internet, by using ICE.
2. For one or more DSCPs an endpoint uses for a traffic flow, what DSCP ICE=
 needs to use.

In the case of #2, the endpoint uses those DSCPs because the network is pro=
visioned that way by the network admin or the service provider. This could =
possibly be achieved through a combination of some browser configuration an=
d JS input. For this case, using the highest DSCP value among those that ar=
e used for a given media or data-channel flow for ICE connectivity/consent =
checks makes sense to me.

On this other hand, #1 looks a bigger problem. In the absence of any inform=
ation, starting with BE or not setting anything in the packet looks a good =
option to me.

Muthu

|-----Original Message-----
|From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlu=
nd
|Sent: Wednesday, March 12, 2014 1:33 PM
|To: Justin Uberti; Simon Perreault
|Cc: rtcweb@ietf.org
|Subject: Re: [rtcweb] DSCP marking for STUN packets
|
|On 2014-03-12 04:53, Justin Uberti wrote:
|>
|>
|>
|> On Tue, Mar 11, 2014 at 7:02 AM, Simon Perreault
|> <simon.perreault@viagenie.ca <mailto:simon.perreault@viagenie.ca>> wrote=
:
|>
|>     Le 2014-03-10 16:57, James Polk a =E9crit :
|>     > At 12:16 AM 3/10/2014, Justin Uberti wrote:
|>     >> I think that the existing guidance should be sufficient in
|>     >> non-multiplexing cases (i.e. BUNDLE). For consent checks, I think=
 the
|>     >> same DSCP markings as any other ICE connectivity checks should be
|>     >> used; for ICE-for-SCTP checks, the same DSCP markings as media
|>     packets
|>     >> (i.e. the SCTP traffic) should be used.
|>     >>
|>     >> If multiplexing is being performed, the connectivity checks proba=
bly
|>     >> should use the highest DSCP value being used by the multiplexed
|>     media.
|>     >
|>     > why is (seemingly) everyone's default position "use the highest DS=
CP
|>     > available" for signaling packets?
|>     >
|>     > You just need to make sure the packets aren't starved or dropped b=
y/at
|>     > congestion points.
|>
|>     The underlying principle is that connectivity checks need to *check
|>     connectivity* (duh!). That's why you use the same DSCP as the media.
|>     Connectivity is affected by the DSCP. For example some DSCPs could b=
e
|>     filtered, or could be placed in a low-priority queue and get dropped
|>     such that connectivity fails.
|>
|>     >From this principle follows this conclusion: on a given 5-tuple, if
|>     your
|>     media uses different DSCPs, you need to check connectivity for all t=
hose
|>     DSCPs. That is, you would send multiple connectivity checks, one for
|>     each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we mi=
ght
|>     need an alternative heuristic: try the lowest-priority DSCP, and ass=
ume
|>     that if that one works, the higher-priority ones should work as well=
.
|>     (Note that Justin suggested the contrary.)
|>
|>     What you want to avoid is STUN packets getting through but not the
|>     corresponding media. That is an unworkable situation. The reverse is=
 not
|>     ideal either (you think you have no connectivity when in fact you do=
),
|>     but at least ICE can work around it (by picking a different candidat=
e).
|>
|>
|> We also need to consider the scenario of STUN consent checks, which will
|> need to get through reliably even when media marked with DSCP is already
|> flowing. In this case I think that using the highest DSCP value for STUN
|> is unavoidable.
|>
|
|(as individual)
|
|Sorry, I don't understand your reasoning here. Is it to provide the
|lowest possible drop probability, i.e. using the AFx class that has
|least drop probability?
|
|If there exist nodes that stomp on some DSCP markings, and not only
|remark them to Best Effort (BE), and instead drop then. Then, it is not
|obvious that that an AFx class is the most appropriate to use. In that
|case using BE might be more suitable for the consent freshness.
|
|I would also point out that highest DSCP value isn't clear. We are
|talking about forwarding behavior and I don't see how Expedited
|Forwarding (EF) can be compared with AF for example.
|
|I think we have to consider what our goals are here. To determine if we
|can get any packets through or determine if a particular 6-tuple (Source
|Address, Destination Address, Source Port, Destination Port, Protocol,
|and DSCP value) works? If it is the later then I think we need to have
|checks running on all the 6-tuples in use. This clearly have overhead
|concerns and helps worsening the ICE The Voice Hammer Attack (See
|section 18.5.1. of RFC 5245).
|
|If the goal is the first, isn't using BE a reasonable choice. Yes, it
|might have higher drop probability than AF, but it is likely to be
|robuster in general deployments and it determine if the peer is there or
|not. Dealing with broken 6-tuples needs to be a separate issue.
|
|Cheers
|
|Magnus Westerlund
|
|----------------------------------------------------------------------
|Services, Media and Network features, Ericsson Research EAB/TXM
|----------------------------------------------------------------------
|Ericsson AB                 | Phone  +46 10 7148287
|F=E4r=F6gatan 6                 | Mobile +46 73 0949079
|SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
|----------------------------------------------------------------------
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

>> It may be that we agree, but talk past each other :)
>>=20
>> My understanding of bundle-only has been that it is a mechanism to=20
>> ensure that the m- line is accepted only if the BUNDLE group is=20
>> accepted, e.g. in order to avoid having to reserve extra ports. BUT,=20
>> the bundle-only m- lines are NOT used to negotiate the actual BUNDLE=20
>> address - they will inherit the selected BUNDLE address, negotiated=20
>> using non-bundle-only m- lines.
>>=20
>
> I think we are agreeing. I think my main point is that the m=3D blocks th=
at contains bundle-only needs to contain the attribute a=3Drtcp-mux if that=
 is included in the first m=3D block for the bundle group that contains the=
 actual addresses.=20
> Similar with the a=3Drtcp attribute. Simply to be explicit that they are =
configured similarly.=20

Correct.

> This I think implies that we need a "to be replaced" value also for a=3Dr=
tcp in the bundle only m=3D blocks.

Yes.

Regards,

Christer


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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] Areas of security concern
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On 12 Mar 2014, at 05:11, Watson Ladd <watsonbladd@gmail.com> wrote:
> Dear all,
> I've jotted down the following notes when reading the drafts from here =
and the W3C as potential problem areas. Maybe there are things I've =
missed in the drafts that address them, but I think they are still worth =
thinking about. Some of these are more W3C, but we seem to be in charge =
of the security. These issues vary widely in seriousness. One of them is =
a demonstrated break of confidentiality, while several are open =
questions about how we communicate to users.
=85
> Problem 7: VBR privacy leaks
>=20
> I can do no better then to refer to the paper. =
http://www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/reports/yes-we-=
can.pdf Spoken phrases could be identified from encrypted data alone, =
using a Hidden Markov Model when the length of packets was preserved by =
the encryption.

The security considerations in draft-ietf-rtcweb-rtp-usage-12 refer to =
RFC 6562, which I believe gives recommendations to address this.

Colin



> Conclusion:
> Some of these issues result from well-known problems in older =
protocols. Others are longstanding difficult problems that have to be =
solved. Some are quite dangerous, while others will only permit the =
occasional prank. However, it is not enough to fix these specifically. =
Large areas of the proposal have not been explored by me. If I had to =
rate these by seriousness, problem 3 and 7 are the worst. They are =
luckily the easiest to fix.
>=20
> Sincerely,
> Watson Ladd
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
http://csperkins.org/




--Apple-Mail=_CB97D6F7-B3BE-4E0D-9437-4346E6DAD96D
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">On 12 =
Mar 2014, at 05:11, Watson Ladd &lt;<a =
href=3D"mailto:watsonbladd@gmail.com">watsonbladd@gmail.com</a>&gt; =
wrote:<div><blockquote type=3D"cite"><div dir=3D"ltr">Dear all,<br>I've =
jotted down the following notes when reading the drafts from here and =
the W3C as potential problem areas. Maybe there are things I've missed =
in the drafts that address them, but I think they are still worth =
thinking about. Some of these are more W3C, but we seem to be in charge =
of the security. These issues vary widely in seriousness. One of them is =
a demonstrated break of confidentiality, while several are open =
questions about how we communicate to =
users.</div></blockquote><div>=85</div><blockquote type=3D"cite"><div =
dir=3D"ltr"><div>Problem 7: VBR privacy leaks</div><div><br></div><div>I =
can do no better then to refer to the paper.&nbsp;<a =
href=3D"http://www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/reports=
/yes-we-can.pdf">http://www.infsec.cs.uni-saarland.de/teaching/WS08/Semina=
r/reports/yes-we-can.pdf</a> Spoken phrases could be identified from =
encrypted data alone, using a Hidden Markov Model when the length of =
packets was preserved by the =
encryption.</div></div></blockquote><div><br></div><div>The security =
considerations in&nbsp;draft-ietf-rtcweb-rtp-usage-12 refer to RFC 6562, =
which I believe gives recommendations to address =
this.</div><div><br></div><div>Colin</div><div><br></div><div><br></div><b=
r><blockquote type=3D"cite"><div =
dir=3D"ltr"><div>Conclusion:</div><div>Some of these issues result from =
well-known problems in older protocols. Others are longstanding =
difficult problems that have to be solved. Some are quite dangerous, =
while others will only permit the occasional prank. However, it is not =
enough to fix these specifically. Large areas of the proposal have not =
been explored by me. If I had to rate these by seriousness, problem 3 =
and 7 are the worst. They are luckily the easiest to fix.</div>
<div><br></div><div>Sincerely,</div><div>Watson Ladd</div></div>
_______________________________________________<br>rtcweb mailing =
list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br><div =
apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px;"><div><br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div><div><br></d=
iv></span><br class=3D"Apple-interchange-newline">

</div>
<br></body></html>=

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From: John Mattsson <john.mattsson@ericsson.com>
To: Watson Ladd <watsonbladd@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Areas of security concern
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Bjoern Hoehrmann <derhoermi@gmx.net>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Transports: RFC 6062 support
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Date: Wed, 12 Mar 2014 13:49:32 +0000
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References: <53183BBB.3040409@viagenie.ca> <CAOJ7v-1ZN=N5w3hFRkrC+LzarGFnt0qimcJbkTxWy6z0vWZFDg@mail.gmail.com> <531874A4.6010908@viagenie.ca> <CAOJ7v-3xasrFG5WVXPd_hA0=OoxvKbAhL2V4erYt7-kGt_JrtA@mail.gmail.com> <53187960.2010709@viagenie.ca> <531DDB5A.4060201@ericsson.com> <CAOW+2dtmLUWO-fKtNMMxn4ozywY2-dM3X0kEYwipRb9LcHgxQw@mail.gmail.com> <531F6B6E.6070609@alvestrand.no> <u5suh9p6srq201drs3r59fuagp5qbhbkdm@hive.bjoern.hoehrmann.de> <531F7812.4030505@alvestrand.no> <cfvuh9pbtdjo1lts32nlramj6436h7fv4g@hive.bjoern.hoehrmann.de>
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Subject: Re: [rtcweb] Transports: RFC 6062 support
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And your example is poor. "If XXX sends foo, then XXX MAY include a bar ele=
ment" and if necessary a corresponding receiving requirement would be more =
correct.

Given that the ABNF probably already specifies how the the element is inclu=
ded, in other standards bodies I would write "can" here as in possibility, =
rather than option. Many such requirements are dynamic, not static, as an o=
ption should really be.

How do you test what you have written.

Keith

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of=20
> Bjoern Hoehrmann
> Sent: 11 March 2014 21:23
> To: Harald Alvestrand
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Transports: RFC 6062 support
>=20
> * Harald Alvestrand wrote:
> >On 03/11/2014 09:26 PM, Bjoern Hoehrmann wrote:
> >> * Harald Alvestrand wrote:
> >>> In RFC 2119 theory, MAY and not mentioning it at all have exactly=20
> >>> the same weight.
> >> (That depends on whether the default is "allowed" or=20
> "prohibited" and=20
> >> there are plenty of specifications where explicit MAY=20
> overrides a de-=20
> >> fault of "prohibited".)
> >
> >Probably true. But I can't think of any at the moment - do you have=20
> >something specific in mind?
>=20
> (A typical example is when RFC 2119 keywords are used to=20
> describe how a protocol element is constructed, say, "<foo>=20
> elements MAY carry bar=3D''
> attributes"; you wouldn't assume you can use <foo bar=3D''>=20
> without that.)
> --
> Bj=F6rn H=F6hrmann =B7 mailto:bjoern@hoehrmann.de =B7=20
> http://bjoern.hoehrmann.de Am Badedeich 7 =B7 Telefon:=20
> +49(0)160/4415681 =B7 http://www.bjoernsworld.de
> 25899 Dageb=FCll =B7 PGP Pub. KeyID: 0xA4357E78 =B7=20
> http://www.websitedev.de/=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =


From nobody Wed Mar 12 08:18:18 2014
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From: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On Mar 8, 2014, at 6:48 AM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 7 March 2014 20:14, Dan Wing <dwing@cisco.com> wrote:
>> About half a year ago, AVTCORE published an updated recommendation =
for CNAME as RFC7022.  Is its guidance insufficient?
>=20
> Necessary, but not sufficient.  Unfortunately, the definition of a
> "session", while sufficient for the description in 7022, is not quite
> precise enough for this use case.  The implication there is that it
> means "RTP session", which is both not at the right level of
> granularity, and not directly actionable.

Perhaps AVTCORE can take on an update to use WebRTC terminology.

> I also note this little gem:
>=20
>   A longer-term persistent RTCP CNAME is sometimes useful to =
facilitate
>   third-party monitoring, consistent with [RFC3550].

Yep, it's true.

-d


From nobody Wed Mar 12 08:32:20 2014
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 2014-03-12 01:51, Dan Wing wrote:
> 
> On Mar 8, 2014, at 6:48 AM, Martin Thomson <martin.thomson@gmail.com> wrote:
> 
>> On 7 March 2014 20:14, Dan Wing <dwing@cisco.com> wrote:
>>> About half a year ago, AVTCORE published an updated recommendation for CNAME as RFC7022.  Is its guidance insufficient?
>>
>> Necessary, but not sufficient.  Unfortunately, the definition of a
>> "session", while sufficient for the description in 7022, is not quite
>> precise enough for this use case.  The implication there is that it
>> means "RTP session", which is both not at the right level of
>> granularity, and not directly actionable.
> 
> Perhaps AVTCORE can take on an update to use WebRTC terminology.

I disagree!

WebRTC is a particular usage of RTP/RTCP. The fact that RFC 7022's
terminology is not sufficiently narrow to describe the particulars of
WebRTC's usage is in my opinion not a bug. It is a feature. If the
AVTCORE document would be using the narrowed down case of WebRTC the
recommendations would no longer be applicable in a number of other
usages of RTP.

Deciding to apply RTP in a particular way to solve ones real-time
transport problems, also means that one have may have to be more
specific within that usage about things, this is such an example.

Cheers

Magnus Westerlund
(As Individual)

----------------------------------------------------------------------
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----------------------------------------------------------------------
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Färögatan 6                 | Mobile +46 73 0949079
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From: Justin Uberti <juberti@google.com>
Date: Wed, 12 Mar 2014 09:05:08 -0700
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On Wed, Mar 12, 2014 at 1:03 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-12 04:53, Justin Uberti wrote:
> >
> >
> >
> > On Tue, Mar 11, 2014 at 7:02 AM, Simon Perreault
> > <simon.perreault@viagenie.ca <mailto:simon.perreault@viagenie.ca>>
> wrote:
> >
> >     Le 2014-03-10 16:57, James Polk a =C3=A9crit :
> >     > At 12:16 AM 3/10/2014, Justin Uberti wrote:
> >     >> I think that the existing guidance should be sufficient in
> >     >> non-multiplexing cases (i.e. BUNDLE). For consent checks, I thin=
k
> the
> >     >> same DSCP markings as any other ICE connectivity checks should b=
e
> >     >> used; for ICE-for-SCTP checks, the same DSCP markings as media
> >     packets
> >     >> (i.e. the SCTP traffic) should be used.
> >     >>
> >     >> If multiplexing is being performed, the connectivity checks
> probably
> >     >> should use the highest DSCP value being used by the multiplexed
> >     media.
> >     >
> >     > why is (seemingly) everyone's default position "use the highest
> DSCP
> >     > available" for signaling packets?
> >     >
> >     > You just need to make sure the packets aren't starved or dropped
> by/at
> >     > congestion points.
> >
> >     The underlying principle is that connectivity checks need to *check
> >     connectivity* (duh!). That's why you use the same DSCP as the media=
.
> >     Connectivity is affected by the DSCP. For example some DSCPs could =
be
> >     filtered, or could be placed in a low-priority queue and get droppe=
d
> >     such that connectivity fails.
> >
> >     >From this principle follows this conclusion: on a given 5-tuple, i=
f
> >     your
> >     media uses different DSCPs, you need to check connectivity for all
> those
> >     DSCPs. That is, you would send multiple connectivity checks, one fo=
r
> >     each DSCPs in the set used on this 5-tuple. Yeah, it sucks, so we
> might
> >     need an alternative heuristic: try the lowest-priority DSCP, and
> assume
> >     that if that one works, the higher-priority ones should work as wel=
l.
> >     (Note that Justin suggested the contrary.)
> >
> >     What you want to avoid is STUN packets getting through but not the
> >     corresponding media. That is an unworkable situation. The reverse i=
s
> not
> >     ideal either (you think you have no connectivity when in fact you
> do),
> >     but at least ICE can work around it (by picking a different
> candidate).
> >
> >
> > We also need to consider the scenario of STUN consent checks, which wil=
l
> > need to get through reliably even when media marked with DSCP is alread=
y
> > flowing. In this case I think that using the highest DSCP value for STU=
N
> > is unavoidable.
> >
>
> (as individual)
>
> Sorry, I don't understand your reasoning here. Is it to provide the
> lowest possible drop probability, i.e. using the AFx class that has
> least drop probability?
>

The idea is to have the same drop probability as the most critical media
that is flowing (e.g. voice), so that we don't end up in a situation where
media flows fine, but the media crowds out STUN consent check packets,
leading to call failure.

>
> If there exist nodes that stomp on some DSCP markings, and not only
> remark them to Best Effort (BE), and instead drop then. Then, it is not
> obvious that that an AFx class is the most appropriate to use. In that
> case using BE might be more suitable for the consent freshness.
>

If nodes stomp on certain AF-marked packets, and this is widespread, this
essentially dooms use of DSCP at all, unless we want to build some
sophisticated detection and fallback mechanism. IOW, it's not clear to me
that marking packets as BE confers any value over not marking at all.

>
> I would also point out that highest DSCP value isn't clear. We are
> talking about forwarding behavior and I don't see how Expedited
> Forwarding (EF) can be compared with AF for example.
>
> I think we have to consider what our goals are here. To determine if we
> can get any packets through or determine if a particular 6-tuple (Source
> Address, Destination Address, Source Port, Destination Port, Protocol,
> and DSCP value) works? If it is the later then I think we need to have
> checks running on all the 6-tuples in use. This clearly have overhead
> concerns and helps worsening the ICE The Voice Hammer Attack (See
> section 18.5.1. of RFC 5245).
>
> If the goal is the first, isn't using BE a reasonable choice. Yes, it
> might have higher drop probability than AF, but it is likely to be
> robuster in general deployments and it determine if the peer is there or
> not. Dealing with broken 6-tuples needs to be a separate issue.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Mar 12, 2014 at 1:03 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 2014-03-12 04:53, Justin =
Uberti wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Tue, Mar 11, 2014 at 7:02 AM, Simon Perreault<br>
</div><div><div class=3D"h5">&gt; &lt;<a href=3D"mailto:simon.perreault@via=
genie.ca">simon.perreault@viagenie.ca</a> &lt;mailto:<a href=3D"mailto:simo=
n.perreault@viagenie.ca">simon.perreault@viagenie.ca</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Le 2014-03-10 16:57, James Polk a =C3=A9crit :<br>
&gt; =C2=A0 =C2=A0 &gt; At 12:16 AM 3/10/2014, Justin Uberti wrote:<br>
&gt; =C2=A0 =C2=A0 &gt;&gt; I think that the existing guidance should be su=
fficient in<br>
&gt; =C2=A0 =C2=A0 &gt;&gt; non-multiplexing cases (i.e. BUNDLE). For conse=
nt checks, I think the<br>
&gt; =C2=A0 =C2=A0 &gt;&gt; same DSCP markings as any other ICE connectivit=
y checks should be<br>
&gt; =C2=A0 =C2=A0 &gt;&gt; used; for ICE-for-SCTP checks, the same DSCP ma=
rkings as media<br>
&gt; =C2=A0 =C2=A0 packets<br>
&gt; =C2=A0 =C2=A0 &gt;&gt; (i.e. the SCTP traffic) should be used.<br>
&gt; =C2=A0 =C2=A0 &gt;&gt;<br>
&gt; =C2=A0 =C2=A0 &gt;&gt; If multiplexing is being performed, the connect=
ivity checks probably<br>
&gt; =C2=A0 =C2=A0 &gt;&gt; should use the highest DSCP value being used by=
 the multiplexed<br>
&gt; =C2=A0 =C2=A0 media.<br>
&gt; =C2=A0 =C2=A0 &gt;<br>
&gt; =C2=A0 =C2=A0 &gt; why is (seemingly) everyone&#39;s default position =
&quot;use the highest DSCP<br>
&gt; =C2=A0 =C2=A0 &gt; available&quot; for signaling packets?<br>
&gt; =C2=A0 =C2=A0 &gt;<br>
&gt; =C2=A0 =C2=A0 &gt; You just need to make sure the packets aren&#39;t s=
tarved or dropped by/at<br>
&gt; =C2=A0 =C2=A0 &gt; congestion points.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 The underlying principle is that connectivity checks nee=
d to *check<br>
&gt; =C2=A0 =C2=A0 connectivity* (duh!). That&#39;s why you use the same DS=
CP as the media.<br>
&gt; =C2=A0 =C2=A0 Connectivity is affected by the DSCP. For example some D=
SCPs could be<br>
&gt; =C2=A0 =C2=A0 filtered, or could be placed in a low-priority queue and=
 get dropped<br>
&gt; =C2=A0 =C2=A0 such that connectivity fails.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 &gt;From this principle follows this conclusion: on a gi=
ven 5-tuple, if<br>
&gt; =C2=A0 =C2=A0 your<br>
&gt; =C2=A0 =C2=A0 media uses different DSCPs, you need to check connectivi=
ty for all those<br>
&gt; =C2=A0 =C2=A0 DSCPs. That is, you would send multiple connectivity che=
cks, one for<br>
&gt; =C2=A0 =C2=A0 each DSCPs in the set used on this 5-tuple. Yeah, it suc=
ks, so we might<br>
&gt; =C2=A0 =C2=A0 need an alternative heuristic: try the lowest-priority D=
SCP, and assume<br>
&gt; =C2=A0 =C2=A0 that if that one works, the higher-priority ones should =
work as well.<br>
&gt; =C2=A0 =C2=A0 (Note that Justin suggested the contrary.)<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 What you want to avoid is STUN packets getting through b=
ut not the<br>
&gt; =C2=A0 =C2=A0 corresponding media. That is an unworkable situation. Th=
e reverse is not<br>
&gt; =C2=A0 =C2=A0 ideal either (you think you have no connectivity when in=
 fact you do),<br>
&gt; =C2=A0 =C2=A0 but at least ICE can work around it (by picking a differ=
ent candidate).<br>
&gt;<br>
&gt;<br>
&gt; We also need to consider the scenario of STUN consent checks, which wi=
ll<br>
&gt; need to get through reliably even when media marked with DSCP is alrea=
dy<br>
&gt; flowing. In this case I think that using the highest DSCP value for ST=
UN<br>
&gt; is unavoidable.<br>
&gt;<br>
<br>
</div></div>(as individual)<br>
<br>
Sorry, I don&#39;t understand your reasoning here. Is it to provide the<br>
lowest possible drop probability, i.e. using the AFx class that has<br>
least drop probability?<br></blockquote><div><br></div><div>The idea is to =
have the same drop probability as the most critical media that is flowing (=
e.g. voice), so that we don&#39;t end up in a situation where media flows f=
ine, but the media crowds out STUN consent check packets, leading to call f=
ailure.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
If there exist nodes that stomp on some DSCP markings, and not only<br>
remark them to Best Effort (BE), and instead drop then. Then, it is not<br>
obvious that that an AFx class is the most appropriate to use. In that<br>
case using BE might be more suitable for the consent freshness.<br></blockq=
uote><div><br></div><div>If nodes stomp on certain AF-marked packets, and t=
his is widespread, this essentially dooms use of DSCP at all, unless we wan=
t to build some sophisticated detection and fallback mechanism. IOW, it&#39=
;s not clear to me that marking packets as BE confers any value over not ma=
rking at all.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
I would also point out that highest DSCP value isn&#39;t clear. We are<br>
talking about forwarding behavior and I don&#39;t see how Expedited<br>
Forwarding (EF) can be compared with AF for example.<br>
<br>
I think we have to consider what our goals are here. To determine if we<br>
can get any packets through or determine if a particular 6-tuple (Source<br=
>
Address, Destination Address, Source Port, Destination Port, Protocol,<br>
and DSCP value) works? If it is the later then I think we need to have<br>
checks running on all the 6-tuples in use. This clearly have overhead<br>
concerns and helps worsening the ICE The Voice Hammer Attack (See<br>
section 18.5.1. of RFC 5245).<br>
<br>
If the goal is the first, isn&#39;t using BE a reasonable choice. Yes, it<b=
r>
might have higher drop probability than AF, but it is likely to be<br>
robuster in general deployments and it determine if the peer is there or<br=
>
not. Dealing with broken 6-tuples needs to be a separate issue.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Transports: RFC 6062 support
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On Wed, Mar 12, 2014 at 1:16 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-10 21:09, Christer Holmberg wrote:
> > Hi,
> >
> >>>>>>> Before I comment on your proposal, let's take a few steps
> >>>>>>> back (maybe it can even solve your issue).
> >>>>>>>
> >>>>>>> BUNDLE currently mandates the usage of both the SDP
> >>>>>>> rtcp-mux and rtcp attributes. I remember we had
> >>>>>>> discussions about that, but I don't remember the
> >>>>>>> justification at the moment. But, I wonder whether that
> >>>>>>> text is from before we made a decision that, unless both
> >>>>>>> endpoints support BUNDLE, no endpoint will use a single
> >>>>>>> address:port.
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>> So, the question is: do we really need the SDP rtcp
> >>>>>>> attribute?
> >>>>>>
> >>>>>> I think it is a question of how you want the fallback to
> >>>>>> work. In the case of bundle only lines, a non  bundle
> >>>>>> supporting peer will reject them, thus no issues with what
> >>>>>> is written around RTCP-mux and a=3Drtcp in those m=3D blocks.
> >>>>>> However, the first one if you want that to support fallback
> >>>>>> that makes sense then you will need both a=3Drtcp-mux and an
> >>>>>> a=3Drtcp (with a different port) to handle that with optimal
> >>>>>> backwards compatibility.
> >>>>>>
> >>>>>> Regarding the motivation for a=3Drtcp-mux and a=3Drtcp, I think
> >>>>>> it is reasonably straight forward. We want a=3Drtcp-mux to
> >>>>>> minimize port usage, i.e. ports=3D1. Not supporting
> >>>>>> a=3Drtcp-mux with bundle that would mean two ports (RTP and
> >>>>>> RTCP) for the bundle group. To get the backwards
> >>>>>> compatibility the usage of a=3Drtcp would be required.
> >>>>>
> >>>>> Keep in mind that the offerer must always be prepared that
> >>>>> the answerer does not support the rtcp-mux attribute (or the
> >>>>> rtcp attribute).
> >>>>
> >>>> Yes, that was what I meant with fallback in the above text.
> >>>> What you do in case the peer doesn't support bundle.
> >>>>
> >>>>> So, at least my understanding is: no matter whether BUNDLE is
> >>>>> used or not, if the answer does not contain a rtcp-mux (e.g.
> >>>>> in case of fallback) attribute, the offerer must use the
> >>>>> default "rtp+1" port.
> >>>>
> >>>> I have a tendency to agree, which means that the first m=3D block
> >>>> in a bunde-only group will be required to contain a=3Drtcp and if
> >>>> RTCP mux is desired also that attribute.
> >>>>
> >>>> I think a=3Drtcp could be skipped for a bound-only m=3D block due
> >>>> to the requirement on supporting a=3Drtcp-mux if you do bundle.
> >>>
> >>> For bundle-only, I actually think BOTH can be skipped, because
> >>> bundle-only will by default use whatever address:port is
> >>> negotiated - both for RTP and RTCP.
> >>
> >> No, I see a path of grief to not explicitly signal things when you
> >> use them.
> >
> > But, the whole idea of using port zero in bundle-only was that, if
> > the answerer supports BUNDLE, port zero will be replaced with the
> > negotiated address:port.
> >
> > When you send the offer, you don't yet know which address:port will
> > be selected, so there is no idea to insert a=3Drtcp. Inserting
> > a=3Drtcp-mux probably doesn't harm, though, so I won't argue about it
> > :)
>
> Well, if you can't insert the port, the same applies to a=3Drtcp as to th=
e
> port in the m=3D line.
>
> Because in the bundle-only case we are assuming and that assumption
> needs to apply also for a=3Drtcp. And for a bundle-only case I think the
> appropriate thing is to assume that a=3Drtcp-mux will be honored. To have
> it being honored you need explicit signal it. The a=3Drtcp can be
> considered redundant in this case, but I am sensitive to removing it as
> that would create a special case for a=3Dbundle-only lines. I think it is
> better to simply include it, even if it will say a=3Drtcp:0
>
> >
> >> I also thought the a=3Drtcp-mux is a MUST to implement, not a MUST
> >> use.
> >
> > Currently the draft says MUST use. But, that may be a mistake, or a
> > leftover from previous procedures. I agree that one should be able to
> > use BUNDLE also without rtcp-mux (i.e. using separate BUNDLE ports
> > for the RTP traffic and the RTCP traffic).
>
> I hope if people disagree that they shout now.
>

I think rtcp-mux has to be a MUST use when BUNDLE is used between the two
endpoints. (If the endpoint doesn't support BUNDLE, rtcp-mux is not
required.)

Otherwise, you get into weird edge cases, such as where you have a data
channel (no RTCP component), and then want to add audio (with RTP and RTCP
components). You could bundle the audio RTP over the existing data channel
transport, but it's not clear what you should do with the RTCP component.
Mandating rtcp-mux avoids this issue completely.

>
> >
> >>> But, for non-bundle-only, I still ask whether we really need
> >>> a=3Drtcp. Isn't a=3Drtcp-mux enough? If the answerer supports it, you
> >>> will use it, otherwise you will use rtp+1.
> >>
> >> My understanding is that a=3Drtcp has been required in any
> >> non-private network deployments due to >NATs for the last 10 years.
> >> Thus, I don't see how you can remove them and expect it to work,
> >> unless >you are doing a=3Drtcp-mux. Thus, my view would be that you
> >> can remove a=3Drtcp if you know that the >peer supprots a=3Drtcp-mux.
> >
> > Are you saying that we should use a=3Drtcp ONLY to indicate the RTP
> > port, in case the answerer does not support a=3Drtcp-mux?
>
> It has its normal usage, i.e. RTCP port in cases when a=3Drtcp-mux is not
> agreed on. In a bundle-only offer, I think the only reasonable approach
> is to handle it identically to the m=3D line port number.
>
> >
> >>> Otherwise, assume that the answerer supports BOTH a=3Drtcp and
> >>> a=3Drtcp-mux. Which has higher "priority"? How does the answerer
> >>> knows whether the offerer wants to use rtcp-mux, or whether it
> >>> wants to use whatever port is indicated in a=3Drtcp?
> >>
> >> That is a=3Drtcp-mux per RFC 5761 that has higher priority, please
> >> see Section 5.1.1.
> >
> > You are right - my mistake.
> >
> > But, never the less, if the offerer sends both a=3Drtcp-mux and a=3Drtc=
p,
> > unless a=3Drtcp points to the rtp port, the offerer needs to be
> > prepared to use 3 different ports for RTCP (rtp+1 port, rtp port and
> > rtp+<a=3Drtcp value>), because it does not know which attribute (if
> > any) the answerer supports, and which (if any) the answerer is
> > willing to use.
>
> This is actually not necessary, if one can get the rtp+1 port for RTCP,
> then include that in the a=3Drtcp line and everything is fine.
>
> I was under the impression that almost all SIP SDP O/A was using a=3Drtcp=
.
> But if that is a misunderstanding from my perspective, I still think the
> rules will hold. If you don't include the a=3Drtcp attribute, then the
> rtp+1 port is the equivalent of the a=3Drtcp value.
>
> From a bundle perspective, I think what needs to be considered is the
> actual or implied values on the m=3D blocks for the rtcp port when using
> bundle-only as well as the other case.
>
> Cheers
>
> Magnus Westerlund
> (As individual)
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Mar 12, 2014 at 1:16 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"HOEnZb"><div class=3D"h5">On 2=
014-03-10 21:09, Christer Holmberg wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Before I comment on your proposal, let&#39;s t=
ake a few steps<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; back (maybe it can even solve your issue).<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; BUNDLE currently mandates the usage of both th=
e SDP<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; rtcp-mux and rtcp attributes. I remember we ha=
d<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; discussions about that, but I don&#39;t rememb=
er the<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; justification at the moment. But, I wonder whe=
ther that<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; text is from before we made a decision that, u=
nless both<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; endpoints support BUNDLE, no endpoint will use=
 a single<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; address:port.<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; So, the question is: do we really need the SDP=
 rtcp<br>
&gt;&gt;&gt;&gt;&gt;&gt;&gt; attribute?<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; I think it is a question of how you want the fallb=
ack to<br>
&gt;&gt;&gt;&gt;&gt;&gt; work. In the case of bundle only lines, a non =C2=
=A0bundle<br>
&gt;&gt;&gt;&gt;&gt;&gt; supporting peer will reject them, thus no issues w=
ith what<br>
&gt;&gt;&gt;&gt;&gt;&gt; is written around RTCP-mux and a=3Drtcp in those m=
=3D blocks.<br>
&gt;&gt;&gt;&gt;&gt;&gt; However, the first one if you want that to support=
 fallback<br>
&gt;&gt;&gt;&gt;&gt;&gt; that makes sense then you will need both a=3Drtcp-=
mux and an<br>
&gt;&gt;&gt;&gt;&gt;&gt; a=3Drtcp (with a different port) to handle that wi=
th optimal<br>
&gt;&gt;&gt;&gt;&gt;&gt; backwards compatibility.<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; Regarding the motivation for a=3Drtcp-mux and a=3D=
rtcp, I think<br>
&gt;&gt;&gt;&gt;&gt;&gt; it is reasonably straight forward. We want a=3Drtc=
p-mux to<br>
&gt;&gt;&gt;&gt;&gt;&gt; minimize port usage, i.e. ports=3D1. Not supportin=
g<br>
&gt;&gt;&gt;&gt;&gt;&gt; a=3Drtcp-mux with bundle that would mean two ports=
 (RTP and<br>
&gt;&gt;&gt;&gt;&gt;&gt; RTCP) for the bundle group. To get the backwards<b=
r>
&gt;&gt;&gt;&gt;&gt;&gt; compatibility the usage of a=3Drtcp would be requi=
red.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Keep in mind that the offerer must always be prepared =
that<br>
&gt;&gt;&gt;&gt;&gt; the answerer does not support the rtcp-mux attribute (=
or the<br>
&gt;&gt;&gt;&gt;&gt; rtcp attribute).<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Yes, that was what I meant with fallback in the above text=
.<br>
&gt;&gt;&gt;&gt; What you do in case the peer doesn&#39;t support bundle.<b=
r>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; So, at least my understanding is: no matter whether BU=
NDLE is<br>
&gt;&gt;&gt;&gt;&gt; used or not, if the answer does not contain a rtcp-mux=
 (e.g.<br>
&gt;&gt;&gt;&gt;&gt; in case of fallback) attribute, the offerer must use t=
he<br>
&gt;&gt;&gt;&gt;&gt; default &quot;rtp+1&quot; port.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I have a tendency to agree, which means that the first m=
=3D block<br>
&gt;&gt;&gt;&gt; in a bunde-only group will be required to contain a=3Drtcp=
 and if<br>
&gt;&gt;&gt;&gt; RTCP mux is desired also that attribute.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I think a=3Drtcp could be skipped for a bound-only m=3D bl=
ock due<br>
&gt;&gt;&gt;&gt; to the requirement on supporting a=3Drtcp-mux if you do bu=
ndle.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; For bundle-only, I actually think BOTH can be skipped, because=
<br>
&gt;&gt;&gt; bundle-only will by default use whatever address:port is<br>
&gt;&gt;&gt; negotiated - both for RTP and RTCP.<br>
&gt;&gt;<br>
&gt;&gt; No, I see a path of grief to not explicitly signal things when you=
<br>
&gt;&gt; use them.<br>
&gt;<br>
&gt; But, the whole idea of using port zero in bundle-only was that, if<br>
&gt; the answerer supports BUNDLE, port zero will be replaced with the<br>
&gt; negotiated address:port.<br>
&gt;<br>
&gt; When you send the offer, you don&#39;t yet know which address:port wil=
l<br>
&gt; be selected, so there is no idea to insert a=3Drtcp. Inserting<br>
&gt; a=3Drtcp-mux probably doesn&#39;t harm, though, so I won&#39;t argue a=
bout it<br>
&gt; :)<br>
<br>
</div></div>Well, if you can&#39;t insert the port, the same applies to a=
=3Drtcp as to the<br>
port in the m=3D line.<br>
<br>
Because in the bundle-only case we are assuming and that assumption<br>
needs to apply also for a=3Drtcp. And for a bundle-only case I think the<br=
>
appropriate thing is to assume that a=3Drtcp-mux will be honored. To have<b=
r>
it being honored you need explicit signal it. The a=3Drtcp can be<br>
considered redundant in this case, but I am sensitive to removing it as<br>
that would create a special case for a=3Dbundle-only lines. I think it is<b=
r>
better to simply include it, even if it will say a=3Drtcp:0<br>
<div class=3D""><br>
&gt;<br>
&gt;&gt; I also thought the a=3Drtcp-mux is a MUST to implement, not a MUST=
<br>
&gt;&gt; use.<br>
&gt;<br>
&gt; Currently the draft says MUST use. But, that may be a mistake, or a<br=
>
&gt; leftover from previous procedures. I agree that one should be able to<=
br>
&gt; use BUNDLE also without rtcp-mux (i.e. using separate BUNDLE ports<br>
&gt; for the RTP traffic and the RTCP traffic).<br>
<br>
</div>I hope if people disagree that they shout now.<br></blockquote><div><=
br></div><div>I think rtcp-mux has to be a MUST use when BUNDLE is used bet=
ween the two endpoints. (If the endpoint doesn&#39;t support BUNDLE, rtcp-m=
ux is not required.)</div>

<div><br></div><div>Otherwise, you get into weird edge cases, such as where=
 you have a data channel (no RTCP component), and then want to add audio (w=
ith RTP and RTCP components). You could bundle the audio RTP over the exist=
ing data channel transport, but it&#39;s not clear what you should do with =
the RTCP component. Mandating rtcp-mux avoids this issue completely.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D""><br>
&gt;<br>
&gt;&gt;&gt; But, for non-bundle-only, I still ask whether we really need<b=
r>
&gt;&gt;&gt; a=3Drtcp. Isn&#39;t a=3Drtcp-mux enough? If the answerer suppo=
rts it, you<br>
&gt;&gt;&gt; will use it, otherwise you will use rtp+1.<br>
&gt;&gt;<br>
&gt;&gt; My understanding is that a=3Drtcp has been required in any<br>
&gt;&gt; non-private network deployments due to &gt;NATs for the last 10 ye=
ars.<br>
&gt;&gt; Thus, I don&#39;t see how you can remove them and expect it to wor=
k,<br>
&gt;&gt; unless &gt;you are doing a=3Drtcp-mux. Thus, my view would be that=
 you<br>
&gt;&gt; can remove a=3Drtcp if you know that the &gt;peer supprots a=3Drtc=
p-mux.<br>
&gt;<br>
&gt; Are you saying that we should use a=3Drtcp ONLY to indicate the RTP<br=
>
&gt; port, in case the answerer does not support a=3Drtcp-mux?<br>
<br>
</div>It has its normal usage, i.e. RTCP port in cases when a=3Drtcp-mux is=
 not<br>
agreed on. In a bundle-only offer, I think the only reasonable approach<br>
is to handle it identically to the m=3D line port number.<br>
<div class=3D""><br>
&gt;<br>
&gt;&gt;&gt; Otherwise, assume that the answerer supports BOTH a=3Drtcp and=
<br>
&gt;&gt;&gt; a=3Drtcp-mux. Which has higher &quot;priority&quot;? How does =
the answerer<br>
&gt;&gt;&gt; knows whether the offerer wants to use rtcp-mux, or whether it=
<br>
&gt;&gt;&gt; wants to use whatever port is indicated in a=3Drtcp?<br>
&gt;&gt;<br>
&gt;&gt; That is a=3Drtcp-mux per RFC 5761 that has higher priority, please=
<br>
&gt;&gt; see Section 5.1.1.<br>
&gt;<br>
&gt; You are right - my mistake.<br>
&gt;<br>
&gt; But, never the less, if the offerer sends both a=3Drtcp-mux and a=3Drt=
cp,<br>
&gt; unless a=3Drtcp points to the rtp port, the offerer needs to be<br>
&gt; prepared to use 3 different ports for RTCP (rtp+1 port, rtp port and<b=
r>
&gt; rtp+&lt;a=3Drtcp value&gt;), because it does not know which attribute =
(if<br>
&gt; any) the answerer supports, and which (if any) the answerer is<br>
&gt; willing to use.<br>
<br>
</div>This is actually not necessary, if one can get the rtp+1 port for RTC=
P,<br>
then include that in the a=3Drtcp line and everything is fine.<br>
<br>
I was under the impression that almost all SIP SDP O/A was using a=3Drtcp.<=
br>
But if that is a misunderstanding from my perspective, I still think the<br=
>
rules will hold. If you don&#39;t include the a=3Drtcp attribute, then the<=
br>
rtp+1 port is the equivalent of the a=3Drtcp value.<br>
<br>
>From a bundle perspective, I think what needs to be considered is the<br>
actual or implied values on the m=3D blocks for the rtcp port when using<br=
>
bundle-only as well as the other case.<br>
<br>
Cheers<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Magnus Westerlund<br>
(As individual)<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7=
148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+4=
6 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br></div></div>

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Subject: [rtcweb] Should we reference the pause/resume I-D?
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Hi,=0A=
=0A=
at the IETF last week there was consensus in the AVTEXT WG meeting to=0A=
adopt the pause/resume draft [1] as a WG draft.=0A=
=0A=
In rtcweb/webrtc we're have the situation that we're discussing so =0A=
called "doo-hickeys" as an API surface where the web app (amongst other =0A=
things) can pause and resume the sending of a track. This can=0A=
be signaled with the direction attribute and a SDP O/A exchange (and the =
=0A=
app pausing/resuming sending of a track would presumably lead to a =0A=
"negotiationneeded" event being fired).=0A=
=0A=
But I think we should in addition require the browser to signal it=0A=
according to one of the methods in [1] (e.g. TMMBN =3D 0), and also=0A=
understand that signaling (a browser receiving TMMBN =3D 0 must know that=
=0A=
the other end-point will pause sending).=0A=
=0A=
My argument is that we know that many dislike SDP in rtcweb, and a=0A=
likely development is that it will be removed in a later version. My=0A=
speculation is that signaling as outlined in [1] will then be used for=0A=
pause/resume. If we support this from the beginning earlier=0A=
implementations could more easily interop with those later versions.=0A=
=0A=
=0A=
Stefan=0A=
=0A=
[1]=0A=
https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.txt=
=0A=


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 12 Mar 2014 12:50:25 -0700
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Subject: Re: [rtcweb] Should we reference the pause/resume I-D?
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While I do like the pause/resume draft, having core RTCWEB WG documents
(such as RTP Usage) depend on it seems like a bit of a stretch. After all,
the document was only adopted last week, and it is a rare IETF WG document
that can go from a -00 WG draft to publication as an RFC in under a year.


On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=E5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> Hi,
>
> at the IETF last week there was consensus in the AVTEXT WG meeting to
> adopt the pause/resume draft [1] as a WG draft.
>
> In rtcweb/webrtc we're have the situation that we're discussing so
> called "doo-hickeys" as an API surface where the web app (amongst other
> things) can pause and resume the sending of a track. This can
> be signaled with the direction attribute and a SDP O/A exchange (and the
> app pausing/resuming sending of a track would presumably lead to a
> "negotiationneeded" event being fired).
>
> But I think we should in addition require the browser to signal it
> according to one of the methods in [1] (e.g. TMMBN =3D 0), and also
> understand that signaling (a browser receiving TMMBN =3D 0 must know that
> the other end-point will pause sending).
>
> My argument is that we know that many dislike SDP in rtcweb, and a
> likely development is that it will be removed in a later version. My
> speculation is that signaling as outlined in [1] will then be used for
> pause/resume. If we support this from the beginning earlier
> implementations could more easily interop with those later versions.
>
>
> Stefan
>
> [1]
> https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.txt
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">While I do like the pause/resume draft, having core RTCWEB=
 WG documents (such as RTP Usage) depend on it seems like a bit of a stretc=
h. After all, the document was only adopted last week, and it is a rare IET=
F WG document that can go from a -00 WG draft to publication as an RFC in u=
nder a year. <br>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed,=
 Mar 12, 2014 at 11:01 AM, Stefan H=E5kansson LK <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blank">stefan.lk=
.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
at the IETF last week there was consensus in the AVTEXT WG meeting to<br>
adopt the pause/resume draft [1] as a WG draft.<br>
<br>
In rtcweb/webrtc we&#39;re have the situation that we&#39;re discussing so<=
br>
called &quot;doo-hickeys&quot; as an API surface where the web app (amongst=
 other<br>
things) can pause and resume the sending of a track. This can<br>
be signaled with the direction attribute and a SDP O/A exchange (and the<br=
>
app pausing/resuming sending of a track would presumably lead to a<br>
&quot;negotiationneeded&quot; event being fired).<br>
<br>
But I think we should in addition require the browser to signal it<br>
according to one of the methods in [1] (e.g. TMMBN =3D 0), and also<br>
understand that signaling (a browser receiving TMMBN =3D 0 must know that<b=
r>
the other end-point will pause sending).<br>
<br>
My argument is that we know that many dislike SDP in rtcweb, and a<br>
likely development is that it will be removed in a later version. My<br>
speculation is that signaling as outlined in [1] will then be used for<br>
pause/resume. If we support this from the beginning earlier<br>
implementations could more easily interop with those later versions.<br>
<br>
<br>
Stefan<br>
<br>
[1]<br>
<a href=3D"https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pau=
se-05.txt" target=3D"_blank">https://tools.ietf.org/id/draft-westerlund-avt=
ext-rtp-stream-pause-05.txt</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--089e010d7efc04b77d04f46e2726--


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Subject: Re: [rtcweb] Should we reference the pause/resume I-D?
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Agreed. Aren't there also patent declarations against this doc from
multiple holders?

While SDP will likely be removed from the API in the future, the
replacement would be a app-specific message sent over websockets, which
seems like it would work just fine.


On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba <bernard.aboba@gmail.com>wr=
ote:

> While I do like the pause/resume draft, having core RTCWEB WG documents
> (such as RTP Usage) depend on it seems like a bit of a stretch. After all=
,
> the document was only adopted last week, and it is a rare IETF WG documen=
t
> that can go from a -00 WG draft to publication as an RFC in under a year.
>
>
> On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=C3=A5kansson LK <
> stefan.lk.hakansson@ericsson.com> wrote:
>
>> Hi,
>>
>> at the IETF last week there was consensus in the AVTEXT WG meeting to
>> adopt the pause/resume draft [1] as a WG draft.
>>
>> In rtcweb/webrtc we're have the situation that we're discussing so
>> called "doo-hickeys" as an API surface where the web app (amongst other
>> things) can pause and resume the sending of a track. This can
>> be signaled with the direction attribute and a SDP O/A exchange (and the
>> app pausing/resuming sending of a track would presumably lead to a
>> "negotiationneeded" event being fired).
>>
>> But I think we should in addition require the browser to signal it
>> according to one of the methods in [1] (e.g. TMMBN =3D 0), and also
>> understand that signaling (a browser receiving TMMBN =3D 0 must know tha=
t
>> the other end-point will pause sending).
>>
>> My argument is that we know that many dislike SDP in rtcweb, and a
>> likely development is that it will be removed in a later version. My
>> speculation is that signaling as outlined in [1] will then be used for
>> pause/resume. If we support this from the beginning earlier
>> implementations could more easily interop with those later versions.
>>
>>
>> Stefan
>>
>> [1]
>> https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.tx=
t
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Agreed. Aren&#39;t there also patent declarations against =
this doc from multiple holders?<div><br></div><div>While SDP will likely be=
 removed from the API in the future, the replacement would be a app-specifi=
c message sent over websockets, which seems like it would work just fine.</=
div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed,=
 Mar 12, 2014 at 12:50 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a=
>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">While I do like the pause/r=
esume draft, having core RTCWEB WG documents (such as RTP Usage) depend on =
it seems like a bit of a stretch. After all, the document was only adopted =
last week, and it is a rare IETF WG document that can go from a -00 WG draf=
t to publication as an RFC in under a year. <br>



</div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><b=
r><br><div class=3D"gmail_quote">On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=
=C3=A5kansson LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansso=
n@ericsson.com" target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;<=
/span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
at the IETF last week there was consensus in the AVTEXT WG meeting to<br>
adopt the pause/resume draft [1] as a WG draft.<br>
<br>
In rtcweb/webrtc we&#39;re have the situation that we&#39;re discussing so<=
br>
called &quot;doo-hickeys&quot; as an API surface where the web app (amongst=
 other<br>
things) can pause and resume the sending of a track. This can<br>
be signaled with the direction attribute and a SDP O/A exchange (and the<br=
>
app pausing/resuming sending of a track would presumably lead to a<br>
&quot;negotiationneeded&quot; event being fired).<br>
<br>
But I think we should in addition require the browser to signal it<br>
according to one of the methods in [1] (e.g. TMMBN =3D 0), and also<br>
understand that signaling (a browser receiving TMMBN =3D 0 must know that<b=
r>
the other end-point will pause sending).<br>
<br>
My argument is that we know that many dislike SDP in rtcweb, and a<br>
likely development is that it will be removed in a later version. My<br>
speculation is that signaling as outlined in [1] will then be used for<br>
pause/resume. If we support this from the beginning earlier<br>
implementations could more easily interop with those later versions.<br>
<br>
<br>
Stefan<br>
<br>
[1]<br>
<a href=3D"https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pau=
se-05.txt" target=3D"_blank">https://tools.ietf.org/id/draft-westerlund-avt=
ext-rtp-stream-pause-05.txt</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--047d7b66f5fb153fd804f46f121f--


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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On 3/10/14 4:09 PM, Christer Holmberg wrote:

>> I also thought the a=rtcp-mux is a MUST to implement, not a MUST use.
>
> Currently the draft says MUST use. But, that may be a mistake, or a leftover from previous procedures. I agree that one should be able to use BUNDLE also without rtcp-mux (i.e. using separate BUNDLE ports for the RTP traffic and the RTCP traffic).

Now that I look carefully, I think you have been a little over zealous 
about this.

Remember, at least in principle, there can be non-RTP m-lines in the 
bundle. You don't want to require using rtcp-mux with those. (And, at 
least in principle, bundle can be used even when there are *no* RTP 
m-lines.)

So I think the draft must at least limit these requirements to RTP 
m-lines. (But finding a clean way to do that may be tricky, since there 
is a potentially open ended set of <proto> values that relate to RTP.)

	Thanks,
	Paul


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Hi Harald,

=20

The answer to your question 26 of February:
> Karl, what on Earth does TRAM milestone 3 have to do with whatever it =
is
you're proposing?
TRAM milestone 3 is: Sep 2014 - Send new proposed standard TURN server
discovery mechanism for enterprises and ISPs to IESG
> There's no (zero, nada, none) mention of QoS within that charter.



--- I didn=92t mention =93QoS=94 (*I want the QoS methods out!*), in my
protest/request=20

http://www.ietf.org/mail-archive/web/tram/current/msg00304.html =93It =
must be
a clear objective of the TRAM WG that ISPs/NSPs are allowed and =
encouraged
to route quality demanding WebRTC media into their IP pipes that are =
capable
of transporting real-time traffic without quality issues, using TURN
servers.=20

=20

The subject from the TRAM mailing initiation spells out:

*Milestone 3: TURN server auto-discovery mechanism for enterprise and =
ISPs*

The TRAM WG must assure that TURN servers offered by ISPs/NSPs can be =
used
for the above purpose.

(The objective of the TRAM WG milestone 3 is *not only* to resolve
restrictive enterprise NAT/firewall traversal problem using TURN =
servers.)=94

=20

However, it is a way of "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff, which =
I
now have started under=20

 <http://www.ietf.org/mail-archive/web/tram/current/msg00331.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html

The way to "Interfacing to QoS", A level 3-5 IP/IETF/WebRTC-thing how to
interface to lower level's QoS-stuff

=20

You were September 20 actually the first to answer my request for a DHCP
provided turn server address (which I withdraw in favor of the anycast
method, which is highly preferable):

http://www.ietf.org/mail-archive/web/rtcweb/current/msg08920.html=20

=20

There I said:

>There are several reasons for a network service provider to supply a =
TURN

server as part of his offered access:

- to keep media paths short, specifically not sending media outside its =
own

network to some distant application provided TURN server

- to support mobility, i.e. you may want to move from a LAN with a

configured TURN server to accessing via WiFi or 3G/4G OTT channels

- to offer a media path with better quality (than best effort data =
traffic).

Getting a =93WebRTC-ready=94 access and we can look forward to =
telepresence for

everyone.

=20

I don=92t really see anyone objecting to those objectives, and I don=92t =
think
you do either.

=20

The Anycast method were suggested October 22 (without further =
discussions)

http://www.ietf.org/mail-archive/web/rtcweb/current/msg09128.html=20

and thereafter brought into TRAM WG list February 8=20

http://www.ietf.org/mail-archive/web/tram/current/msg00132.html=20

=20

It took me some time to find the relevant emails from the discussions in
September and October which ended up with my entry to the [tram]-list
February 8, which is copied in full below, where I still say it was 95%
ready (There are a few relevant points from Simon that I will address
later). Thereafter, the QoS diversion and confusion started around =
February
12th in what looks to me as effort to make it a =93Mission =
Impossible=94.

=20

I don=92t know how the TRAM WG was created, but I thought Milestrone 3 =
was a
result of Cullen=92s. See next email in response to Cullen=92s questions =
and
requests.

=20

/Karl

=20

Footnote:

[1] I think both you (Harald) and Cullen, are good, have the best =
intentions
and integrity, but no one of us have infinite memory capacity =
(especially
when the VP8/H.264 flooding came in between). I only learned =93by =
accident=94
that the TRAM WG was in progress at the November WebRTC conference=94 =
and
should handle the Auto Discovery of TURN-servers.

=20

=20

Fr=E5n: Karl Stahl [mailto:karl.stahl@intertex.se]=20
Skickat: den 8 februari 2014 14:11
Till: 'tram@ietf.org'; 'tireddy@icisco.com'; 'Simon Perreault'
=C4mne: [tram] Milestone 3: TURN server auto-discovery mechanism for
enterprise and ISPs

=20

Karl from Ingate and Intertex just added himself to this list for =
exactly
the purpose of the subject of Milestone 3 above=20

=20

I welcome this initiative that I saw yesterday after being reminded by =
China
Mobile Research Institute who had seen the previous October discussion =
on
the WebRTC-list, parts of which are inserted below. Large carriers have =
also
expressed the importance of this milestone.

--------------------

We are working on this draft, will publish it next week.

-Tiru.

=20

Great, please consider the input and suggestions below:

=20

> -----Original Message-----

> From: tram [ <mailto:tram-bounces> mailto:tram-bounces at ietf.org] On
Behalf Of Simon Perreault

> Sent: Friday, February 07, 2014 7:58 PM

> To: tram at ietf.org

> Subject: [tram] Milestone 3: TURN server auto-discovery mechanism for

> enterprise and ISPs

>=20

=85

> Enterprises or ISPs wishing to provide

> their own TURN server, in an attempt to reduce so-called "triangle
routing",

> need a new auto-discovery mechanism.

--- There are more reasons heard for auto-discovery of a network =
provided
TURN server:

=20

- NSPs (Network Service Providers) want to provide a path where the
bandwidth of WebRTC is better coped with.

- NSPs or Enterprises want to offer an Internet access quality pipe for
prioritized RTC (Real Time Communication) traffic.=20

- Enterprises having restrictive firewalls, want to provide a UDP-path =
for
WebRTC and possibly also for better quality where RTC do not compete =
with
data traffic.=20

- Mobility; It is common to move from a LAN to accessing via WiFi or =
3G/4G
OTT channels, all should be able to automatically offer their own =
optimal
TURN server

=20

- Note that to achieve some of the above points, TURN must be favored =
over
STUN to enforce that the TURN-path actually is used. (The Anycast method
suggested below, =93automatically=94 does this.)=20

=20

-- In my view, a TURN server is the network providers=92 responsibility =
to
provide (just like the IP address, the default gateway, the DNS etc.) =
=96
rather than the application provider=92s. Our current Internet accesses =
may
not cope with or be optimized for WebRTC usage =96 The NSP (or =
Enterprise LAN
administrator) should then be able to fix the access (here by offering a
TURN server).=20

=20

> IMHO this is also a top-priority milestone. We need to quickly have a

> mechanism that people can implement in WebRTC browsers now, while

> there is still frenetic development happening.=20

--- Agree

I don't foresee actual server-

> side deployment happening quickly though. But as soon as clients are
ready,

> any ISP or enterprise can deploy and immediately benefit.

- There is a definitive interest among forward SPs already, especially
carriers owning the network.

(They have learned that their networks (especially mobile) seem to =
become
data crowded no matter how much bandwidth they invest in.)

=20

> I imagine that the solution will be fairly simple, spec- and
implementation-

> wise.=20

--- Agree that such solution should and could be found, but it is not
obvious. Below you find 3 proposals (initiated by me or colleagues, =
where I
see flaws in the two first and now only suggest the third (the Anycast
mechanism). From below:

=20

- 1st: =85 withdraw my suggestion to use DHCP to find a network provider
offered TURN server in favor of the DNS-Based Service Discovery method
(inserted last below). The major problem with the DHCP usage was that =
you
then also have to do something similar for: RA - Router Advertisement - =
in
IPv6, addition to the IPCP protocol for PPPoE and something for the =
mobile
OTT channel =96 wherever DHCP is NOT the method to give you an IP =
address.
(There were also concerns whether OSs actually supports forwards such
extended DHCP information for the browser to use.)

=20

- 2nd Reverse DNS-Based Service Discovery method:

Justin Uberti pointed out:  How does this get bootstrapped? That is, how =
is
the STUN server found?

[Karl] Oops, that got lost when leaving the DHCP track, and is a problem
when using DNS discovery.

(There were also hesitations of having to provision this special reverse =
DNS
for every access.)

=20

- 3rd The Anycast method below =96 I see no problem

It also has the advantage of encouraging (but not requiring) the =
STUN/TURN
to be built in the default gateway or NAT/firewall/access router itself,
with a second interface to a public IP address on the WAN side. (Current
volume deployed, low cost NSP triple play modems usually have a quality
assured level 2 or level 3 WAN pipe for just voice (and another for =
IPTV) =96
The anycast discovered TURN-server can be the access gateway to such =
quality
pipe for WebRTC media, in a single NSP provided CPE, scaling from
residential and up.)

=20

So I think we can finish this very quickly once we have a candidate

> draft. We just need authors. So I would target not long after Toronto.

>=20

> Simon

=20

WEB BROWSER BEHAVIOUR:

=20

Network provided TURN servers will not appear over night, applications =
may
for long provide a TURN server address, and there are exceptions where =
the
TURN server address is preferred to be =93manually=94 configured. It is
previously suggested, and to some extent discussed, that the WebRTC =
browser
should select the TURN server to use in the following priority order, =
where
ICE would assure that you get some connectivity if several candidates =
are
found and needs to be tested:

=20

1) TURN server address configured in the browser by the user (special =
cases,
normally not used, but handy for testing)

2) TURN server address configured by the network administrator via an =
=93admin
policy template=94 or a WPAD method as mentioned below

3) TURN server address auto-discovered by the mechanism discussed here

4) TURN server address being supplied by the web application

=20

With a good step 3), step 2) becomes obsolete since the network
administrator e.g. simply can set a route in the enterprise firewall to =
use
the Anycast mechanism instead.=20

=20

Even if the need for auto-discovery of the TURN server comes from WebRTC
usage, a general mechanism that also can be used by SIP clients (and =
other
protocols using TURN via ICE) is preferable. The anycast method has no
application dependence, but e.g. WPAD has (a SIP Client typically does =
not
have the luxury of a JS engine=85)

=20

/Karl

=20

Fr=E5n: tram [mailto:tram-bounces@ietf.org] F=F6r Harald Alvestrand
Skickat: den 26 februari 2014 04:49
Till: Colin Perkins; Karl Stahl
Kopia: Magnus Westerlund; rtcweb@ietf.org; tram@ietf.org
=C4mne: Re: [tram] [rtcweb] Payload Types assignments

=20

Karl,

what on Earth does TRAM milestone 3 have to do with whatever it is =
you're
proposing?

TRAM milestone 3 is:=20

Sep 2014 - Send new proposed standard TURN server discovery mechanism =
for
enterprises and ISPs to IESG

There's no (zero, nada, none) mention of QoS within that charter.




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link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 Harald,<o:p></o:p></span></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'><o:p>&nbsp;</o:p></span></b></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>T=
he answer to your question 26 of February:<br></span><span =
lang=3DEN-US>&gt; Karl, what on Earth does TRAM milestone 3 have to do =
with whatever it is you're proposing?<br>TRAM milestone 3 is: Sep 2014 - =
Send new proposed standard TURN server discovery mechanism for =
enterprises and ISPs to IESG<br>&gt; There's no (zero, nada, none) =
mention of QoS within that charter.<br><br></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- I didn&#8217;t mention &#8220;QoS&#8221; <b>(*I want the QoS methods =
out!*)</b>, in my protest/request</span><span lang=3DEN-US> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00304.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00304.html</a> =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:windowte=
xt'>&#8220;It must be a clear objective of the TRAM WG that ISPs/NSPs =
are allowed and encouraged to route quality demanding WebRTC media into =
their IP pipes that are capable of transporting real-time traffic =
without quality issues, using TURN servers. <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:windowte=
xt'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:windowte=
xt'>The subject from the TRAM mailing initiation spells =
out:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:windowte=
xt'>*<b>Milestone 3: TURN server auto-discovery mechanism for enterprise =
and ISPs</b>*<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:windowte=
xt'>The TRAM WG must assure that TURN servers offered by ISPs/NSPs can =
be used for the above purpose.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:windowte=
xt'>(The objective of the TRAM WG milestone 3 is *<b>not only</b>* to =
resolve restrictive enterprise NAT/firewall traversal problem using TURN =
servers.)&#8221;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ho=
wever, it is a way of </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>&quot;Interf=
acing to QoS&quot;, A level 3-5 IP/IETF/WebRTC-thing how to interface to =
lower level's QoS-stuff, </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>wh=
ich I now have started under </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00331.html">=
<span =
lang=3DEN-US>http://www.ietf.org/mail-archive/web/tram/current/msg00331.h=
tml</span></a></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>The way to =
&quot;Interfacing to QoS&quot;, A level 3-5 IP/IETF/WebRTC-thing how to =
interface to lower level's QoS-stuff<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Yo=
u were September 20 actually the first to answer my request for a DHCP =
provided turn server address (which I withdraw in favor of the anycast =
method, which is highly preferable):<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg08920.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg08920.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
ere I said:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>&gt;There are several reasons for a =
network service provider to supply a TURN<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>server as part of his offered =
access:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>- to keep media paths short, specifically =
not sending media outside its own<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>network to some distant application =
provided TURN server<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>- to support mobility, i.e. you may want =
to move from a LAN with a<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>configured TURN server to accessing via =
WiFi or 3G/4G OTT channels<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>- to offer a media path with better =
quality (than best effort data traffic).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>Getting a &#8220;WebRTC-ready&#8221; =
access and we can look forward to telepresence =
for<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'>everyone.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
don&#8217;t really see anyone objecting to those objectives, and I =
don&#8217;t think you do either.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e Anycast method were suggested October 22 (without further =
discussions)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09128.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09128.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>an=
d thereafter brought into TRAM WG list February 8 =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00132.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00132.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>It=
 took me some time to find the relevant emails from the discussions in =
September and October which ended up with my entry to the [tram]-list =
February 8, which is copied in full below, where I still say it was 95% =
ready (There are a few relevant points from Simon that I will address =
later). Thereafter, the QoS diversion and confusion started around =
February 12th in what looks to me as effort to make it a &#8220;Mission =
Impossible&#8221;.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
don&#8217;t know how the TRAM WG was created, but I thought Milestrone 3 =
was a result of Cullen&#8217;s. See next email in response to =
Cullen&#8217;s questions and requests.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
otnote:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[1=
] I think both you (Harald) and Cullen, are good, have the best =
intentions and integrity, but no one of us have infinite memory capacity =
(especially when the VP8/H.264 flooding came in between). I only learned =
&#8220;by accident&#8221; that the TRAM WG was in progress at the =
November WebRTC conference&#8221; and should handle the </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Au=
to Discovery of TURN-servers.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Karl Stahl =
[mailto:karl.stahl@intertex.se] <br><b>Skickat:</b> den 8 februari 2014 =
14:11<br><b>Till:</b> 'tram@ietf.org'; 'tireddy@icisco.com'; 'Simon =
Perreault'<br><b>=C4mne:</b> [tram] Milestone 3: TURN server =
auto-discovery mechanism for enterprise and ISPs<o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ka=
rl from Ingate and Intertex just added himself to this list for exactly =
the purpose of the subject of Milestone 3 above <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
welcome this initiative that I saw yesterday after being reminded by =
China Mobile Research Institute who had seen the previous October =
discussion on the WebRTC-list, parts of which are inserted below. Large =
carriers have also expressed the importance of this =
milestone.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
------------------<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>We are working on this draft, will publish it next =
week.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>-Tiru.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:blue'>Great, please =
consider the input and suggestions below:</span><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>&gt; -----Original Message-----<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; From: =
tram [</span><span style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'><a href=3D"mailto:tram-bounces"><span =
lang=3DEN-US>mailto:tram-bounces</span></a></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'> at =
ietf.org] On Behalf Of Simon Perreault<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; Sent: =
Friday, February 07, 2014 7:58 PM<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; To: =
tram at ietf.org<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>&gt; Subject: [tram] Milestone 3: TURN server =
auto-discovery mechanism for<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; =
enterprise and ISPs<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>&gt; <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>&#8230;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>&gt; Enterprises or ISPs wishing to =
provide<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; their =
own TURN server, in an attempt to reduce so-called &quot;triangle =
routing&quot;,<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>&gt; need a new auto-discovery =
mechanism.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- There are more reasons heard for auto-discovery of a network provided =
TURN server:<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
NSPs (Network Service Providers) want to provide a path where the =
bandwidth of WebRTC is better coped with.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
NSPs or Enterprises want to offer an Internet access quality pipe for =
prioritized RTC (Real Time Communication) traffic. =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
Enterprises having restrictive firewalls, want to provide a UDP-path for =
WebRTC and possibly also for better quality where RTC do not compete =
with data traffic. <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
Mobility; It is common to move from a LAN to accessing via WiFi or 3G/4G =
OTT channels, all should be able to automatically offer their own =
optimal TURN server<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
Note that to achieve some of the above points, TURN must be favored over =
STUN to enforce that the TURN-path actually is used. (The Anycast method =
suggested below, &#8220;automatically&#8221; does this.) =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
 In my view, a TURN server is the network providers&#8217; =
responsibility to provide (just like the IP address, the default =
gateway, the DNS etc.) &#8211; rather than the application =
provider&#8217;s. Our current Internet accesses may not cope with or be =
optimized for WebRTC usage &#8211; The NSP (or Enterprise LAN =
administrator) should then be able to fix the access (here by offering a =
TURN server). <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; IMHO =
this is also a top-priority milestone. We need to quickly have =
a<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; =
mechanism that people can implement in WebRTC browsers now, =
while<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; there =
is still frenetic development happening. <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- Agree</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>I don't foresee actual server-<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; side =
deployment happening quickly though. But as soon as clients are =
ready,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; any =
ISP or enterprise can deploy and immediately =
benefit.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
There is a definitive interest among forward SPs already, especially =
carriers owning the network.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(T=
hey have learned that their networks (especially mobile) seem to become =
data crowded no matter how much bandwidth they invest in.)</span><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'>&gt; I imagine that the solution will be fairly simple, =
spec- and implementation-<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; wise. =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- Agree that such solution should and could be found, but it is not =
obvious. Below you find 3 proposals (initiated by me or colleagues, =
where I see flaws in the two first and now only suggest the third (the =
Anycast mechanism). From below:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
1<sup>st</sup>: &#8230; withdraw my suggestion to use DHCP to find a =
network provider offered TURN server in favor of the DNS-Based Service =
Discovery method (inserted last below). The major problem with the DHCP =
usage was that you then also have to do something similar for: RA - =
Router Advertisement - in IPv6, addition to the IPCP protocol for PPPoE =
and something for the mobile OTT channel &#8211; wherever DHCP is NOT =
the method to give you an IP address. (There were also concerns whether =
OSs actually supports forwards such extended DHCP information for the =
browser to use.)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
2<sup>nd</sup> Reverse DNS-Based Service Discovery =
method:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ju=
stin Uberti pointed out: &nbsp;</span><span lang=3DEN-US>How does this =
get bootstrapped? That is, how is the STUN server found?</span><span =
lang=3DEN-US style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:windowtext'><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[K=
arl] Oops, that got lost when leaving the DHCP track, and is a problem =
when using DNS discovery.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(T=
here were also hesitations of having to provision this special reverse =
DNS for every access.)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
3<sup>rd</sup> The Anycast method below &#8211; I see no =
problem<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>It=
 also has the advantage of encouraging (but not requiring) the STUN/TURN =
to be built in the default gateway or NAT/firewall/access router itself, =
with a second interface to a public IP address on the WAN side. (Current =
volume deployed, low cost NSP triple play modems usually have a quality =
assured level 2 or level 3 WAN pipe for just voice (and another for =
IPTV) &#8211; The anycast discovered TURN-server can be the access =
gateway to such quality pipe for WebRTC media, in a single NSP provided =
CPE, scaling from residential and up.)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>So I think =
we can finish this very quickly once we have a =
candidate<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; draft. =
We just need authors. So I would target not long after =
Toronto.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'>&gt; =
Simon<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>WE=
B BROWSER BEHAVIOUR:<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ne=
twork provided TURN servers will not appear over night, applications may =
for long provide a TURN server address, and there are exceptions where =
the TURN server address is preferred to be &#8220;manually&#8221; =
configured. It is previously suggested, and to some extent discussed, =
that the WebRTC browser should select the TURN server to use in the =
following priority order, where ICE would assure that you get some =
connectivity if several candidates are found and needs to be =
tested:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>1)=
 TURN server address configured in the browser by the user (special =
cases, normally not used, but handy for testing)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>2)=
 TURN server address configured by the network administrator via an =
&#8220;admin policy template&#8221; or a WPAD method as mentioned =
below<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>3)=
 TURN server address auto-discovered by the mechanism discussed =
here<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>4)=
 TURN server address being supplied by the web =
application<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th a good step 3), step 2) becomes obsolete since the network =
administrator e.g. simply can set a route in the enterprise firewall to =
use the Anycast mechanism instead. <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ev=
en if the need for auto-discovery of the TURN server comes from WebRTC =
usage, a general mechanism that also can be used by SIP clients (and =
other protocols using TURN via ICE) is preferable. The anycast method =
has no application dependence, but e.g. WPAD has (a SIP Client typically =
does not have the luxury of a JS engine&#8230;)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'>Fr=E5n:</span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'> tram [mailto:tram-bounces@ietf.org] <b>F=F6r </b>Harald =
Alvestrand<br><b>Skickat:</b> den 26 februari 2014 04:49<br><b>Till:</b> =
Colin Perkins; Karl Stahl<br><b>Kopia:</b> Magnus Westerlund; =
rtcweb@ietf.org; tram@ietf.org<br><b>=C4mne:</b> Re: [tram] [rtcweb] =
Payload Types assignments<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>Karl,<br><br>what on Earth does TRAM milestone 3 have =
to do with whatever it is you're proposing?<br><br>TRAM milestone 3 is: =
<br><br>Sep 2014 - Send new proposed standard TURN server discovery =
mechanism for enterprises and ISPs to IESG<br><br>There's no (zero, =
nada, none) mention of QoS within that =
charter.<br><br><o:p></o:p></p></div></div></body></html>
------=_NextPart_000_021F_01CF3E96.89173610--



From nobody Thu Mar 13 00:38:46 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
Thread-Index: AQHPPCh4uGvaKjLTukeJMcClx2WsG5rZ+rmggABOAICAABMwcP//9NqAgAASK5D///t0AIAAX2mwgANKxACAAJwfoA==
Date: Thu, 13 Mar 2014 07:38:34 +0000
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/GDm3jQHzh3r4MQr_DdFQA--XZTM
Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

>>> I also thought the a=3Drtcp-mux is a MUST to implement, not a MUST use.
>>
>> Currently the draft says MUST use. But, that may be a mistake, or a left=
over from previous procedures. I agree that one should be=20
>> able to use BUNDLE also without rtcp-mux (i.e. using separate BUNDLE por=
ts for the RTP traffic and the RTCP traffic).
>
> Now that I look carefully, I think you have been a little over zealous ab=
out this.
>
> Remember, at least in principle, there can be non-RTP m-lines in the bund=
le. You don't want to require using rtcp-mux with=20
> those. (And, at least in principle, bundle can be used even when there ar=
e *no* RTP
> m-lines.)
>
> So I think the draft must at least limit these requirements to RTP m-line=
s. (But finding a clean way to do that may be tricky, since there is a pote=
ntially open ended set of <proto> values that relate to RTP.)

I think it is enough if we limit it to protocols for which the rtcp-mux att=
ribute has been defined. I don't think it's up to BUNDLE to generally defin=
e for which type of m- lines rtcp-mux can be used, only the (if any) BUNDLE=
 specific aspects.

Regards,

Christer


From nobody Thu Mar 13 00:57:01 2014
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'IESG IESG'" <iesg@ietf.org>
References: <20140214030712.30321.21888.idtracker@ietfa.amsl.com> <CAKhHsXGzA=ZTFGTK7ht9hQbfG70iqKrDtxrZCdQNNMzBYZCk8A@mail.gmail.com> <530604ea.c5bf440a.5cfd.ffffde18SMTPIN_ADDED_BROKEN@mx.google.com> <CAKhHsXGsp6ma6Ko9op+YFRqSGM_Ex-jFo_fjz69rN0SEHfNK2A@mail.gmail.com> <CALDtMrKb3_38Rs0vaGnpEvNvTYz8YUTo89STvLJNXfkfdipDSQ@mail.gmail.com> <93BEDDC39A54294B9E78C7860516FA4724AA4422@AZ-US1EXMB06.global.avaya.com> <B7FA2629-6D48-4569-BB62-56395C3EE4BC@cisco.com> <04ff01cf31b4$8eb73780$ac25a680$@stahl@intertex.se> <AA208926-C949-4580-B20B-DCF172D3C21B@cisco.com> 
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Date: Thu, 13 Mar 2014 08:56:44 +0100
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Cc: tram@ietf.org, 'Spencer Dawkins' <spencer@wonderhamster.org>, rtcweb@ietf.org
Subject: Re: [rtcweb] [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification regardig Milestone 3: TURN server auto-discovery mechanism for enterprise and ISPs
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Hi Cullen,

=20

You may first want to read the previous email to Harald:=20

http://www.ietf.org/mail-archive/web/tram/current/msg00349.html=20

=20

Regarding your request from February 26:
> Karl,=20

> I am totally lost on this thread. Could you start a new tmead that
summarize what the issues is, what seems to be the point of debate, and =
what
your view is on what we should do. I think that would help make =
progress.=20

> Thank you, Cullen

=20

I already February 26 started a new thread for the =93mission to bring =
quality
to real-time traffic over our best effort Internet=94
<http://www.ietf.org/mail-archive/web/tram/current/msg00331.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html

[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff [1]=20

=20

And from your email February 25 below:

> Reading the charter, the above is *not* at all a clear objective of =
the WG
(note I am not the chair of this WG or the responsible AD).=20

=20

>That said, I think you have pointed out this charter is abysmally vague =
-
it does not say what the WG is not going to do. If I decided to do BGP =
for
routing updates over TURN it would be within the scope of this charter.=20

=20

>My advice to the responsible AD is recharter this WG before IETF 90 or
close it. I would be glad to help write a charter that is not an =
infinite
blank cheque.

=20

I actually thought it was you (Cullen) that at least initiated tram
Milestone 3 with Subject: =93TURN server auto-discovery mechanism for
enterprise and ISPs=94 [2]

=20

After your request
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09007.html=20

[Karl, I, had said]=93> I've heard several carries expressing that they =
want
to provide their own TURN servers with their accesses.

=20

[Cullen asked] Any chance you could put me in touch with some of them. =
I'd
love to get more information on what is needed so we can solve this.=93

=20

The =93 huge European carrier and the cable operators in general =96 =
CableLabs=94
that I refer to in
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html  were =
the
ones I put you in touch with, and then China Mobile reached out for =
getting
the auto-discovery mechanism in place: =93Let me also point out
draft-deng-tram-isp-turn-00
http://www.ietf.org/mail-archive/web/tram/current/msg00214.html from =
China
Mobile that appeared on this mailing list yesterday. It points out the =
need
and willingness from the ISP/NSP side to do something about QoS for =
WebRTC
traffic, that they expect to be large and have to bring to their =
customers
with best QoE. In fact they and some more (a huge European carrier and =
the
cable operators in general - CableLabs) have expressed similar concerns =
to
me *=93This is exactly something we want to work on as the current way =
will
cause severe problem on traffic when rtcweb applications get popular=94* =
is a
direct quote from one of those.=93

=20

Whoever you left this to, it somehow appeared as [tram] Milestone 3. [2]

=20

The complete Anycast method for auto discovery of ISP=92s TURN servers =
was
suggested October 22 (without further discussions)

http://www.ietf.org/mail-archive/web/rtcweb/current/msg09128.html=20

and thereafter brought into [tram] Milestone 3 by me February 8=20

http://www.ietf.org/mail-archive/web/tram/current/msg00132.html=20

=20

It took me some time to find the relevant emails from the discussions in
September and October which ended up with my entry to the [tram]-list
February 8, where I still say it was 95% ready (there are a few relevant
points from Simon that I will address later). Thereafter, the QoS =
diversion
and confusion started around February 12th in what looks to me as an =
effort
to make it a =93Mission Impossible=94.

=20

So referring to
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html, step A) =
is
for the TRAM WG  to allow ISPs/NSPs to route quality demanding WebRTC =
media
into their IP pipes that are capable of transporting real-time traffic,
using TURN servers. Step B), how the WebRTC browser shall use the auto
discovered TURN server, which I suggested in=20

http://www.ietf.org/mail-archive/web/tram/current/msg00132.html=20

=20

WEB BROWSER BEHAVIOUR:

=20

Network provided TURN servers will not appear over night, applications =
may
for long provide a TURN server address, and there are exceptions where =
the
TURN server address is preferred to be =93manually=94 configured. It is
previously suggested, and to some extent discussed, that the WebRTC =
browser
should select the TURN server to use in the following priority order, =
where
ICE would assure that you get some connectivity if several candidates =
are
found and needs to be tested:

=20

1) TURN server address configured in the browser by the user (special =
cases,
normally not used, but handy for testing)

2) TURN server address configured by the network administrator via an =
=93admin
policy template=94 or a WPAD method as mentioned below

3) TURN server address auto-discovered by the mechanism discussed here

4) TURN server address being supplied by the web application

=20

may be for the RTCWEB WG rather than TRAM WG to handle. Further I would
guess step D) is an AVTEXT WG thing =96 What do you think?

=20

/Karl

=20

Footnotes:

=20

[1] Please do not divert or confuse this with QoS methods in themselves
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the
interface from the application to the network level, where all networks
types should be able to use the traffic information for QoS methods =
relevant
to the particular network.

=20

[2] I think both you (Cullen) and Harald, are good, have the best =
intentions
and integrity, but no one of us have infinite memory capacity =
(especially
when the VP8/H.264 flooding came in between). I only learned =93by =
accident=94
that the TRAM WG was in progress at the November WebRTC conference and
should handle the Auto Discovery of TURN-servers.

=20

PS: To avoid being stopped by the =93to many recipients=94 spam filter, =
the Guys
(including Mary) from=20

http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html=20

and the ones mentioned in the email are copied separately.

=20

-----Ursprungligt meddelande-----

Fr=E5n: Cullen Jennings (fluffy) [ <mailto:fluffy@cisco.com>
mailto:fluffy@cisco.com]

Skickat: den 26 februari 2014 01:49

Till: Karl Stahl

Kopia: Alan Johnston;  <mailto:rtcweb@ietf.org> rtcweb@ietf.org;
<mailto:tram@ietf.org> tram@ietf.org; Bernard Aboba; David Singer; =
Harald
Tveit Alvestrand; Marc Robins; Eric Burger; Mary Barnes; Henning =
Schulzrinne

=C4mne: Re: [rtcweb] [tram] I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt

=20

=20

> Karl,=20

=20

> I am totally lost on this thread. Could you start a new tmead that
summarize what the issues is, what seems to be the point of debate, and =
what
your view is on what we should do. I think that would help make =
progress.=20

=20

> Thank you,=20

=20

> Cullen

=20

I just did something like that in:

[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff=20

 <http://www.ietf.org/mail-archive/web/tram/current/msg00331.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html=20

Hoping this thread subject amplifies what this SHOULD be about. (An how =
it
will lead to the summary, after all diversion.)

=20

Actually, the outcome of "Interfacing to QoS" will be=20

=20

=20

I challenge anyone that this will be outcome, and will start =
"Interfacing to
QoS" with what IP "Interfacing to QoS" that we have the tiny ADSL modem =
IX78
that implements such CLEAR INTERFACE and the actual QoS methods, for =
both
the Congestion, Default gateway, Surf Pipe=20

Manipulating the RTP extension header etc etc on 6 USD CPU and also =
include
the ADSL part where that quality handling is based on heavy 57 bytes
chopping up

   at this exact=20

=20

=20

=20

=20

-----Ursprungligt meddelande-----

Fr=E5n: Cullen Jennings (fluffy) [ <mailto:fluffy@cisco.com>
mailto:fluffy@cisco.com]=20

Skickat: den 26 februari 2014 02:06

Till: Karl Stahl; IESG IESG

Kopia: Magnus Westerlund; Simon Perreault; Ted Hardie; Gonzalo =
Camarillo;
<mailto:rtcweb@ietf.org> rtcweb@ietf.org;  <mailto:tram@ietf.org>
tram@ietf.org; Spencer Dawkins

=C4mne: Re: [tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter
Clarification regardig Milestone 3: TURN server auto-discovery mechanism =
for
enterprise and ISPs

=20

=20

=20

On Feb 25, 2014, at 7:02 AM, Karl Stahl < =
<mailto:karl.stahl@intertex.se>
karl.stahl@intertex.se> wrote:

=20

> To the TRAM WG and RTCWEB WG and ADs:

> =20

> It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed =
and
encouraged to route quality demanding WebRTC media into their IP pipes =
that
are capable of transporting real-time traffic without quality issues, =
using
TURN servers.

> =20

=20

Reading the charter, the above is *not* at all a clear objective of the =
WG
(note I am not the chair of this WG or the responsible AD).=20

=20

That said, I think you have pointed out this charter is abysmally vague =
- it
does not say what the WG is not going to do. If I decided to do BGP for
routing updates over TURN it would be within the scope of this charter.=20

=20

My advice to the responsible AD is recharter this WG before IETF 90 or =
close
it. I would be glad to help write a charter that is not an infinite =
blank
cheque.=20

=20

=20

=20

=20

=20


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</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 Cullen,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Yo=
u may first want to read the previous email to Harald: =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00349.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00349.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p>&nbsp;<=
/o:p></span></b></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>R=
egarding your request from February 26:<br></span><span =
lang=3DEN-US>&gt; Karl, <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; I am totally lost on this =
thread. Could you start a new tmead that summarize what the issues is, =
what seems to be the point of debate, and what your view is on what we =
should do. I think that would help make progress. =
<o:p></o:p></span></p><p class=3DMsoPlainText>&gt; Thank you, =
Cullen<o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'><=
o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
already February 26 started a new thread for the</span><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif";color:blue'> =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;</span><span lang=3DEN-US>mission to bring quality to real-time =
traffic over our best effort Internet&#8221;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00331.html">=
<span lang=3DEN-US =
style=3D'color:blue'>http://www.ietf.org/mail-archive/web/tram/current/ms=
g00331.html</span></a></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
[tram] [rtcweb] The way to &quot;Interfacing to QoS&quot;, A level 3-5 =
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>[=
1] </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'><=
o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>A=
nd from your email February 25 below:</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt; Reading the charter, the above is *not* at all a clear =
objective of the WG (note I am not the chair of this WG or the =
responsible AD). <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;That said, I think you have pointed out this charter is =
abysmally vague - it does not say what the WG is not going to do. If I =
decided to do BGP for routing updates over TURN it would be within the =
scope of this charter. <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt;My advice to the responsible =
AD is recharter this WG before IETF 90 or close it. I would be glad to =
help write a charter that is not an infinite blank =
cheque.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
actually thought it was you (Cullen) that at least initiated tram =
Milestone 3 with Subject: &#8220;</span><span lang=3DEN-US>TURN server =
auto-discovery mechanism for enterprise and ISPs</span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221; [2]<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Af=
ter your request <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09007.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09007.html</a> =
<o:p></o:p></span></p><pre><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:blue'>[Karl, I, had =
said]&#8220;</span><span lang=3DEN-US>&gt; I've heard several carries =
expressing that they want to provide their own TURN servers with their =
accesses.<o:p></o:p></span></pre><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[C=
ullen asked]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier New","serif"'> Any chance =
you could put me in touch with some of them. I'd love to get more =
information on what is needed so we can solve this.</span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e &#8220;</span><span lang=3DEN-US> huge European carrier and the cable =
operators in general &#8211; CableLabs&#8221; that I refer to in =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00275.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html</a> =
=A0were the ones I put you in touch with, and then China Mobile reached =
out for getting the auto-discovery mechanism in place: =
&#8220;</span><span lang=3DEN-US>Let me also point out =
draft-deng-tram-isp-turn-00 =A0<a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00214.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00214.html</a> from =
China Mobile that appeared on this mailing list yesterday. It points out =
the need and willingness from the ISP/NSP side to do something about QoS =
for WebRTC traffic, that they expect to be large and have to bring to =
their customers with best QoE. In fact they and some more (a huge =
European carrier and the cable operators in general - CableLabs) have =
expressed similar concerns to me *&#8220;This is exactly something we =
want to work on as the current way will cause severe problem on traffic =
when rtcweb applications get popular&#8221;* is a direct quote from one =
of those.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wh=
oever you left this to, it somehow appeared as [tram] Milestone 3. =
[2]</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e complete Anycast method for auto discovery of ISP&#8217;s TURN servers =
was suggested October 22 (without further =
discussions)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09128.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09128.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>an=
d thereafter brought into [tram] Milestone 3 by me February 8 =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00132.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00132.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>It=
 took me some time to find the relevant emails from the discussions in =
September and October which ended up with my entry to the [tram]-list =
February 8, where I still say it was 95% ready (there are a few relevant =
points from Simon that I will address later). Thereafter, the QoS =
diversion and confusion started around February 12th in what looks to me =
as an effort to make it a &#8220;Mission =
Impossible&#8221;.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>So=
 referring to <a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00275.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html</a>, =
step A) is for the TRAM WG =A0to allow</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
ISPs/NSPs to route quality demanding WebRTC media into their IP pipes =
that are capable of transporting real-time traffic, using TURN servers. =
Step B), how the WebRTC browser shall use the auto discovered TURN =
server, which I suggested in <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00132.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00132.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>WEB BROWSER BEHAVIOUR</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>:<=
/span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>Network provided TURN servers will not appear over night, applications =
may for long provide a TURN server address, and there are exceptions =
where the TURN server address is preferred to be &#8220;manually&#8221; =
configured. It is previously suggested, and to some extent discussed, =
that the WebRTC browser should select the TURN server to use in the =
following priority order, where ICE would assure that you get some =
connectivity if several candidates are found and needs to be =
tested:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>1) TURN server address configured in the browser by the user (special =
cases, normally not used, but handy for testing)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>2) TURN server address configured by the network administrator via an =
&#8220;admin policy template&#8221; or a WPAD method as mentioned =
below<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>3) TURN server address auto-discovered by the mechanism discussed =
here<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>4) TURN server address being supplied by the web =
application<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
y be for the RTCWEB WG rather than TRAM WG to handle. Further I would =
guess step D) is an AVTEXT WG thing &#8211; What do you =
think?<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
otnotes:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[1=
] Please do not divert or confuse this with QoS methods in themselves =
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the interface from the application to the network level, where all =
networks types should be able to use the traffic information for QoS =
methods relevant to the particular network.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[2=
] I think both you (Cullen) and Harald, are good, have the best =
intentions and integrity, but no one of us have infinite memory capacity =
(especially when the VP8/H.264 flooding came in between). I only learned =
&#8220;by accident&#8221; that the TRAM WG was in progress at the =
November WebRTC conference and should handle the Auto Discovery of =
TURN-servers.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>PS=
: To avoid being stopped by the &#8220;to many recipients&#8221; spam =
filter, the Guys (including Mary) from <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11548.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>an=
d the ones mentioned in the email are copied =
separately.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText>-----Ursprungligt meddelande-----<o:p></o:p></p><p =
class=3DMsoPlainText>Fr=E5n: Cullen Jennings (fluffy) [<a =
href=3D"mailto:fluffy@cisco.com"><span =
style=3D'color:windowtext;text-decoration:none'>mailto:fluffy@cisco.com</=
span></a>]<o:p></o:p></p><p class=3DMsoPlainText>Skickat: den 26 =
februari 2014 01:49<o:p></o:p></p><p class=3DMsoPlainText>Till: Karl =
Stahl<o:p></o:p></p><p class=3DMsoPlainText>Kopia: Alan Johnston; <a =
href=3D"mailto:rtcweb@ietf.org"><span =
style=3D'color:windowtext;text-decoration:none'>rtcweb@ietf.org</span></a=
>; <a href=3D"mailto:tram@ietf.org"><span =
style=3D'color:windowtext;text-decoration:none'>tram@ietf.org</span></a>;=
 Bernard Aboba; David Singer; Harald Tveit Alvestrand; Marc Robins; Eric =
Burger; Mary Barnes; Henning Schulzrinne<o:p></o:p></p><p =
class=3DMsoPlainText>=C4mne: Re: [rtcweb] [tram] I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>&gt; =
Karl, <o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>&gt; I am totally lost on this thread. <span =
lang=3DEN-US>Could you start a new tmead that summarize what the issues =
is, what seems to be the point of debate, and what your view is on what =
we should do. </span>I think that would help make progress. =
<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>&gt; Thank you, <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>&gt; =
Cullen<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>I just did something like that in:<o:p></o:p></p><p =
class=3DMsoPlainText>[tram] [rtcweb] The way to &quot;Interfacing to =
QoS&quot;, A level 3-5 IP/IETF/WebRTC-thing how to interface to lower =
level's QoS-stuff <o:p></o:p></p><p class=3DMsoPlainText><a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00331.html">=
<span =
style=3D'color:windowtext;text-decoration:none'>http://www.ietf.org/mail-=
archive/web/tram/current/msg00331.html</span></a> <o:p></o:p></p><p =
class=3DMsoPlainText>Hoping this thread subject amplifies what this =
SHOULD be about. (An how it will lead to the summary, after all =
diversion.)<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>Actually, the outcome of &quot;Interfacing to =
QoS&quot; will be <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>I =
challenge anyone that this will be outcome, and will start =
&quot;Interfacing to QoS&quot; with what IP &quot;Interfacing to =
QoS&quot; that we have the tiny ADSL modem IX78 that implements such =
CLEAR INTERFACE and the actual QoS methods, for both the Congestion, =
Default gateway, Surf Pipe <o:p></o:p></p><p =
class=3DMsoPlainText>Manipulating the RTP extension header etc etc on 6 =
USD CPU and also include the ADSL part where that quality handling is =
based on heavy 57 bytes chopping up<o:p></o:p></p><p =
class=3DMsoPlainText>=A0=A0 at this exact <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>-----Ursprungligt meddelande-----<o:p></o:p></p><p =
class=3DMsoPlainText>Fr=E5n: Cullen Jennings (fluffy) [<a =
href=3D"mailto:fluffy@cisco.com"><span =
style=3D'color:windowtext;text-decoration:none'>mailto:fluffy@cisco.com</=
span></a>] <o:p></o:p></p><p class=3DMsoPlainText>Skickat: den 26 =
februari 2014 02:06<o:p></o:p></p><p class=3DMsoPlainText>Till: Karl =
Stahl; IESG IESG<o:p></o:p></p><p class=3DMsoPlainText>Kopia: Magnus =
Westerlund; Simon Perreault; Ted Hardie; Gonzalo Camarillo; <a =
href=3D"mailto:rtcweb@ietf.org"><span =
style=3D'color:windowtext;text-decoration:none'>rtcweb@ietf.org</span></a=
>; <a href=3D"mailto:tram@ietf.org"><span =
style=3D'color:windowtext;text-decoration:none'>tram@ietf.org</span></a>;=
 Spencer Dawkins<o:p></o:p></p><p class=3DMsoPlainText>=C4mne: Re: =
[tram] [RTCWEB] [TRAM] Protesting: Requesting TRAM Charter Clarification =
regardig Milestone 3: TURN server auto-discovery mechanism for =
enterprise and ISPs<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>On Feb =
25, 2014, at 7:02 AM, Karl Stahl &lt;<a =
href=3D"mailto:karl.stahl@intertex.se"><span =
style=3D'color:windowtext;text-decoration:none'>karl.stahl@intertex.se</s=
pan></a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>&gt; =
To the TRAM WG and RTCWEB WG and ADs:<o:p></o:p></p><p =
class=3DMsoPlainText>&gt;=A0 <o:p></o:p></p><p class=3DMsoPlainText>&gt; =
It must be a clear objective of the TRAM WG that ISPs/NSPs are allowed =
and encouraged to route quality demanding WebRTC media into their IP =
pipes that are capable of transporting real-time traffic without quality =
issues, using TURN servers.<o:p></o:p></p><p =
class=3DMsoPlainText>&gt;=A0 <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>Reading the charter, the above is *not* at all a =
clear objective of the WG (note I am not the chair of this WG or the =
responsible AD). <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>That =
said, I think you have pointed out this charter is abysmally vague - it =
does not say what the WG is not going to do. If I decided to do BGP for =
routing updates over TURN it would be within the scope of this charter. =
<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>My advice to the responsible AD is recharter this =
WG before IETF 90 or close it. I would be glad to help write a charter =
that is not an infinite blank cheque. <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p></div></body></html>
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Pal Martinsen \(palmarti\)'" <palmarti@cisco.com>, "'Colin Perkins'" <csp@csperkins.org>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] [tram] RTP header extension for "Interfacing to QoS" WAS: I-D Action: draft-thomson-tram-turn-bandwidth-00.txt
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For the =93mission to bring quality to real-time traffic over our best =
effort
Internet=94 I have started
<http://www.ietf.org/mail-archive/web/tram/current/msg00331.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html

[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff [1] [2]

=20

=20

*SUMMARY* to be brought into the common "Interfacing to QoS"-thread

(From there it will be clearer how to split =93Interfacing to QoS=94 =
into
[tram], [rtcweb] and [avtext]-threads.)

=20

The steps A), B) to D) for achieving the mission:

http://www.ietf.org/mail-archive/web/tram/current/msg00273.html

=20

a: Step D) using the RTP extension header RCF5285 (that works for all IP
network types) can obsolete step C) (Using STUN-attributes which is an
incomplete transfer of traffic info to the network level.) We need =
traffic
info (bandwidth and type) that is unchanged through various networks and =
can
be read and used by network elements (here in a TURN-server flow) and =
for
traffic in both directions.

=20

b: Here it is only discussed for RTP (not for the data channel), but it
could be good to also consider this for the webrtc data channel. I will
discuss that in next email responding to Magnus=92
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11549.html coming
soon.

=20

c: How to find and pick out the traffic information from the RTP =
extension
header when RCF 5285 is used for this purpose is trivial, especially =
when
=93in-band=94 in a known TURN-flow.=20

=20

There are further detail in-line below for P=E5l, Colin, Magnus and =
other
interested.

=20

/Karl

=20

Footnotes:

=20

[1] Please do not divert or confuse this with QoS methods in themselves
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the
interface from the application to the network level, where all networks
types should be able to use the traffic information for QoS methods =
relevant
to the particular network.

=20

[2] Here it is about recreation of the idea/intention of the RTP payload
type (PT) header that is not available for the network level anymore, by
instead having the application/browser filling the RTP header extension =
with
relevant traffic info that all IP network types can use. There will be =
more
detail in the response to Magnus=92
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11549.html coming
soon.

=20

Fr=E5n: Pal Martinsen (palmarti) [mailto:palmarti@cisco.com]=20
Skickat: den 25 februari 2014 13:48
Till: Karl Stahl
Kopia: tram@ietf.org; rtcweb@ietf.org; Oleg Moskalenko; Alan Johnston;
Yoakum, John H (John)
=C4mne: Re: [tram] [rtcweb] I-D Action:
draft-thomson-tram-turn-bandwidth-00.txt

=20

On 24 Feb 2014, at 23:22 pm, Karl Stahl < =
<mailto:karl.stahl@intertex.se>
karl.stahl@intertex.se> wrote:

=20

Hi P=E5l,

=20

You did not comment nor answer, in response to any of:

 <http://www.ietf.org/mail-archive/web/tram/current/msg00275.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00275.html

 <http://www.ietf.org/mail-archive/web/tram/current/msg00273.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00273.html

=20

Sorry about that. I did not notice any clear items to answer or comment =
on.=20

[Karl -regarding =93Interfacing to QoS=94] From what you answer below, I
conclude:

a: step D) (that works for all IP network types) can obsolete step C) =
(using
STUN-attributes which is an incomplete transfer of traffic info to the
network level) (We need a quality marking that is unchanged and can be =
read
and used throughout various networks and for traffic in both =
directions).

b: here it is only discussed for RTP (not for the data channel, but I =
will
discuss that in next email responding to Magnus.

c: you have hesitations how to find and pick out the traffic information
from the RTP extension header when RCF5285 is used for this purpose, but
that is trivial, especially when =93in-band=94 in a known TURN-flow. See =
further
below.

=20

=20

(DISCUSS/MALICE) by allowing the application/browser to transfer =
relevant
real-time traffic information (types and bandwidth) into every RTP =
package
by using the already IETF standardized RTP extension header RCF 5285, =
which
could be used by any current or future Internet (including OTT) network
where WebRTC real-time traffic may flow. It seems to have been =
standardized
already 2008 to allow such usage.





whether step D) can obsolete step C) (DISCUSS/MALICE) by allowing the
application/browser to transfer relevant real-time traffic information
(types and bandwidth) into every RTP package by using the already IETF
standardized RTP extension header RCF 5285, which could be used by any
current or future Internet (including OTT) network where WebRTC =
real-time
traffic may flow. It seems to have been standardized already 2008 to =
allow
such usage.

=20

[P=E5l] draft-martinsen-tram-discuss is intended to solve internal =
application
stream prioritisation pr flow, i.e. it is more important for me to get =
the
video stream across than my application stream at this moment in time. =
Its
intention is to describe the 5-topple the STUN message was sent on.=20

=20

Moving to a pr packet prioritisation is interesting as it can solve the
bundle problem, and mark the relative importance of specific video =
frames
(I-Frames for example). Since its pr packet the network element might =
not
keep stream state so it might be easier that way. But the information =
needs
to be in a fixed offset for quick network element processing. And we =
want to
describe other streams than just RTP. That makes RTP header extensions
unsuitable.=20

=20

QoS related step C) draft-martinsen-tram-discuss can then be taken out =
of
TRAM, and step D) (that was never intended to be handled by TRAM anyway)
will be handled elsewhere.

=20

[P=E5l] I was hoping to argue that the draft is only new signalling of =
old
concepts (DSCP and ECN). In the end it is the WG that decides if there =
is
enough interest.=20

=20

=20

I have repeated my suggestion to RTCWEB WG from the October discussions =
on
the relevant [rtcweb] [avtext] [mmusic] lists
<http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html>
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html to
introduce the QoS related step D) into draft-ietf-rtcweb-rtp-usage
suggesting usage of RFC5285, where traffic information in RTP packets =
simply
would be filled by any browser under any OS.

=20

Without step B), TRAM =93Enforcing the real-time traffic through the
offered/discovered TURN servers=94, the real-time traffic may happen =
through
the IP default gateways often congested by data traffic and QoS =
insensitive
streaming video and file sharing. Without specific QoS methods at those
points, the network raw bandwidth capacity may have to be 10-folded to
achieve sufficient QoE when WebRTC usage becomes popular. Methods like
DISCUSS addresses such things occurring when STUN instead TURN is used =
(by
allowing flows into such congestion points), but only provides direct
traffic info for outgoing (not incoming) real-time traffic and locally,
therefore are not even applicable to reservation type of networks like =
Cable
Networks and Mobile OTT.

=20

[P=E5l] If both clients supports draft-martinsen-tram-discuss you would =
get an
bi-directional end to end description of the flow that works through =
NAT.

The information in the STUN packet can be used to remark DSCP bits to
whatever values that is of best use for the network the packet is =
flowing
though.

[Karl] Again, this is only applicable for all-diffserve networks and you
cannot even route the flow to a quality pipe, not even a duplicate pipe =
that
is not the default gateway surf pipe (which is a very common method used =
for
VoIP).

=20

[P=E5l] I dont see how TURN gives that ability.=20

[Karl - ???] Haven=92t we many times agreed that you can add exactly the =
same
STUN-attributes to the TURN allocate request AND in addition to that,
control where the flow goes?

BUT THIS IS ALSO INSUFFICIENT.=20

I discarded such (similar) interface to QoS already October 22

 <http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html>
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=20

=85

PS Microsoft seems to have done work in this field, defining a =
proprietary
attribute =93MS Service Quality=94;=20

However that seems to apply to the TURN server allocation request and =
would
therefore:

--- Apply to the whole UDP flow, and could not be set for each stream
individually (with different requirements), and

--- Does not handle the bandwidth requirement for incoming real-time =
traffic
(required to reserve in RSVP type of networks)

However the quality attributes conveyed and their encoding may  be
considered.

This is 2.2.2.19 MS-Service Quality Attribute from=20

 <http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx>
http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx=20

=85

[P=E5l] Once the channel is bound you only have a channel number in =
front of
the UDP payload. And TURN is designed to carry alls sort of traffic. For
this to work there must be some sort of only webRTC traffic is allowed
through this TURN server. (And Ill bet bitorrent or others find a way to
mimic webRTC to set up only a data channel..)

[Karl] I owe Simon a response to misuse of prioritized channels =96 will =
do
later after first collecting these diversions. This has nothing to do =
with
STUN vs TURN! TURN just gives the addition of ALSO directing the flow. =
In
both cases it is insufficient to transfer traffic info in the STUN/TURN
attributes.
=20

That would leave certain network types to investing in raw bandwidth
increase, instead of simply borrowing the bandwidth (from much larger =
data
traffic and QoS insensitive streaming video and file sharing at no =
adverse
effect) and making that borrowed no-cost bandwidth very valuable for the
carriers when delivering to their customers.

I know a few of you Cisco guys are among the ones that have been =
fighting
against the general =93it is all about bandwidth=94 and =93it will go =
away with
time=94-attitude within IETF work, e.g. see
<http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html>
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html, but =
the
pointers given in the current QoS discussion within TRAM:

=20

(i) will *not allow* ISP=92s to use already available and currently =
deployable
quality IP pipes for real-time traffic to also be used for WebRTC =
generated
real-time traffic.

=20

(ii) will *not allow* some network types (e.g. Cable Networks and Mobile
OTT) to borrow bandwidth from data traffic and QoS insensitive streaming
video and file sharing. All networks will *also be inhibited* from the =
very
common and used method of simply providing an extra IP pipe (often =
provided
over the same wire but level-2 separated) dedicated for real-time usage

(*by resisting TRAM implementation of step B*)

Newer, fiber only type of networks, can still borrow bandwidth at no =
extra
cost, *by proprietary usage* of RFC5285 by browsers.

=20

(iii) will leave most ISPs to *only use* raw bandwidth capacity increase
that may has to be 10-folded to reach sufficient QoE when WebRTC usage
becomes popular (if at all possible, since unmanaged IP pipes =
intermittently
are filled, whatever bandwidth is available).

=20

Further explainations about QoS methods and their implementation were =
given
a few days ago in:

 <http://www.ietf.org/mail-archive/web/tram/current/msg00274.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00274.html

=20

(i), (ii) and (iii) will *seriously delay* usage of telepresence capable =
and
quality demanding WebRTC (bandwidth being around 30 times higher over =
best
effort Internet, than telephony over quality managed networks) and will
*vastly increase cost* for ISPs to offer WebRTC with good QoE.

=20

Below you are giving pointers saying =93They are currently working on a
problem-statement draft and a use-case draft, any input to those would =
be
very helpful.=94 Those are unnecessary, risking leading to (i), (ii) and =
(iii)
above.

=20

[P=E5l] We find them quite useful. It allows us to engage with potential =
users
and see if there are real problems to solve. If the customer/users do =
not
see a problem there is little incentive for us to invest in creating a
solution that nobody wants. That said, it is always the danger of =
crating a
to complicated solution that solves to many problems. =20

[Karl] This is QoS stuff. Let=92s stick to =93Interfacing to QoS=94 and =
with a
general interface, that the =93QoS device including the TURN server=94 =
should be
able to do whatever network QoS-method you arrive at.



So are pointers such as: =93RMCAT where congestion avoidance for RTP is =
being
developed=94. TRAMs Milestone 3 is for the purpose of directing =
real-time
traffic where congestion control isn=92t already in place. Using RFC5285
available since 2008, to fill traffic information in RTP packets (step =
D))
is probably a necessity for most future =93congestion control=94 =
discussed.

=20

[P=E5l] There are some good example of RTP header extension usage. For =
example
draft-avtext-berger-framemarking. But that information may very well be
encrypted and unavailable to the network element.

[Karl] You (P=E5l), Colin and Magnus seem to worry about trivial things =
when
using the RTP header extension, for transferring traffic information =
from
the application (here browser) to the network level. Don=92t worry! J  =
These
are really trivial things :

- If we select to use the RTP extension header RCF5285 for this purpose, =
it
should of course NOT use RFC6904 encryption, since it is meant for =
network
level reading/usage!

- Finding the RTP header extension (especially when =93in-band=94 in a =
known
TURN-flow) is really trivial. =20

Linking each RTP-flow by its SSRC and sequence number, and picking =
exactly
the right traffic type and bandwidth parameters is no problem (I am not
aware of any =93Zero CPU=94 DPI device that needs to be considered =96 =
But if we
can ease the job, we should.)

This link shows much more advanced things are available in Cisco =
devices:=20

http://www.cisco.com/en/US/products/ps6616/products_white_paper09186a0080=
110
040.shtml#wp39132=20

and the ADSL 2+ modem doing a lot of advanced QoS-stuff, described in=20

http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html=20

is our Intertex/Ingate IX78, just powered by a 6 USD 264 MHz ARM ADSL =
chip
set CPU and we have implemented a check box TO REMOVE the RTP Header
extension (for the ugly reason that the ITSP=92s billing system requires =
G.711
VoIP packets of a certain length) =96 It does not take noticeable CPU =
power to
do this at 100 Mbps wire speed!=20



/Karl=20

.-.

P=E5l-Erik





This chicken-and-egg circular also applies to RTCWEB direction of QoS =
issues
to:

 <http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos>
http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos

focusing on DSCP-mapping without even mentioning RFC 5285 available =
since
2008, to fill traffic information in RTP packets where it  is a =
necessity
and will allow:

(1) applications to directly convey QoS related real-time traffic info =
to
the network at points where RTP flow is directed to by TRAM Milstone 3, =
to
be used by *any network element implementing any suitable QoS methods =
for
the particular network* for

(2) *all* WebRTC browsers *and* dedicated clients (not using WebRTC), =
under
*all* OSs, and *all* current and future IP networks

(3) *without* having to be forced into application specific networks =
(PSTN,
IMS) instead of using the Internet (including OTT).

=20

The only activity required, is to call for ISPs=92 review whether the =
traffic
information transferred by RFC 5285 is sufficient for current and future
needs in their network as suggested in
<http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html>
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html

<snip>

=85two parameters (e.g. two bytes each) are encoded into the RTP header
extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each for quality type e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

</snip>

=20

Then this could be assigned numbers to have the RFC in place.

With TRAM milestone 3 also place,

market forces will drive ISPs and browser makers to implement just that,
without even having it MUST-established.

=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and

=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with
much better QoE?=94 and vice versa.

 l

Please see further emails soon following this one, for details and =
history.

=20

/Karl

=20

Fr=E5n: tram [ <mailto:tram-bounces@ietf.org> =
mailto:tram-bounces@ietf.org]
F=F6r Pal Martinsen (palmarti)
Skickat: den 21 februari 2014 10:37
Till: tram@ietf.org
Kopia: Karl Stahl; Oleg Moskalenko; Alan Johnston; Yoakum, John H (John)
=C4mne: Re: [tram] I-D Action: draft-thomson-tram-turn-bandwidth-00.txt

=20

Hi,

=20

I agree the full QoS discussion should _not_ happen in TRAM. If you are
interested in helping out in that area I suggest you looking into the =
AEON
mailing list at:  <https://www.ietf.org/mailman/listinfo/aeon>
https://www.ietf.org/mailman/listinfo/aeon . They are currently working =
on a
problem-statement draft and a use-case draft, any input to those would =
be
very helpful. ( =
<http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01>
http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01,
<http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00>
http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00).

=20

That said, STUN have a few nice characteristics that makes it a perfect
candidate for transporting some of the QoS information.  IMHO that would =
be
extending the STUN spec and should be within the TRAM charter.  The main
goal of draft-martinsen-tram-discuss was to show how already existing =
QoS
mechanisms could be transported with STUN to provide more value, and to
start the discussion if TRAM is the appropriate place to have those on =
the
wire format discussions.

=20

.-.

P=E5l-Erik


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lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;</span><span lang=3DEN-US>mission to bring quality to real-time =
traffic over our best effort Internet&#8221;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
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<a =
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[tram] [rtcweb] The way to &quot;Interfacing to QoS&quot;, A level 3-5 =
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>[=
1] [2]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></b></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>*S=
UMMARY*</span></b><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
to be brought into the common &quot;Interfacing to =
QoS&quot;-thread<o:p></o:p></span></b></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(F=
rom there it will be clearer how to split &#8220;Interfacing to =
QoS&#8221; into [tram], [rtcweb] and =
[avtext]-threads.)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e steps A), B) to D) for achieving the mission:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00273.html">=
http://www.ietf.org/mail-archive/web/tram/current/msg00273.html</a><o:p><=
/o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>a:=
 Step D) using the RTP extension header RCF5285 (that works for all IP =
network types) can obsolete step C) (Using STUN-attributes which is an =
incomplete transfer of traffic info to the network level.) We need =
traffic info (bandwidth and type) that is unchanged through various =
networks and can be read and used by network elements (here in a =
TURN-server flow) and for traffic in both =
directions.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>b:=
 Here it is only discussed for RTP (not for the data channel), but it =
could be good to also consider this for the webrtc data channel. I will =
discuss that in next email responding to Magnus&#8217; <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11549.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11549.html</a> =
coming soon.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>c:=
 How to find and pick out the traffic information from the RTP extension =
header when RCF 5285 is used for this purpose is trivial, especially =
when &#8220;in-band&#8221; in a known TURN-flow. </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
ere are further detail in-line below for P=E5l, Colin, Magnus and other =
interested.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
otnotes:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[1=
] Please do not divert or confuse this with QoS methods in themselves =
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the interface from the application to the network level, where all =
networks types should be able to use the traffic information for QoS =
methods relevant to the particular network.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[2=
] Here it</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
is about recreation of the idea/intention of the RTP payload type (PT) =
header that is not available for the network level anymore, by instead =
having the application/browser filling the </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>RT=
P header extension with relevant traffic info that all IP network types =
can use. There will be more detail in the response to Magnus&#8217; <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11549.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11549.html</a> =
coming soon.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Pal =
Martinsen (palmarti) [mailto:palmarti@cisco.com] <br><b>Skickat:</b> den =
25 februari 2014 13:48<br><b>Till:</b> Karl Stahl<br><b>Kopia:</b> =
tram@ietf.org; rtcweb@ietf.org; Oleg Moskalenko; Alan Johnston; Yoakum, =
John H (John)<br><b>=C4mne:</b> Re: [tram] [rtcweb] I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt<o:p></o:p></span></p></div></div=
><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US>On 24 Feb 2014, at 23:22 pm, Karl =
Stahl &lt;</span><a href=3D"mailto:karl.stahl@intertex.se"><span =
lang=3DEN-US>karl.stahl@intertex.se</span></a><span lang=3DEN-US>&gt; =
wrote:<o:p></o:p></span></p></div><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 P=E5l,</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Yo=
u did not comment nor answer, in response to any =
of:</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00275.html">=
<span =
style=3D'color:purple'>http://www.ietf.org/mail-archive/web/tram/current/=
msg00275.html</span></a></span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00273.html">=
<span =
style=3D'color:purple'>http://www.ietf.org/mail-archive/web/tram/current/=
msg00273.html</span></a></span><o:p></o:p></p></div></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Sorry about that. I did not notice any clear items to =
answer or comment on.&nbsp;<o:p></o:p></p></div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#C00000'>[Karl =
-regarding &#8220;Interfacing to QoS&#8221;] From what you answer below, =
I conclude:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:#C00000'>a: step D) (that works for all IP network types) =
can obsolete step C) (using STUN-attributes which is an incomplete =
transfer of traffic info to the network level) (We need a quality =
marking that is unchanged and can be read and used throughout various =
networks and for traffic in both directions).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#C00000'>b: here it =
is only discussed for RTP (not for the data channel, but I will discuss =
that in next email responding to Magnus.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#C00000'>c: you have =
hesitations how to find and pick out the traffic information from the =
RTP extension header when RCF5285 is used for this purpose, but that is =
trivial, especially when &#8220;in-band&#8221; in a known TURN-flow. See =
further below.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(D=
ISCUSS/MALICE) by allowing the application/browser to transfer relevant =
real-time traffic information (types and bandwidth) into every RTP =
package by using the already IETF standardized RTP extension header RCF =
5285, which could be used by any current or future Internet (including =
OTT) network where WebRTC real-time traffic may flow. It seems to have =
been standardized already 2008 to allow such usage.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><br><br><o:p></o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>wh=
ether step D) can obsolete step C) (DISCUSS/MALICE) by allowing the =
application/browser to transfer relevant real-time traffic information =
(types and bandwidth) into every RTP package by using the already IETF =
standardized RTP extension header RCF 5285, which could be used by any =
current or future Internet (including OTT) network where WebRTC =
real-time traffic may flow. It seems to have been standardized already =
2008 to allow such usage.</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div></div><div><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:blue'>[P=E5l] </span><span =
lang=3DEN-US>draft-martinsen-tram-discuss is intended to solve internal =
application stream prioritisation pr flow, i.e. it is more important for =
me to get the video stream across than my application stream at this =
moment in time. </span>Its intention is to describe the 5-topple the =
STUN message was sent on.&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Moving to a pr packet prioritisation is interesting as =
it can solve the bundle problem, and mark the relative importance of =
specific video frames (I-Frames for example). Since its pr packet the =
network element might not keep stream state so it might be easier that =
way. But the information needs to be in a fixed offset for quick network =
element processing. And we want to describe other streams than just RTP. =
That makes RTP header extensions =
unsuitable.&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Qo=
S related step C)<span =
class=3Dapple-converted-space>&nbsp;</span></span><span =
lang=3DEN-US>draft-martinsen-tram-discuss<span =
class=3Dapple-converted-space>&nbsp;</span></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ca=
n then be taken out of TRAM, and step D) (that was never intended to be =
handled by TRAM anyway) will be handled =
elsewhere.</span><o:p></o:p></p></div></div></blockquote><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:blue'>[P=E5l] =
</span><span lang=3DEN-US>I was hoping to argue that the draft is only =
new signalling of old concepts (DSCP and ECN). </span>In the end it is =
the WG that decides if there is enough =
interest.&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
have repeated my suggestion to RTCWEB WG from the October discussions on =
the relevant<span =
class=3Dapple-converted-space>&nbsp;</span></span><span =
lang=3DEN-US>[rtcweb] [avtext] [mmusic]</span><span =
class=3Dapple-converted-space><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>li=
sts<span class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
"><span =
style=3D'color:purple'>http://www.ietf.org/mail-archive/web/rtcweb/curren=
t/msg09129.html</span></a><span =
class=3Dapple-converted-space>&nbsp;</span>to introduce the QoS related =
step D) into<span =
class=3Dapple-converted-space>&nbsp;</span></span><span =
lang=3DEN-US>draft-ietf-rtcweb-rtp-usage s</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ug=
gesting usage of RFC5285, where traffic information in RTP packets =
simply would be filled by any browser under any OS.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
thout step B),<span class=3Dapple-converted-space>&nbsp;</span>TRAM<span =
class=3Dapple-converted-space>&nbsp;</span>&#8220;Enforcing the =
real-time traffic through the offered/discovered TURN servers&#8221;, =
the real-time traffic may happen through the IP default gateways often =
congested by data traffic and QoS insensitive streaming video and file =
sharing. Without specific QoS methods at those points, the network raw =
bandwidth capacity may have to be 10-folded to achieve sufficient QoE =
when WebRTC usage becomes popular. Methods like DISCUSS addresses such =
things occurring when STUN instead TURN is used (by allowing flows into =
such congestion points), but only provides direct traffic info for =
outgoing (not incoming) real-time traffic and locally, therefore are not =
even applicable to reservation type of networks like Cable Networks and =
Mobile OTT.</span><o:p></o:p></p></div></div></blockquote><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:blue'>[P=E5l] =
</span><span lang=3DEN-US>If both clients supports =
draft-martinsen-tram-discuss you would get an bi-directional end to end =
description of the flow that works through =
NAT.<o:p></o:p></span></p></div><div><p class=3DMsoNormal>The =
information in the STUN packet can be used to remark DSCP bits to =
whatever values that is of best use for the network the packet is =
flowing though.<span =
style=3D'color:blue'><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#C00000'>[Karl] =
Again, this is only applicable for all-diffserve networks and you cannot =
even route the flow to a quality pipe, not even a duplicate pipe that is =
not the default gateway surf pipe (which is a very common method used =
for VoIP).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:blue'>[P=E5l] </span><span lang=3DEN-US>I =
dont see how TURN gives that ability. <span =
style=3D'color:blue'><o:p></o:p></span></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#C00000'>[Karl - =
???] Haven&#8217;t we many times agreed that you can add exactly the =
same STUN-attributes to the TURN allocate request AND in addition to =
that, control where the flow goes?<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#C00000'>BUT THIS IS =
ALSO INSUFFICIENT. <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:#C00000'>I discarded such (similar) =
interface to QoS already October 22<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
><a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
"><span =
style=3D'color:#C00000'>http://www.ietf.org/mail-archive/web/rtcweb/curre=
nt/msg09129.html</span></a> <o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>&#8230;<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>PS Microsoft seems to have done work in this field, defining a =
proprietary attribute &#8220;MS Service Quality&#8221;; =
<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>However that seems to apply to the TURN server allocation request and =
would therefore:<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>--- Apply to the whole UDP flow, and could not be set for each stream =
individually (with different requirements), and<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>--- <span style=3D'background:aqua;mso-highlight:aqua'>Does not handle =
the bandwidth requirement for incoming real-time traffic (required to =
reserve in RSVP type of networks)</span><o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>However the quality attributes conveyed and their encoding may=A0 be =
considered.<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>This is 2.2.2.19 MS-Service Quality Attribute from =
<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
><a =
href=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).a=
spx"><span =
style=3D'color:#C00000'>http://msdn.microsoft.com/en-us/library/cc431507(=
v=3Doffice.12).aspx</span></a> <o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>&#8230;<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:blue'>[P=E5l] </span><span lang=3DEN-US>Once the channel =
is bound you only have a channel number in front of the UDP payload. =
</span>And TURN is designed to carry alls sort of traffic. For this to =
work there must be some sort of only webRTC traffic is allowed through =
this TURN server. (And Ill bet bitorrent or others find a way to mimic =
webRTC to set up only a data channel..)<o:p></o:p></p></div><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
>[Karl] I owe Simon a response to misuse of prioritized channels &#8211; =
will do later after first collecting these diversions. </span><span =
lang=3DEN-US style=3D'color:#C00000'>This has nothing to do with STUN vs =
TURN! TURN just gives the addition of ALSO directing the flow. In both =
cases it is insufficient to transfer traffic info in the STUN/TURN =
attributes.<br></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
at would leave certain network types to investing in raw bandwidth =
increase, instead of simply borrowing the bandwidth (from much larger =
data traffic and QoS insensitive streaming video and file sharing at no =
adverse effect) and making that borrowed no-cost bandwidth very valuable =
for the carriers when delivering to their =
customers.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
know a few of you Cisco guys are among the ones that have been fighting =
against the general &#8220;it is all about bandwidth&#8221; and =
&#8220;it will go away with time&#8221;-attitude within IETF work, e.g. =
see<span class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html=
"><span =
style=3D'color:purple'>http://www.ietf.org/mail-archive/web/rtcweb/curren=
t/msg09031.html</span></a>, but the pointers given in the current QoS =
discussion within TRAM:</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
) will *<b>not allow</b>* ISP&#8217;s to use already available and =
currently deployable quality IP pipes for real-time traffic to also be =
used for WebRTC generated real-time =
traffic.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
i) will *<b>not allow</b>* some network types (e.g. Cable Networks and =
Mobile OTT) to borrow bandwidth from data traffic and QoS insensitive =
streaming video and file sharing. All networks will *<b>also be =
inhibited</b>* from the very common and used method of simply providing =
an extra IP pipe (often provided over the same wire but level-2 =
separated) dedicated for real-time =
usage</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(<=
b>*by resisting TRAM implementation of step =
B</b>*)</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ne=
wer, fiber only type of networks, can still borrow bandwidth at no extra =
cost, *<b>by proprietary usage</b>* of RFC5285 by browsers.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
ii) will leave most ISPs to *<b>only use</b>* raw bandwidth capacity =
increase that may has to be 10-folded to reach sufficient QoE when =
WebRTC usage becomes popular (if at all possible, since unmanaged IP =
pipes intermittently are filled, whatever bandwidth is =
available).</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fu=
rther explainations about QoS methods and their implementation were =
given a few days ago in:</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00274.html">=
<span =
style=3D'color:purple'>http://www.ietf.org/mail-archive/web/tram/current/=
msg00274.html</span></a></span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(i=
), (ii) and (iii) will *<b>seriously delay</b>* usage of telepresence =
capable and quality demanding WebRTC (bandwidth being around 30 times =
higher over best effort Internet, than telephony over quality managed =
networks) and will *<b>vastly increase cost</b>* for ISPs to offer =
WebRTC with good QoE.</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Be=
low you are giving pointers saying &#8220;</span><span lang=3DEN-US>They =
are currently working on a problem-statement draft and a use-case draft, =
any input to those would be very helpful.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221; Those are unnecessary, risking leading to (i), (ii) and (iii) =
above.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div></div></blockquote><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:blue'>[P=E5l] =
</span><span lang=3DEN-US>We find them quite useful. </span>It allows us =
to engage with potential users and see if there are real problems to =
solve. If the customer/users do not see a problem there is little =
incentive for us to invest in creating a solution that nobody wants. =
That said, it is always the danger of crating a to complicated solution =
that solves to many problems. &nbsp;<o:p></o:p></p></div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#C00000'>[Karl] This =
is QoS stuff. Let&#8217;s stick to &#8220;Interfacing to QoS&#8221; and =
with a general interface, that the &#8220;QoS device including the TURN =
server&#8221; should be able to do whatever network QoS-method you =
arrive at.<br><br><o:p></o:p></span></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>So=
 are pointers such as: &#8220;</span><span lang=3DEN-US>RMCAT where =
congestion avoidance for RTP is being developed</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221;. TRAMs Milestone 3 is for the purpose of directing real-time =
traffic where congestion control isn&#8217;t already in place. Using =
RFC5285 available since 2008, to fill traffic information in RTP packets =
(step D)) is probably a necessity for most future &#8220;congestion =
control&#8221; discussed.</span><span =
lang=3DEN-US><o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span lang=3DEN-US><o:p></o:p></span></p></div></div><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:blue'>[P=E5l] =
</span><span lang=3DEN-US>There are some good example of RTP header =
extension usage. </span>For =
example&nbsp;draft-avtext-berger-framemarking. But that information may =
very well be encrypted and unavailable to the network =
element.<o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:#C00000'>[Karl] You (P=E5l), Colin and =
Magnus seem to worry about trivial things when using the RTP header =
extension, for transferring traffic information from the application =
(here browser) to the network level. Don&#8217;t worry! </span><span =
lang=3DEN-US style=3D'font-family:Wingdings;color:#C00000'>J</span><span =
lang=3DEN-US style=3D'color:#C00000'> =A0These are really trivial things =
:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:#C00000'>- If we select to use the RTP extension header =
RCF5285 for this purpose, it should of course NOT use RFC6904 =
encryption, since it is meant for network level =
reading/usage!<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:#C00000'>- Finding the RTP header extension =
(especially when &#8220;in-band&#8221; in a known TURN-flow) is really =
trivial. =A0<o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:#C00000'>Linking each RTP-flow by its SSRC =
and sequence number, and picking exactly the right traffic type and =
bandwidth parameters is no problem (I am not aware of any &#8220;Zero =
CPU&#8221; DPI device that needs to be considered &#8211; But if we can =
ease the job, we should.)<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'color:#C00000'>This link shows much more advanced things are =
available in Cisco devices: <o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'color:#C00000'><a =
href=3D"http://www.cisco.com/en/US/products/ps6616/products_white_paper09=
186a0080110040.shtml#wp39132">http://www.cisco.com/en/US/products/ps6616/=
products_white_paper09186a0080110040.shtml#wp39132</a> =
<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'color:#C00000'>and the ADSL 2+ modem doing a lot of advanced =
QoS-stuff, described in <o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#C00000'=
><a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html</a></=
span><span lang=3DEN-US style=3D'color:#C00000'> =
<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span lang=3DEN-US =
style=3D'color:#C00000'>is our Intertex/Ingate IX78, just powered by a 6 =
USD 264 MHz ARM ADSL chip set CPU and we have implemented a check box TO =
REMOVE the RTP Header extension (for the ugly reason that the =
ITSP&#8217;s billing system requires G.711 VoIP packets of a certain =
length) &#8211; It does not take noticeable CPU power to do this at 100 =
Mbps wire speed! <o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:1.0cm'><span style=3D'color:#C00000'><img =
border=3D0 width=3D341 height=3D229 id=3D"Bild_x0020_1" =
src=3D"cid:image001.png@01CF3D69.B8EE2930"></span><span lang=3DEN-US =
style=3D'color:#C00000'><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:#C00000'>/Karl =
<o:p></o:p></span></p></div><p =
class=3DMsoNormal>.-.<o:p></o:p></p></div><div><p =
class=3DMsoNormal>P=E5l-Erik<o:p></o:p></p></div><div><p =
class=3DMsoNormal><br><br><o:p></o:p></p><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
is chicken-and-egg circular also applies to RTCWEB direction of QoS =
issues to:</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://datatracker.ietf.org/doc/draft-dhesikan-tsvwg-rtcweb-qos">=
<span =
style=3D'color:purple'>http://datatracker.ietf.org/doc/draft-dhesikan-tsv=
wg-rtcweb-qos</span></a></span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>fo=
cusing on DSCP-mapping without even mentioning RFC 5285 available since =
2008, to fill traffic information in RTP packets where it &nbsp;is a =
necessity and will allow:</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
) applications to directly convey QoS related real-time traffic info to =
the network at points where RTP flow is directed to by TRAM Milstone 3, =
to be used by *<b>any network element implementing any suitable QoS =
methods for the particular network</b>* =
for</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
) *<b>all</b>* WebRTC browsers *<b>and</b>* dedicated clients (not using =
WebRTC), under *<b>all</b>* OSs, and *<b>all</b>* current and future IP =
networks</span><o:p></o:p></p></div><div><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without*<span =
class=3Dapple-converted-space>&nbsp;</span></span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to be forced into application specific networks (PSTN, IMS) instead =
of using the Internet (including =
OTT).</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e only activity required, is to call for ISPs&#8217; review whether the =
traffic information transferred by RFC 5285 is sufficient for current =
and future needs in their network as suggested in<span =
class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
"><span =
style=3D'color:purple'>http://www.ietf.org/mail-archive/web/rtcweb/curren=
t/msg09129.html</span></a></span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;snip&gt;<=
/span><o:p></o:p></p></div><div><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&#8230;two =
parameters (e.g. two bytes each) are encoded into the RTP header =
extension:</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p></div><div><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A) The =
maximum bandwidth requirement: Two bytes could contain everything from =
some bps for real-time text to Gbps for future 3D supersize =
telepresence&#8230; on a logarithmic =
scale.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p></div><div><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B) The =
quality characteristics for the stream, with the highest bit set to 1, =
we could allocate a bit each for quality type =
e.g:</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Best Effort, =
Audio, Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. =
video streaming), Minimum Delay, Reliable Delivery, Prioritize X, =
Variation Y, that could be combined as required to describe the =
stream.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p></div><div><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And with =
highest bit set to 0, there could instead be a number for special usage =
that does not fit the general description of the individual =
bits.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;/snip&gt;=
</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
en this could be assigned numbers to have the RFC in =
place.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th TRAM milestone 3 also place,</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
rket forces will drive ISPs and browser makers to implement just that, =
without even having it =
MUST-established.</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who does not want a &#8220;WebRTC-Ready&#8221; Internet =
access?&#8221; and</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?&#8221; and vice =
versa.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span>l<o:p></o:p></p></div></div></div><div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see further emails soon following this one, for details and =
history.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p></div><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><div><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span class=3Dapple-converted-space><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>&nbsp;</span=
></span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>tram [<a =
href=3D"mailto:tram-bounces@ietf.org"><span =
style=3D'color:purple'>mailto:tram-bounces@ietf.org</span></a>]<span =
class=3Dapple-converted-space>&nbsp;</span><b>F=F6r<span =
class=3Dapple-converted-space>&nbsp;</span></b>Pal Martinsen =
(palmarti)<br><b>Skickat:</b><span =
class=3Dapple-converted-space>&nbsp;</span>den 21 februari 2014 =
10:37<br><b>Till:</b><span class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:tram@ietf.org">tram@ietf.org</a><br><b>Kopia:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Karl Stahl; Oleg Moskalenko; =
Alan Johnston; Yoakum, John H (John)<br><b>=C4mne:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Re: [tram] I-D Action: =
draft-thomson-tram-turn-bandwidth-00.txt</span><o:p></o:p></p></div></div=
></div><div><p class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><div><p =
class=3DMsoNormal>Hi,<o:p></o:p></p></div></div><div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div></div><div><div><p =
class=3DMsoNormal>I agree the full QoS discussion should _not_ happen in =
TRAM. If you are interested in helping out in that area I suggest you =
looking into the AEON mailing list at:&nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/aeon"><span =
style=3D'color:purple'>https://www.ietf.org/mailman/listinfo/aeon</span><=
/a><span class=3Dapple-converted-space>&nbsp;</span>. They are currently =
working on a problem-statement draft and a use-case draft, any input to =
those would be very helpful. (<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-use-cases-01"><span =
style=3D'color:purple'>http://tools.ietf.org/html/draft-eckel-aeon-use-ca=
ses-01</span></a>,&nbsp;<a =
href=3D"http://tools.ietf.org/html/draft-eckel-aeon-problem-statement-00"=
><span =
style=3D'color:purple'>http://tools.ietf.org/html/draft-eckel-aeon-proble=
m-statement-00</span></a>).<o:p></o:p></p></div></div><div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div></div><div><div><p =
class=3DMsoNormal>That said, STUN have a few nice characteristics that =
makes it a perfect candidate for transporting some of the QoS =
information. &nbsp;IMHO that would be extending the STUN spec and should =
be within the TRAM charter. &nbsp;The main goal =
of&nbsp;draft-martinsen-tram-discuss was to show how already existing =
QoS mechanisms could be transported with STUN to provide more value, and =
to start the discussion if TRAM is the appropriate place to have those =
on the wire format discussions.<o:p></o:p></p></div></div><div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div></div><div><div><p =
class=3DMsoNormal>.-.<o:p></o:p></p></div></div><div><div><p =
class=3DMsoNormal>P=E5l-Erik<o:p></o:p></p></div></div></div></blockquote=
></div></div></body></html>
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'Colin Perkins'" <csp@csperkins.org>, "'Pal Martinsen \(palmarti\)'" <palmarti@cisco.com>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com>	<9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>	<07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se>	<07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se>	<07e401ceb713$bef87a60$3ce96f20$@stahl@intertex.se>	<524AB730.7040809@ericsson.com>	<00b001cebfc1$ba8e89e0$2fab9da0$@stahl@intertex.se>	<525272E8.5050300@ericsson.com>	<050801cec3f6$6172aec0$24580c40$@stahl@intertex.se>	<5253E5EB.8030608@alvestrand.no> <02bf01cecf34$35e174a0$a1a45de0$@stahl@intertex.se> <04dd01cf31ad$0fe62d00$2fb28700$@stahl@intertex.se> <580B467D-4679-4DE1-96DE-CA37DE755563@csperkins.org> <052e01cf31cb$5311a0a0$f934e1e0$@stahl@intertex.se> <530C56CD.3010003@ericsson.com>
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For the =93mission to bring quality to real-time traffic over our best =
effort
Internet=94 I have started
<http://www.ietf.org/mail-archive/web/tram/current/msg00331.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html

[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff [1] [2]

=20

Hi Magnus,

=20

With the response just sent to P=E5l
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html, =
skipping
all the QoS methods and functions that really does not belong here when
creating an =93Interface to QoS=94, I only see the suggestion of =93data =
channel
information=94, i.e. also marking each data channel packet with traffic
bandwidth and type information  to be relevant and interesting. I =
checked
with one of our developers and I think the same traffic information as =
for
RTP can be transferred in data channel packets as follows:

=20

The RTP header has the following format (RFC 3550):

    0                   1                   2                   3

    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |V=3D2|P|X|  CC   |M|     PT      |       sequence number         |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |                           timestamp                           |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |           synchronization source (SSRC) identifier            |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            contributing source (CSRC) identifiers             |

   |                             ....                              |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            RFC5285 header extension ...                      |

   |     2 bytes Max Bandwidth     |   2 bytes Traffic Type        |

   |                             ....                              |

RTP Payload (often encrypted)

=20

A Data Channel header could have the following format:

    0                   1                   2                   3

    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |   Data Channel Magic Cookie   |      (sequence number)        |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |                           timestamp                           |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |           synchronization source (SSRC) identifier            |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            contributing source (CSRC) identifiers             |

   |                             ....                              |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            RFC5285 header extension ...                      |

   |     2 bytes Max Bandwidth     |   2 bytes Traffic Type        |

   |                             ....                              |

  DTLS Header 13 bytes

  SCTP

=20

The RFC5285 header extension is not in a fixed position in any of these
cases, but still trivial to find (especially when in  a TURN flow). This
could be introduced in
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-5=20

by making room for these bytes after the UDP header, but before the DTLS
header.

=20

What do you think?

=20

Please apply the thought that we must allow ISPs/NSPs to route the =
WebRTC
media to where it can be handled, by using TURN servers. A TURN server =
in a
QoS-box can then use the traffic info (max bandwidth and traffic type) =
found
in the extension header, to do whatever QoS method or function that is
appropriate for the specific network type (none discriminated).=20

=20

Regarding what to encode, I have several times suggested, e.g.

already October 22
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=20

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each for quality type e.g: Best Effort, =
Audio,
Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. video
streaming), Minimum Delay, Reliable Delivery, Prioritize X(2 bits),
Variation Y(2 bits), that could be combined as required to describe the
stream.

And with the highest bit set to 0, there could instead be a number for
special usage that does not fit the general description of the =
individual
bits.

=20

Please provide what more you see required (for any QoS method/function =
for
any network type)!

=20

/Karl

=20

Footnotes:

=20

[1] Please do not divert or confuse this with QoS methods in themselves
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the
interface from the application to the network level, where all networks
types should be able to use the traffic information for QoS methods =
relevant
to the particular network.

=20

[2] Here it is about recreation of the idea/intention of the RTP payload
type (PT) header that is not available for the network level anymore, by
instead having the application/browser filling the RTP header extension =
with
relevant traffic info that all IP network types can use. Here, similar
traffic type marking is suggested for the data channel.

=20

=20

-----Ursprungligt meddelande-----
Fr=E5n: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Skickat: den 25 februari 2014 09:40
Till: Karl Stahl
Kopia: rtcweb@ietf.org; tram@ietf.org
=C4mne: Re: SV: [rtcweb] [tram] Payload Types assignments

=20

Karl,

(As individual)

=20

I also wish for more usable QoS mechanisms. However, I don't see that =
being
achieved by your proposal. In addition I agree with Colin about the =
issues
with putting the information in the RTP header extension. I would also =
note
that this would be very RTP specific, and not at all help with the data
channels multiple streams and their priorities. There might be data =
channel
information that is more crucial than any of the RTP media stream =
packets.

=20

You are pushing for a small piece in the middle. A piece that will not =
help
with the more general issue of QoS.=20

--- Not at all! I am pushing for "Interfacing to QoS" rather mixing the
application layer directly to lower level QoS methods like setting DSCP =
bits
etc..

=20

How does the application and the multiple ISPs that carries the traffic
reach an agreement on what properties that can be provided, that the
application in this instance have the right to request those properties =
and
that any cost is correctly associated with the user or the user's agent =
in
regards to carrying the associated cost.

=20

When it comes to setting DSCP from user land in the OS, that is =
restricted
due to the security implications. If those implications where resolved, =
then
OS could open up those interfaces.  There are many interlinked reasons =
why
things look like they do today. I don't believe in tugging on a single
random thread in ball of yarn and hope that it comes out without any =
knots
and ties on it.

=20

Show me the framework for the QoS functions you have in mind that at =
least
has less issues than the currently deployed and take care of at least =
some
of the bigger issues. If that requires information in the RTP streams =
for
some reasons, then let us talk about how to best encoded it.

But, we need that framework first, the architecture that makes this a =
better
solution than the current Diffserv architecture or any other QoS
architecture.

=20

Cheers

=20

Magnus Westerlund

=20

----------------------------------------------------------------------

Services, Media and Network features, Ericsson Research EAB/TXM

----------------------------------------------------------------------

Ericsson AB                 | Phone  +46 10 7148287

F=E4r=F6gatan 6                 | Mobile +46 73 0949079

SE-164 80 Stockholm, Sweden | mailto:
<mailto:magnus.westerlund@ericsson.com> magnus.westerlund@ericsson.com

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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
r the</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:blue'> </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;</span><span lang=3DEN-US>mission to bring quality to real-time =
traffic over our best effort Internet&#8221;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;I have started </span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00331.html">=
<span lang=3DEN-US =
style=3D'color:blue'>http://www.ietf.org/mail-archive/web/tram/current/ms=
g00331.html</span></a></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
[tram] [rtcweb] The way to &quot;Interfacing to QoS&quot;, A level 3-5 =
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>[=
1] [2]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 Magnus,<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th the response just sent to P=E5l <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html</a>, =
skipping all the QoS methods and functions that really does not belong =
here when creating an &#8220;Interface to QoS&#8221;, I only see the =
suggestion of &#8220;</span><span lang=3DEN-US>data channel =
information</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221;, i.e. also marking each data channel packet with traffic bandwidth =
and type information =A0to be relevant and interesting. I checked with =
one of our developers and I think the same traffic information as for =
RTP can be transferred in data channel packets as =
follows:<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>The RTP header has the following format (RFC =
3550):<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp;&nbsp; =
0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 3 =
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; |V=3D2|P|X|&nbsp; CC&nbsp;&nbsp; =
|M|&nbsp;&nbsp;&nbsp;&nbsp; PT&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; sequence =
number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp; =
timestamp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
synchronization source (SSRC) =
identifier&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; |<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+<o:p></o:p></span=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
contributing source (CSRC) =
identifiers&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp; |<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+<o:p></o:p></span=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
RFC5285 header extension =
...&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp; 2 bytes =
Max Bandwidth&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; 2 bytes Traffic =
Type&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>RTP Payload (often =
encrypted)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:black'>&nbsp;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>A Data Channel header could have the following =
format:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp;&nbsp; =
0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 3 =
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; |&nbsp;&nbsp; Data Channel Magic =
Cookie&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(sequence =
number)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp; =
timestamp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o=
:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
synchronization source (SSRC) =
identifier&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; |<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+<o:p></o:p></span=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
contributing source (CSRC) =
identifiers&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp; |<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+<o:p></o:p></span=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
RFC5285 header extension =
...&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp; 2 bytes =
Max Bandwidth&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; 2 bytes Traffic =
Type&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp; DTLS Header 13 =
bytes<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Courier =
New","serif";color:black'>&nbsp; SCTP<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'color:black'>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e RFC5285 header extension is not in a fixed position in any of these =
cases, but still trivial to find (especially when in=A0 a TURN flow). =
This could be introduced in <a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#sect=
ion-5">http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#secti=
on-5</a> <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>by=
 making room for these bytes after the UDP header, but before the DTLS =
header.<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wh=
at do you think?<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease apply the thought that we must allow ISPs/NSPs to route the WebRTC =
media to where it can be handled, by using TURN servers. A TURN server =
in a QoS-box can then use the traffic info (max bandwidth and traffic =
type) found in the extension header, to do whatever QoS method or =
function that is appropriate for the specific network type (none =
discriminated). <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Re=
garding what to encode, I have several times suggested, =
e.g.<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>al=
ready October 22 <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<span lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A) The =
maximum bandwidth requirement: Two bytes could contain everything from =
some bps for real-time text to Gbps for future 3D supersize =
telepresence&#8230; on a logarithmic scale.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<span lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B) The =
quality characteristics for the stream, with the highest bit set to 1, =
we could allocate a bit each for quality type e.g: Best Effort, Audio, =
Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. video =
streaming), Minimum Delay, Reliable Delivery, Prioritize X(2 bits), =
Variation Y(2 bits), that could be combined as required to describe the =
stream.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And with the =
highest bit set to 0, there could instead be a number for special usage =
that does not fit the general description of the individual =
bits.<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease provide what more you see required (for any QoS method/function for =
any network type)!<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
otnotes:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[1=
] Please do not divert or confuse this with QoS methods in themselves =
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the interface from the application to the network level, where all =
networks types should be able to use the traffic information for QoS =
methods relevant to the particular network.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[2=
] Here it</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
is about recreation of the idea/intention of the RTP payload type (PT) =
header that is not available for the network level anymore, by instead =
having the application/browser filling the </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>RT=
P header extension with relevant traffic info that all IP network types =
can use. Here, similar traffic type marking is suggested for the data =
channel.<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText>-----Ursprungligt meddelande-----<br>Fr=E5n: Magnus =
Westerlund [mailto:magnus.westerlund@ericsson.com] <br>Skickat: den 25 =
februari 2014 09:40<br>Till: Karl Stahl<br>Kopia: rtcweb@ietf.org; =
tram@ietf.org<br>=C4mne: Re: SV: [rtcweb] [tram] Payload Types =
assignments<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>Karl,<o:p></o:p></p><p class=3DMsoPlainText>(As =
individual)<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>I also =
wish for more usable QoS mechanisms. However, I don't see that being =
achieved by your proposal. In addition I agree with Colin about the =
issues with putting the information in the RTP header extension. I would =
also note that this would be very RTP specific, and not at all help with =
the data channels multiple streams and their priorities. There might be =
data channel information that is more crucial than any of the RTP media =
stream packets.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>You =
are pushing for a small piece in the middle. A piece that will not help =
with the more general issue of QoS. <o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>--- Not at all! I am pushing for =
&quot;Interfacing to QoS&quot; rather mixing the application layer =
directly to lower level QoS methods like setting DSCP bits =
etc..<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>How does the application and the multiple ISPs that carries =
the traffic reach an agreement on what properties that can be provided, =
that the application in this instance have the right to request those =
properties and that any cost is correctly associated with the user or =
the user's agent in regards to carrying the associated =
cost.<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>When it comes to setting DSCP from user land in the OS, =
that is restricted due to the security implications. </span>If those =
implications where resolved, then OS could open up those interfaces.=A0 =
There are many interlinked reasons why things look like they do today. I =
don't believe in tugging on a single random thread in ball of yarn and =
hope that it comes out without any knots and ties on =
it.<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>Show me the framework for the QoS functions you =
have in mind that at least has less issues than the currently deployed =
and take care of at least some of the bigger issues. If that requires =
information in the RTP streams for some reasons, then let us talk about =
how to best encoded it.<o:p></o:p></p><p class=3DMsoPlainText>But, we =
need that framework first, the architecture that makes this a better =
solution than the current Diffserv architecture or any other QoS =
architecture.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>Cheers<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Magnus =
Westerlund<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>----------------------------------------------------=
------------------<o:p></o:p></p><p class=3DMsoPlainText>Services, Media =
and Network features, Ericsson Research EAB/TXM<o:p></o:p></p><p =
class=3DMsoPlainText>----------------------------------------------------=
------------------<o:p></o:p></p><p class=3DMsoPlainText>Ericsson =
AB=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 | Phone=A0 +46 10 =
7148287<o:p></o:p></p><p class=3DMsoPlainText>F=E4r=F6gatan =
6=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 | Mobile +46 73 =
0949079<o:p></o:p></p><p class=3DMsoPlainText>SE-164 80 Stockholm, =
Sweden | mailto: <a href=3D"mailto:magnus.westerlund@ericsson.com"><span =
style=3D'color:windowtext;text-decoration:none'>magnus.westerlund@ericsso=
n.com</span></a><o:p></o:p></p><p =
class=3DMsoPlainText>----------------------------------------------------=
------------------<o:p></o:p></p></div></body></html>
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Colin Perkins'" <csp@csperkins.org>
References: <C5E08FE080ACFD4DAE31E4BDBF944EB11667BBA0@xmb-aln-x02.cisco.com>	<5238446D.8050700@ericsson.com>	<9F33F40F6F2CD847824537F3C4E37DDF17BCF581@MCHP04MSX.global-ad.net>	<07a601ceb64e$5caaba00$16002e00$@stahl@intertex.se>	<07b001ceb65f$ce3f0cf0$6abd26d0$@stahl@intertex.se>	<07e401ceb713$bef87a60$3ce96f20$@stahl@intertex.se>	<524AB730.7040809@ericsson.com>	<00b001cebfc1$ba8e89e0$2fab9da0$@stahl@intertex.se>	<525272E8.5050300@ericsson.com>	<050801cec3f6$6172aec0$24580c40$@stahl@intertex.se>	<5253E5EB.8030608@alvestrand.no> <02bf01cecf34$35e174a0$a1a45de0$@stahl@intertex.se> <04dd01cf31ad$0fe62d00$2fb28700$@stahl@intertex.se> <580B467D-4679-4DE1-96DE-CA37DE755563@csperkins.org> <052e01cf31cb$5311a0a0$f934e1e0$@stahl@intertex.se> <D06C438A-8894-402C-AE9F-D7787ECF77B3@csperkins.org>
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Subject: Re: [rtcweb] [tram] Payload Types assignments
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For the =93mission to bring quality to real-time traffic over our best =
effort
Internet=94 I have started
<http://www.ietf.org/mail-archive/web/tram/current/msg00331.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html

[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff [1] [2]

=20

Hi Colin,

=20

Please see the just sent email to P=E5l and Magnus where I address your
concerns.=20

http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html

http://www.ietf.org/mail-archive/web/rtcweb/current/msg11847.html

=20

/Karl

=20

Footnotes:

=20

[1] Please do not divert or confuse this with QoS methods in themselves
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the
interface from the application to the network level, where all networks
types should be able to use the traffic information for QoS methods =
relevant
to the particular network.

=20

[2] Here it is about recreation of the idea/intention of the RTP payload
type (PT) header that is not available for the network level anymore, by
instead having the application/browser filling the RTP header extension =
with
relevant traffic info that all IP network types can use. Similar traffic
type marking is suggested for the data channel.

=20

Fr=E5n: Colin Perkins [mailto:csp@csperkins.org]=20
Skickat: den 26 februari 2014 00:49
Till: Karl Stahl
Kopia: rtcweb@ietf.org; Magnus Westerlund; tram@ietf.org; Harald =
Alvestrand
=C4mne: Re: [rtcweb] [tram] Payload Types assignments

=20

Karl,

=20

I sympathise with your goal, but I really do not think RTP header =
extensions
are appropriate for this purpose.

=20

Ignoring the semantic mismatch, in order to identify an RTP header
extension, you need access to the signalling data, and if you have =
access to
the signalling, you can signal the QoS parameters without using RTP.=20

=20

You state that only the payload is encrypted in SRTP. That is not
necessarily true. RTP header extensions can be encrypted when RFC 6904 =
is in
use, and rtcweb-rtp-usage draft recommends this be done in some cases.=20

=20

The signalling channel is also likely encrypted end-to-end in new
applications, so making it difficult to extract the information you need =
to
parse the RTP header extension, even if it is unencrypted.=20

=20

Besides these focussed issues, I would also urge you to consider the =
much
broader comments Magnus made. The QoS problem is broad, and a point =
solution
based on RTP header extensions - even if it were workable, which I doubt =
-
would address only a small part of the problem space.

=20

=20

=20

Colin

=20

=20

=20

=20

On 25 Feb 2014, at 01:45, Karl Stahl <karl.stahl@intertex.se> wrote:

Colin,

=20

If the below were the case, it would be =93DPI guesswork=94 that I also =
advice
against. RTP doesn=92t even have unique protocol header within UDP =96 =
it can
even be confused with other UDP payload.=20

=20

However, if the RTP is captured in a TURN-flow, as in TRAM Milestone 3, =
the
network point that this flow is directed to and can apply QoS methods
relevant to the network (which is not diffserve in Mobile OTT and Cable
Networks) has not a too difficult tasks. Linking each RTP-flow by its ID =
and
sequence number, and picking exactly the right traffic type and =
bandwidth
parameters is doable (see inline below, we at Ingate do it already)!

=20

Also DiffServ DSCP-bits are seldom maintained crossing network =
boundaries,
thus carrying no relevant information at the receiving end, while the =
RTP
extension header remains unchanged end-to-end. In DSCP-bits, there is no
bandwidth requirement information when entering networks requiring
reservation. That is always (and dynamically set) available in the RTP
extension header for each packet.

=20

And, I am sure you are aware of the difficulties of getting DSCP-bits
through OS sockets, which is even worse with multiple streams over the =
same
UDP-port

=20

Further, defining the QoS RTP extension header as in RFC5285, does not =
in
anyway conflict with other RTP extension headers or DSCP transfer or
settings.

=20

/Karl

=20

Fr=E5n: Colin Perkins [mailto:csp@csperkins.org]=20
Skickat: den 24 februari 2014 23:52
Till: Karl Stahl
Kopia: rtcweb@ietf.org; Magnus Westerlund; tram@ietf.org; Harald =
Alvestrand
=C4mne: Re: [rtcweb] [tram] Payload Types assignments

=20

Karl,

=20

I strongly disagree with this suggestion. An RTP header extension, =
located
at an unknown and variable offset into a packet that does not have a
well-defined magic number in the header,=20

This is how to find the traffic info:

   The payload of the classifier header extension element can be encoded

   using either the one-byte or two-byte header defined in [rfc5285
<http://tools.ietf.org/html/rfc5285> ] and

   shown below in Figure 1 and 2 below.

=20

      0                   1                   2                   3

      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

     |  ID   | len=3D1 |   Namespace   |    Value      |    0 (pad)    |

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

=20

             Figure 1: Classifier Using the One-Byte Header

=20

      0                   1                   2                   3

      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

     |      ID       |    len=3D2      |   Namespace   |    Value      |

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

=20

=20

             Figure 2: Classifier Using the Two-Byte Header

=20

indicated using a dynamically assigned identifier that is conveyed in an
out-of-band=20

--- It is the TURN flow!!! Requested by the ICE protocol for the =
specific
media to come!

and encrypted signalling channel,=20

--- Only the payload is encrypted in SRTP =96 leaving id, seq no and =
header
ext to be used as intended!

--- Thus being the appropriate place=85

is not an appropriate place to put QoS information that has to be =
processed
on a per-packet basis.=20

If you want DiffServ, you know where to find it.

--- True, but if it cannot be used=85

Colin

=20

=20

On 24 Feb 2014, at 22:09, Karl Stahl <karl.stahl@intertex.se> wrote:

I suggest to the RTCWEB WG that the below from the September and October
discussions on the relevant [rtcweb] [avtext] [mmusic] lists
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html is
introduced into draft-ietf-rtcweb-rtp-usage for usage of RFC 5285, to =
allow:

=20

(1) WebRTC applications to directly convey QoS related real-time traffic
info to the network at points where RTP flow is directed to by TRAM =
Milstone
3, to be used by *any network element implementing any suitable QoS =
methods
for the particular network* for=20

(2) *all* WebRTC browsers *and* clients, under *all* OSs, and *all* =
current
and future IP network, to achieve best QoE=20

(3) *without* having to force WebRTC into application specific networks
(such as IMS) instead of using the Internet (including OTT).

=20

The only further activity required, is to call for ISPs=92 to review =
whether
the traffic information transferred by RFC 5285 is sufficient for =
current
and future needs in their network as suggested in below repeated
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html

<snip>

=85two parameters (e.g. two bytes each) are encoded into the RTP header
extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each for quality type e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

</snip>

=20

Then this could be assigned numbers to have an RFC in place.

=20

With TRAM milestone 3 also place,=20

market forces will drive ISPs and browser makers to implement just this,
without even having it MUST-established.

=20

=93Who does not want a =93WebRTC-Ready=94 Internet access?=94 and

=93Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with
much better QoE?=94 and vice versa.

=20

Please see further emails soon following this one, for details and =
history.

=20

/Karl

=20

=20

Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Karl
Stahl
Skickat: den 22 oktober 2013 16:37
Till: 'Harald Alvestrand'; rtcweb@ietf.org; 'Magnus Westerlund'
Kopia: 'Colin Perkins'
=C4mne: [rtcweb] [avtext] Payload Types assignments was Re: SV: [mmusic] =
WGLC
of draft-ietf-rtcweb-use-cases-and-requirements-11

=20

Harald, I mostly agree with the quality requirements of different =
real-time
traffic that the WebRTC browser/application may use. But rather than =
asking
the application, let's convey the bandwidth and priority requirements to =
the
network. Just like with the Payload type (that is hard to squeeze that
information into) it must be visible to the network (and not changed by =
the
network, like diffserv bits are). Such marking must also be available =
for
incoming traffic, which is especially important in RSVP type of =
networks,
that has to reserve bandwidth for it. =20

=20

There is actually a good way to show these needs to the network (without
using the PT, or diffserv bits, which aren=92t sufficient anyway).=20

=20

Let's use the RTP header extension field that also is visible outside =
the
encrypted payload. A week ago came
http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00  that
outlines the usage of the extension field for classification of traffic!
This document does not yet outline what to put in there and how to =
encode it
though.

=20

Today's http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10 =
discusses
other webrtc usages of the RTP header extension in 5.2 (there can be =
many
header extensions according to RFC 5285) and in 9 there is "WebRTC Use =
of
RTP: Future Extensions".

=20

So, it looks obvious to use the RTP header extension to show the
characteristics and bandwidth requirements to the network. It should not
introduce any backward incompatibilities either.

=20

Such marking is done in every RTP packet so it can be set individually =
for
each stream and could even be changed during a session (e.g. when =
limiting
the bandwidth based on RTCP feedback). RFC 5286 also specifies how RTP
extension header usage can be negotiated in SDP. I think this could be
easily done by the WebRTC browser for "all current and future needs" if
properly specified now.

=20

I suggest that two parameters (e.g. two bytes each) are encoded into the =
RTP
header extension:

=20

A) The maximum bandwidth requirement: Two bytes could contain everything
from some bps for real-time text to Gbps for future 3D supersize
telepresence=85 on a logarithmic scale.

=20

B) The quality characteristics for the stream, with the highest bit set =
to
1, we could allocate a bit each quality e.g:

Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
Prioritize X, Variation Y, that could be combined as required to =
describe
the stream.

=20

And with highest bit set to 0, there could instead be a number for =
special
usage that does not fit the general description of the individual bits.

=20

Please note the totally different requirements a diffserv and an RSVP
network have to know, so let=92s put all into these bytes. (E.g. a =
diffserv
network don't need the bandwidth usage, but RSVP reservation networks =
(e.g.
cable and 3G/4G OTT) do. There one should initially reserve the maximum
bandwidth indicated, but can later re-reserve.)

=20

/Karl

=20

PS Microsoft seems to have done work in this field, defining a =
proprietary
attribute =93MS Service Quality=94;=20

However that seems to apply to the TURN server allocation request and =
would
therefore:

--- Apply to the whole UDP flow, and could not be set for each stream
individually (with different requirements), and

--- Does not handle the bandwidth requirement for incoming real-time =
traffic
(required to reserve in RSVP type of networks)

However the quality attributes conveyed and their encoding may  be
considered.

=20

This is 2.2.2.19 MS-Service Quality Attribute from=20

http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).aspx=20

=20

MS-Service Quality Attribute

The MS-Service Quality attribute is used to convey information about the
data stream that the protocol client is intending to transfer over an
allocated port. The protocol client SHOULD<21> include this attribute as
part of an Allocate request message. A TURN server SHOULD use the
information in this attribute to make decisions about resource =
allocation,
bandwidth prioritization, and data delivery methods. If the attribute is =
not
present in the Allocate request message, the TURN server SHOULD assume =
that
the data stream is audio with best effort delivery. The format of this
attribute is as follows...=20

...

The following stream types are supported in this extension. All other =
stream
types are reserved for future use.

=A7 "0x0001": Audio

=A7 "0x0002": Video

=A7 "0x0003": Supplemental Video

=A7 "0x0004": Data

Service Quality (2 bytes): The service quality level required by the
protocol client for the stream.

The following service quality levels are supported in this extension. =
All
other service quality levels are reserved for future use.

=A7 "0x0000": Best effort delivery.

=A7 "0x0001": Reliable delivery.

=20

=20

-----Ursprungligt meddelande-----

Fr=E5n: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] F=F6r =
Harald
Alvestrand

Skickat: den 8 oktober 2013 13:01

Till: rtcweb@ietf.org

=C4mne: Re: [rtcweb] Payload Types assignments was Re: SV: [mmusic] WGLC =
of
draft-ietf-rtcweb-use-cases-and-requirements-11

=20

On 10/08/2013 09:17 AM, Karl Stahl wrote:

> Hej Magnus,

>=20

>> Also, are you really interested in knowing that it is VP9 vs H.264,=20

>> isn't

> the questions this is video of this priority that is important?

>> I think you need to more carefully consider what are the goals you=20

>> try to

> achieve them.

>=20

> Actually, my concern is to get an idea of the maximum bandwidth that=20

> could be required for a WebRTC (ICE) setup media flow. Both voice and=20

> video should be prioritized over data (their individual priority is of =


> less importance as long as there is sufficient bandwidth for both).

=20

You don't know that without knowing what the application is for.

In, for instance, a shooter game with voice backchannels, the movement =
and
event information (data) is MORE time sensitive than the voice data.

=20

>=20

> With diffserv you don=92t need to know the bandwidth requirement, but=20

> with RSVP reservation (like in cable and mobile networks) you need to=20

> know how much to reserve. Voice is like 100's kbit/s, video VP8 or=20

> H.264 is like 3,5 mbps.

=20

Again, without knowing the application, you don't know that.

The application could decide to use QCIF or HD, and the bandwidth =
variation
of screencast (semi-static with sudden, large changes) is completely
different from that of a talking head, which is again completely =
different
from a high-movement scene.

=20

>=20

> To add to the complication of codec variants, the video codecs in=20

> question for WebRTC have variable bandwidth, and when there is a poor=20

> connection we see Chrome reducing the video window size to reduce the
bandwidth used...

>=20

> I think the payload type field at best can reflect a maximum bandwidth =


> to initially reserve bandwidth for, and thereafter make new=20

> reservations if the bandwidth changes during the call. So could we=20

> change RTP to show maximum bandwidth instead of payload type in that=20

> field outside the encrypted payload :) ... Or maybe that is not a =
joke?

=20

I think these ruminations only lead to one conclusion:

=20

You can't tell what the needed bandwidth is up front without asking the
application.

You can't tell what the right priority ranking is without asking the
application.

=20

If you need to know the bandwidth or the priority up front, the =
application
has to tell you. Anything else is pure heuristics.

=20

_______________________________________________

rtcweb mailing list

 <mailto:rtcweb@ietf.org> rtcweb@ietf.org

 <https://www.ietf.org/mailman/listinfo/rtcweb>
https://www.ietf.org/mailman/listinfo/rtcweb

=20

=20

--=20

Colin Perkins

http://csperkins.org/

=20

=20

=20

=20

=20

--=20

Colin Perkins

http://csperkins.org/

=20

=20

=20


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</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
r the</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:blue'> </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;</span><span lang=3DEN-US>mission to bring quality to real-time =
traffic over our best effort Internet&#8221;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;I have started </span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00331.html">=
<span =
lang=3DEN-US>http://www.ietf.org/mail-archive/web/tram/current/msg00331.h=
tml</span></a></span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
[tram] [rtcweb] The way to &quot;Interfacing to QoS&quot;, A level 3-5 =
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>[=
1] [2]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 Colin,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see the just sent email to P=E5l and Magnus where I address your =
concerns. <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html</a><o=
:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><a=
 =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11847.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11847.html</a><o=
:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
otnotes:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[1=
] Please do not divert or confuse this with QoS methods in themselves =
(like diffserve, bandwidth reservation, congestion control etc.) This is =
the interface from the application to the network level, where all =
networks types should be able to use the traffic information for QoS =
methods relevant to the particular network.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[2=
] Here it</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
is about recreation of the idea/intention of the RTP payload type (PT) =
header that is not available for the network level anymore, by instead =
having the application/browser filling the </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>RT=
P header extension with relevant traffic info that all IP network types =
can use. </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Si=
milar traffic type marking is suggested for the data =
channel.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Colin =
Perkins [mailto:csp@csperkins.org] <br><b>Skickat:</b> den 26 februari =
2014 00:49<br><b>Till:</b> Karl Stahl<br><b>Kopia:</b> rtcweb@ietf.org; =
Magnus Westerlund; tram@ietf.org; Harald Alvestrand<br><b>=C4mne:</b> =
Re: [rtcweb] [tram] Payload Types =
assignments<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Karl,<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
sympathise with your goal, but I really do not think RTP header =
extensions are appropriate for this purpose.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US>Ignoring the semantic mismatch, in =
order to identify an RTP header extension, you need access to the =
signalling data, and if you have access to the signalling, you can =
signal the QoS parameters without using RTP.&nbsp;<span =
style=3D'color:blue'><o:p></o:p></span></span></p></div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p></div><div><p =
class=3DMsoNormal>You state that only the payload is encrypted in SRTP. =
That is not necessarily true. RTP header extensions can be encrypted =
when RFC 6904 is in use, and rtcweb-rtp-usage draft recommends this be =
done in some cases. <span style=3D'color:blue'><o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>The =
signalling channel is also likely encrypted end-to-end in new =
applications, so making it difficult to extract the information you need =
to parse the RTP header extension, even if it is =
unencrypted.&nbsp;<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p></div><div><p =
class=3DMsoNormal>Besides these focussed issues, I would also urge you =
to consider the much broader comments Magnus made. The QoS problem is =
broad, and a point solution based on RTP header extensions - even if it =
were workable, which I doubt - would address only a small part of the =
problem space.<o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Colin<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=3DMsoNormal>On =
25 Feb 2014, at 01:45, Karl Stahl &lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:<o:p></o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Co=
lin,</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>If=
 the below were the case, it would be &#8220;DPI guesswork&#8221; that I =
also advice against. RTP doesn&#8217;t even have unique protocol header =
within UDP &#8211; it can even be confused with other UDP payload. =
</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ho=
wever, if the RTP is captured in a TURN-flow, as in TRAM Milestone 3, =
the network point that this flow is directed to and can apply QoS =
methods relevant to the network (which is not diffserve in Mobile OTT =
and Cable Networks) has not a too difficult tasks. Linking each RTP-flow =
by its ID and sequence number, and picking exactly the right traffic =
type and bandwidth parameters is doable (see inline below, we at Ingate =
do it already)!</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Al=
so DiffServ DSCP-bits are seldom maintained crossing network boundaries, =
thus carrying no relevant information at the receiving end, while the =
RTP extension header remains unchanged end-to-end. In DSCP-bits, there =
is no bandwidth requirement information when entering networks requiring =
reservation. That is always (and dynamically set) available in the RTP =
extension header for each packet.</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>An=
d, I am sure you are aware of the difficulties of getting DSCP-bits =
through OS sockets, which is even worse with multiple streams over the =
same UDP-port</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fu=
rther, defining the QoS RTP extension header as in RFC5285, does not in =
anyway conflict with other RTP extension headers or DSCP transfer or =
settings.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Colin =
Perkins [<a =
href=3D"mailto:csp@csperkins.org">mailto:csp@csperkins.org</a>] =
<br><b>Skickat:</b> den 24 februari 2014 23:52<br><b>Till:</b> Karl =
Stahl<br><b>Kopia:</b> <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; Magnus Westerlund; =
<a href=3D"mailto:tram@ietf.org">tram@ietf.org</a>; Harald =
Alvestrand<br><b>=C4mne:</b> Re: [rtcweb] [tram] Payload Types =
assignments</span><o:p></o:p></p></div></div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><p =
class=3DMsoNormal>Karl,<o:p></o:p></p><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><p class=3DMsoNormal>I =
strongly disagree with this suggestion. An RTP header extension, located =
at an unknown and variable offset into a packet that does not have a =
well-defined magic number in the header, <o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
is is how to find the traffic info:</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp; The payload of the classifier header =
extension element can be encoded</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp; using either the one-byte or two-byte header =
defined in [<a href=3D"http://tools.ietf.org/html/rfc5285" =
title=3D"&quot;A General Mechanism for RTP Header =
Extensions&quot;">rfc5285</a>] and</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp; shown below in Figure 1 and 2 =
below.</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 3 =
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; ID&nbsp;&nbsp; | len=3D1 =
|&nbsp;&nbsp; Namespace&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
Value&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; 0 =
(pad)&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp; Figure 1: Classifier Using the One-Byte =
Header</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 3 =
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
ID&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
len=3D2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; =
Namespace&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
Value&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p =
class=3DMsoNormal style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'page-break-before:always'><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Courier =
New","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp; Figure 2: Classifier Using the Two-Byte =
Header</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US>indicated using a dynamically assigned identifier that is =
conveyed in an out-of-band </span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- It is the TURN flow!!! Requested by the ICE protocol for the specific =
media to come!</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US>and encrypted signalling channel, </span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- Only the payload is encrypted in SRTP &#8211; leaving id, seq no and =
header ext to be used as intended!</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- Thus being the appropriate place&#8230;</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US>is not an appropriate place to put =
QoS information that has to be processed on a per-packet basis. =
</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US>If you =
want DiffServ, you know where to find =
it.</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'color:blue'>--- True, but if it cannot be =
used&#8230;</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal>Colin<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><div><div><p class=3DMsoNormal>On =
24 Feb 2014, at 22:09, Karl Stahl &lt;<a =
href=3D"mailto:karl.stahl@intertex.se">karl.stahl@intertex.se</a>&gt; =
wrote:<o:p></o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
suggest to the RTCWEB WG that the below from the September and October =
discussions on the relevant </span><span lang=3DEN-US>[rtcweb] [avtext] =
[mmusic]</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
lists <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a> =
is introduced into </span><span lang=3DEN-US>draft-ietf-rtcweb-rtp-usage =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>fo=
r usage of RFC 5285, to allow:</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(1=
) WebRTC applications to directly convey QoS related real-time traffic =
info to the network at points where RTP flow is directed to by TRAM =
Milstone 3, to be used by *<b>any network element implementing any =
suitable QoS methods for the particular network</b>* for =
</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(2=
) *<b>all</b>* WebRTC browsers *<b>and</b>* clients, under *<b>all</b>* =
OSs, and *<b>all</b>* current and future IP network, to achieve best QoE =
</span><o:p></o:p></p><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(3=
) *without* </span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ha=
ving to force WebRTC into application specific networks (such as IMS) =
instead of using the Internet (including OTT).</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e only further activity required, is to call for ISPs&#8217; to review =
whether the traffic information transferred by RFC 5285 is sufficient =
for current and future needs in their network as suggested in below =
repeated <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html</a></=
span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;snip&gt;<=
/span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&#8230;two =
parameters (e.g. two bytes each) are encoded into the RTP header =
extension:</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A) The =
maximum bandwidth requirement: Two bytes could contain everything from =
some bps for real-time text to Gbps for future 3D supersize =
telepresence&#8230; on a logarithmic scale.</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B) The =
quality characteristics for the stream, with the highest bit set to 1, =
we could allocate a bit each for quality type =
e.g:</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Best Effort, =
Audio, Video, Supplemental Video, Gaming, Data, Delay Insensitive (e.g. =
video streaming), Minimum Delay, Reliable Delivery, Prioritize X, =
Variation Y, that could be combined as required to describe the =
stream.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>And with =
highest bit set to 0, there could instead be a number for special usage =
that does not fit the general description of the individual =
bits.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&lt;/snip&gt;=
</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
en this could be assigned numbers to have an RFC in =
place.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th TRAM milestone 3 also place, </span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ma=
rket forces will drive ISPs and browser makers to implement just this, =
without even having it MUST-established.</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who does not want a &#8220;WebRTC-Ready&#8221; Internet =
access?&#8221; and</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;Who wants to use Chrome, if Firefox, Internet Explorer or Safari =
comes with much better QoE?&#8221; and vice =
versa.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Pl=
ease see further emails soon following this one, for details and =
history.</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] <b>F=F6r </b>Karl Stahl<br><b>Skickat:</b> den 22 oktober 2013 =
16:37<br><b>Till:</b> 'Harald Alvestrand'; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; 'Magnus =
Westerlund'<br><b>Kopia:</b> 'Colin Perkins'<br><b>=C4mne:</b> [rtcweb] =
[avtext] Payload Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p></di=
v></div><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>Harald, I mostly agree with the quality requirements of =
different real-time traffic that the WebRTC browser/application may use. =
But rather than asking the application, let's convey the bandwidth and =
priority requirements to the network. Just like with the Payload type =
(that is hard to squeeze that information into) it must be visible to =
the network (and not changed by the network, like diffserv bits are). =
Such marking must also be available for incoming traffic, which is =
especially important in RSVP type of networks, that has to reserve =
bandwidth for it. &nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>There is actually a good way to =
show these needs to the network (without using the PT, or diffserv bits, =
which aren&#8217;t sufficient anyway). </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Let's use the RTP header =
extension field that also is visible outside the encrypted payload. A =
week ago came <a =
href=3D"http://tools.ietf.org/html/draft-carlberg-avtext-classifier-00">h=
ttp://tools.ietf.org/html/draft-carlberg-avtext-classifier-00</a> =
&nbsp;that outlines the usage of the extension field for classification =
of traffic! This document does not yet outline what to put in there and =
how to encode it though.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Today's <a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10">http:/=
/tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-10</a> discusses other =
webrtc usages of the RTP header extension in 5.2 (there can be many =
header extensions according to RFC 5285) and in 9 there is &quot;WebRTC =
Use of RTP: Future Extensions&quot;.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>So, it looks obvious to use the =
RTP header extension to show the characteristics and bandwidth =
requirements to the network. It should not introduce any backward =
incompatibilities either.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Such marking is done in every =
RTP packet so it can be set individually for each stream and could even =
be changed during a session (e.g. when limiting the bandwidth based on =
RTCP feedback). RFC 5286 also specifies how RTP extension header usage =
can be negotiated in SDP. I think this could be easily done by the =
WebRTC browser for &quot;all current and future needs&quot; if properly =
specified now.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>I suggest that two parameters (e.g. two bytes each) are =
encoded into the RTP header extension:</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>A) The maximum bandwidth =
requirement: Two bytes could contain everything from some bps for =
real-time text to Gbps for future 3D supersize telepresence&#8230; on a =
logarithmic scale.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>B) The quality characteristics for the stream, with the =
highest bit set to 1, we could allocate a bit each quality =
e.g:</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>Best Effort, Audio, Video, Supplemental Video, Gaming, =
Data, Delay Insensitive (e.g. video streaming), Minimum Delay, Reliable =
Delivery, Prioritize X, Variation Y, that could be combined as required =
to describe the stream.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>And with highest bit set to 0, =
there could instead be a number for special usage that does not fit the =
general description of the individual bits.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Please note the totally =
different requirements a diffserv and an RSVP network have to know, so =
let&#8217;s put all into these bytes. (E.g. a diffserv network don't =
need the bandwidth usage, but RSVP reservation networks (e.g. cable and =
3G/4G OTT) do. There one should initially reserve the maximum bandwidth =
indicated, but can later re-reserve.)</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>/Karl</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>PS </span><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>Microsoft seems to have =
done work in this field, defining a proprietary attribute &#8220;MS =
Service Quality&#8221;; </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>However that seems to apply =
to the TURN server allocation request and would =
therefore:</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'>--- Apply to =
the whole UDP flow, and could not be set for each stream individually =
(with different requirements), and</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>--- Does not handle the =
bandwidth requirement for incoming real-time traffic (required to =
reserve in RSVP type of networks)</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>However the quality =
attributes conveyed and their encoding may &nbsp;be =
considered.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&nbsp;</span><o:p></o:p></p>=
<p class=3DMsoNormal><span lang=3DEN-US>This is 2.2.2.19 MS-Service =
Quality Attribute from </span><o:p></o:p></p><p class=3DMsoNormal><a =
href=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).a=
spx" =
title=3D"http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).=
aspx">http://msdn.microsoft.com/en-us/library/cc431507(v=3Doffice.12).asp=
x</a>&nbsp;<o:p></o:p></p><p class=3DMsoNormal>&nbsp;<o:p></o:p></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>MS-Service Quality =
Attribute</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'>The MS-Service =
Quality attribute is used to convey information about the data stream =
that the protocol client is intending to transfer over an allocated =
port. The protocol client SHOULD&lt;21&gt; include this attribute as =
part of an Allocate request message. A TURN server SHOULD use the =
information in this <span =
style=3D'background:yellow;mso-highlight:yellow'>attribute to make =
decisions about resource allocation, bandwidth prioritization, and data =
delivery methods</span>. If the attribute is not present in the Allocate =
request message, the TURN server SHOULD assume that the data stream is =
audio with best effort delivery. The format of this attribute is as =
follows... </span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>...</span></em><o:p></o:p></=
p><p class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>The following stream types =
are supported in this extension. All other stream types are reserved for =
future use.</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
style=3D'font-family:Symbol'>=A7 </span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0001&quot;: =
Audio</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
style=3D'font-family:Symbol'>=A7 </span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0002&quot;: =
Video</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
style=3D'font-family:Symbol'>=A7 </span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0003&quot;: =
Supplemental Video</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0004&quot;: =
Data</span></em><o:p></o:p></p><p class=3DMsoNormal><em><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif"'>Service =
Quality (2 bytes): The service quality level required by the protocol =
client for the stream.</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>The following service =
quality levels are supported in this extension. All other service =
quality levels are reserved for future use.</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0000&quot;: Best =
effort delivery.</span></em><o:p></o:p></p><p =
class=3DMsoNormal><em><span style=3D'font-family:Symbol'>=A7 =
</span></em><em><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif"'>&quot;0x0001&quot;: =
Reliable delivery.</span></em><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText>-----Ursprungligt meddelande-----<o:p></o:p></p><p =
class=3DMsoPlainText>Fr=E5n: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] F=F6r Harald Alvestrand<o:p></o:p></p><p =
class=3DMsoPlainText>Skickat: den 8 oktober 2013 13:01<o:p></o:p></p><p =
class=3DMsoPlainText>Till: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>=C4mne: Re: [rtcweb] Payload =
Types assignments was Re: SV: [mmusic] WGLC of =
draft-ietf-rtcweb-use-cases-and-requirements-11</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>On 10/08/2013 09:17 AM, Karl =
Stahl wrote:</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt; Hej Magnus,</span><o:p></o:p></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt;&gt; Also, are you really =
interested in knowing that it is VP9 vs H.264, </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt;&gt; =
isn't</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt; the questions this is video of this priority that is =
important?</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&gt; I think you need to more carefully consider what =
are the goals you </span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&gt; try to</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; achieve =
them.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; Actually, my concern is to =
get an idea of the maximum bandwidth that </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; could be required for a =
WebRTC (ICE) setup media flow. Both voice and </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; video should be prioritized =
over data (their individual priority is of </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; less importance as long as =
there is sufficient bandwidth for both).</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>You don't know that without =
knowing what the application is for.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>In, for instance, a shooter game =
with voice backchannels, the movement and event information (data) is =
MORE time sensitive than the voice data.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; With diffserv you =
don&#8217;t need to know the bandwidth requirement, but =
</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
with RSVP reservation (like in cable and mobile networks) you need to =
</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
know how much to reserve. Voice is like 100's kbit/s, video VP8 or =
</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>&gt; =
H.264 is like 3,5 mbps.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>Again, without knowing the =
application, you don't know that.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>The application could decide to =
use QCIF or HD, and the bandwidth variation of screencast (semi-static =
with sudden, large changes) is completely different from that of a =
talking head, which is again completely different from a high-movement =
scene.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; To add to the complication =
of codec variants, the video codecs in </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; question for WebRTC have =
variable bandwidth, and when there is a poor </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; connection we see Chrome =
reducing the video window size to reduce the bandwidth =
used...</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>&gt;&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; I think the payload type =
field at best can reflect a maximum bandwidth </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; to initially reserve =
bandwidth for, and thereafter make new </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; reservations if the =
bandwidth changes during the call. So could we </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; change RTP to show maximum =
bandwidth instead of payload type in that </span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&gt; field outside the encrypted =
payload :) ... Or maybe that is not a joke?</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>I think these ruminations only =
lead to one conclusion:</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>You can't tell what the needed =
bandwidth is up front without asking the =
application.</span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US>You can't tell what the right priority ranking is without =
asking the application.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>If you need to know the =
bandwidth or the priority up front, the application has to tell you. =
Anything else is pure heuristics.</span><o:p></o:p></p><p =
class=3DMsoPlainText><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span =
lang=3DEN-US>_______________________________________________</span><o:p><=
/o:p></p><p class=3DMsoPlainText><span lang=3DEN-US>rtcweb mailing =
list</span><o:p></o:p></p><p class=3DMsoPlainText><a =
href=3D"mailto:rtcweb@ietf.org"><span =
lang=3DEN-US>rtcweb@ietf.org</span></a><o:p></o:p></p><p =
class=3DMsoPlainText><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"><span =
lang=3DEN-US>https://www.ietf.org/mailman/listinfo/rtcweb</span></a><o:p>=
</o:p></p></blockquote></div><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;</span><o:p></o:p></p><div><div><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>--&nbsp;</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>Colin Perkins</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></span><o:p></o:p=
></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;</span><o:p></o:p></p></div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;</span><o:p></o:p></p></div><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;</span><o:p></o:p></p></div></div></blockquote></di=
v><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p><div><div><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
class=3Dapple-style-span><o:p>&nbsp;</o:p></span></p></div><div><p =
class=3DMsoNormal><span class=3Dapple-style-span><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>--&nbsp;<o:p></o:p></span></span></p></div><div><p =
class=3DMsoNormal><span class=3Dapple-style-span><span =
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>Colin =
Perkins<o:p></o:p></span></span></p></div><div><p =
class=3DMsoNormal><span class=3Dapple-style-span><span =
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a><o:p></o:p></span=
></span></p></div><div><p class=3DMsoNormal><span =
class=3Dapple-style-span><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></span></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p></div></div></body></html>
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From nobody Thu Mar 13 05:41:11 2014
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On 2014-03-12 17:05, Justin Uberti wrote:
>
> On Wed, Mar 12, 2014 at 1:03 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 

> 
>     Sorry, I don't understand your reasoning here. Is it to provide the
>     lowest possible drop probability, i.e. using the AFx class that has
>     least drop probability?
> 
> 
> The idea is to have the same drop probability as the most critical media
> that is flowing (e.g. voice), so that we don't end up in a situation
> where media flows fine, but the media crowds out STUN consent check
> packets, leading to call failure. 

Okay, that makes a certain sense. First of all this is not likely to
have the "highest" DSCP value. In fact it might not have the lowest drop
probability either. But, yes it might be prioritized over BE. This is
assuming that this DSCP marking do work and doesn't prevent the 6-tuple
to work. But lets take this as a separate issue.

> 
> 
>     If there exist nodes that stomp on some DSCP markings, and not only
>     remark them to Best Effort (BE), and instead drop then. Then, it is not
>     obvious that that an AFx class is the most appropriate to use. In that
>     case using BE might be more suitable for the consent freshness.
> 
> 
> If nodes stomp on certain AF-marked packets, and this is widespread,
> this essentially dooms use of DSCP at all, unless we want to build some
> sophisticated detection and fallback mechanism. IOW, it's not clear to
> me that marking packets as BE confers any value over not marking at all.

BE is no marking at all, i.e. DSCP = 0. But, I wanted to be clear that I
really meant best effort.

But, the question is what we do about nodes that not simple down mark
DSCP code points !=0. I see some potential solutions.

1) We recommend that one track the loss rates the different DSCP
markings individually and in cases when loss rates for some markings
doesn't match the expected relative behavior between different markings,
then one stops using DSCP markings and send everything as BE.

2) One expands the initial STUN exchange to a second round after
nomination to verify the DSCP code points that one intended to use is
not being blocked.

The alternative to do something is to ignore the issue. It might be
small enough that it isn't significant. I don't have data on this. If
someone has any data source it would be appreciated if they share.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Thu Mar 13 05:52:50 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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From nobody Thu Mar 13 07:00:21 2014
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Karl Stahl <karl.stahl@intertex.se> wrote:
> 
> [879 lines]
> [2135 lines]
> [816 lines]
> [1598 lines]

   I do appreciate your commitment to our tasks, Karl, but there is
no hope that very many of us can slog through all that text.

   If you want wide readership, may I suggest you limit the number
of lines you impose on the rest of us per day?

--
John Leslie <john@jlc.net>


From nobody Thu Mar 13 08:17:02 2014
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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 13 Mar 2014 08:16:01 -0700
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Subject: Re: [rtcweb] Areas of security concern
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Watson,

Thanks for your comments. Responses below.


On Tue, Mar 11, 2014 at 10:11 PM, Watson Ladd <watsonbladd@gmail.com> wrote:

>
> Problem 1: Which John Smith?
>
> Currently IdPs link attributes to identities.
>

The identities are phrased as RFC822-style names subordinated beneath
the domain name of the IdP (or if the IdP is a universal trust point, then
not subordinated). See:

http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09#section-5.6.5.3.3.1

There used to be a human-readable display-name property as well but that's
gone as of this version, precisely because of concerns of this type. Perhaps
you are reading a pre-09 version?



> Problem 2: IdP pointy bits
>

> An IdP has to cryptographically bind a username to some data. It would be
> nice if we could come up with a secure method to do this, and have the
> browser take care of it for the identity provider as much as possible,
> instead of everyone having to do it themselves in likely wrong ways. The
> webcrypto API has been roundly critiqued as being too low-level.
>

We certainly considered this, but that would basically amount to choosing
a specific identity system or standardizing our own, neither of which seems
totally awesome. Incidentally, cryptography isn't actually required here
(though it's probably the method of choice). The IdP could also just
store all the identity assertions it generates in a database and retrieve
it on demand.





Problem 3: Renegotiation
>
> The DTLS handshake being used is specified in RFC 4347. In particular, I
> don't see anything about the renegotiation bug being fixed. This enables an
> unknown key-share attack in which A thinks they are transmitting video to
> B, but really are transmitting it to C who knows that they are talking to A.
>

I'm certainly aware of this paper --- though as you know it just came
out recently --- but it's not clear to me how applicable it is,
since the client-authentication version of the attack (which I believe
is the appropriate one here) depends on the TLS server changing
its identity during the session, and RFC5763 requires that the
certificates presented during the DTLS handshake match the
fingerprints in the SDP:

   o  The certificate presented during the DTLS handshake MUST match the
      fingerprint exchanged via the signaling path in the SDP.  The
      security properties of this mechanism are described in Section 8.

RFC 5763 doesn't explicitly talk about renegotiation, so we probably
could add something about handling renegotiation.

If you think of some way in which the mechanisms above aren't sufficient
please drop me a private e-mail so we can discuss.


Problem 4: HMAC-SHA1 in SRTP
>
> If I've chased the chain of references correctly this is the sole MAC
> provided. Is it okay? I have no idea: SHA-1 has been significantly weakened
> in recent years.
>

As far as I know, HMAC-SHA1 is sufficiently secure for this purpose. Three
notes about this:

1. We are discussing audio and video packets so there are a lot of them
but modifying any particular one doesn't generally get you much.

2. The MACs are truncated to 80 or 32 bits.

3. There are GCM ciphersuites in the pipeline but they haven't landed yet.

This is really an issue for the IETF AVTCORE WG, however, since it applies
to all of SRTP, not just rtcweb.






> Problem 5: What you see is not what I sent
>
> This is an area that is somewhat underspecified, depending as it does on
> the media API's capabilities. The video frame could be silently replaced by
> a different video frame with rather different contents. Any authentication
> of the contents of the video frame would not carry over, but if the
> contents were sufficiently shocking/subtle this might not be noticed.
> Authentication independent of the website offering the video chat makes
> this attack worse, because users will trust what they see more. Isolation
> doesn't help because this is wholesale replacement, not editing.
>
> Most uses of this attack are for puerile purposes. I don't think it's a
> big concern. Then again I could be very, very wrong.
>

We certainly know about this form of attack. For instance, you can just
pretend to operate a calling service but instead of showing the remote
video you can just embed a YouTube video. There are effectively two
cases here:

- The call is not using isolated streams in which case you didn't have
  any protection for the media from the site anyway.

- The call is using isolated streams in which case the site will have
  no access to the true media and so can only blindly inject their
  own.

Obviously this isn't ideal but it's pretty hard to get rid of since the Web
already includes video and even without WebRTC, it's trivial for a site
to play any video it wants to you. We've talked about having some sort
of UI for showing each video element and its provenance, but it's a hard
UX problem.




> Problem 6: General UI issues
>
> Currently quite a bit of the security model depends on UI presentations of
> subtly different states. I don't know how well users will understand this,
> or how well the UI will work. Past experience suggests not well. Luckily we
> can mostly change this later/it isn't part of standardisation.
>

Yes, we are aware of this. Not sure there's much to do about it.



> Problem 7: VBR privacy leaks
>
> I can do no better then to refer to the paper.
> http://www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/reports/yes-we-can.pdfSpoken phrases could be identified from encrypted data alone, using a
> Hidden Markov Model when the length of packets was preserved by the
> encryption.
>

Yeah, this is also a known issue. See RFC 6562 for some recommendations
on how to address this. If you want to work on this issue, you should
probably
take it to the AVTCORE WG.

-Ekr

--00248c0d7914c28bcb04f47e70f0
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">Wats=
on,</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">Th=
anks for your comments. Responses below.</div><div class=3D"gmail_quote"><b=
r></div><div class=3D"gmail_quote">

<br></div><div class=3D"gmail_quote">On Tue, Mar 11, 2014 at 10:11 PM, Wats=
on Ladd <span dir=3D"ltr">&lt;<a href=3D"mailto:watsonbladd@gmail.com" targ=
et=3D"_blank">watsonbladd@gmail.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px=
;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1e=
x">

<div dir=3D"ltr"><br>Problem 1: Which John Smith?<br><br>Currently IdPs lin=
k attributes to identities. </div></blockquote><div><br></div><div>The iden=
tities are phrased as RFC822-style names subordinated beneath</div><div>
the domain name of the IdP (or if the IdP is a universal trust point, then<=
/div>
<div>not subordinated). See:</div><div><br></div><div><a href=3D"http://too=
ls.ietf.org/html/draft-ietf-rtcweb-security-arch-09#section-5.6.5.3.3.1">ht=
tp://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09#section-5.6.5.3=
.3.1</a><br>

</div><div><br></div><div>There used to be a human-readable display-name pr=
operty as well but that&#39;s</div><div>gone as of this version, precisely =
because of concerns of this type. Perhaps</div><div>you are reading a pre-0=
9 version?</div>

<div><br></div><div>=A0<br></div><blockquote class=3D"gmail_quote" style=3D=
"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,2=
04,204);border-left-style:solid;padding-left:1ex"><div dir=3D"ltr">Problem =
2: IdP pointy bits<br>

</div></blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border=
-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><br>An IdP has to cryp=
tographically bind a username to some data. It would be nice if we could co=
me up with a secure method to do this, and have the browser take care of it=
 for the identity provider as much as possible, instead of everyone having =
to do it themselves in likely wrong ways. The webcrypto API has been roundl=
y critiqued as being too low-level.<br>

</div></blockquote><div><br></div><div>We certainly considered this, but th=
at would basically amount to choosing</div><div>a specific identity system =
or standardizing our own, neither of which seems</div><div>totally awesome.=
 Incidentally, cryptography isn&#39;t actually required here</div>

<div>(though it&#39;s probably the method of choice). The IdP could also ju=
st</div><div>store all the identity assertions it generates in a database a=
nd retrieve</div><div>it on demand.</div><div><br></div><div><br></div>

<div><br></div><div><br></div><div><br></div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-col=
or:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir=3D"l=
tr">

<div>Problem 3: Renegotiation</div><div><br></div><div>The DTLS handshake b=
eing used is specified in RFC 4347. In particular, I don&#39;t see anything=
 about the renegotiation bug being fixed. This enables an unknown key-share=
 attack in which A thinks they are transmitting video to B, but really are =
transmitting it to C who knows that they are talking to A.</div>

</div></blockquote><div><br></div><div>I&#39;m certainly aware of this pape=
r --- though as you know it just came</div><div>out recently --- but it&#39=
;s not clear to me how applicable it is,</div><div>since the client-authent=
ication version of the attack (which I believe</div>

<div>is the appropriate one here) depends on the TLS server changing</div><=
div>its identity during the session, and RFC5763 requires that the</div><di=
v>certificates presented during the DTLS handshake match the</div><div>

fingerprints in the SDP:</div><div><br></div><div><div>=A0 =A0o =A0The cert=
ificate presented during the DTLS handshake MUST match the</div><div>=A0 =
=A0 =A0 fingerprint exchanged via the signaling path in the SDP. =A0The</di=
v><div>=A0 =A0 =A0 security properties of this mechanism are described in S=
ection 8.</div>

<div><br></div></div><div>RFC 5763 doesn&#39;t explicitly talk about renego=
tiation, so we probably<br></div><div>could add something about handling re=
negotiation.</div><div><br></div><div>If you think of some way in which the=
 mechanisms above aren&#39;t sufficient</div>

<div>please drop me a private e-mail so we can discuss.</div><div><br></div=
><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0=
px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-le=
ft-style:solid;padding-left:1ex">

<div dir=3D"ltr"><div>Problem 4: HMAC-SHA1 in SRTP</div><div><br></div><div=
>If I&#39;ve chased the chain of references correctly this is the sole MAC =
provided. Is it okay? I have no idea: SHA-1 has been significantly weakened=
 in recent years.</div>

</div></blockquote><div><br></div><div>As far as I know, HMAC-SHA1 is suffi=
ciently secure for this purpose. Three</div><div>notes about this:</div><di=
v><br></div><div>1. We are discussing audio and video packets so there are =
a lot of them</div>

<div>but modifying any particular one doesn&#39;t generally get you much.</=
div><div><br></div><div>2. The MACs are truncated to 80 or 32 bits.</div><d=
iv><br></div><div>3. There are GCM ciphersuites in the pipeline but they ha=
ven&#39;t landed yet.</div>

<div><br></div><div>This is really an issue for the IETF AVTCORE WG, howeve=
r, since it applies</div><div>to all of SRTP, not just rtcweb.</div><div><b=
r></div><div><br></div><div><br></div><div><br></div><div>=A0</div><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wid=
th:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-l=
eft:1ex">

<div dir=3D"ltr"><div>Problem 5: What you see is not what I sent</div><div>=
<br></div><div>This is an area that is somewhat underspecified, depending a=
s it does on the media API&#39;s capabilities. The video frame could be sil=
ently replaced by a different video frame with rather different contents. A=
ny authentication of the contents of the video frame would not carry over, =
but if the contents were sufficiently shocking/subtle this might not be not=
iced. Authentication independent of the website offering the video chat mak=
es this attack worse, because users will trust what they see more. Isolatio=
n doesn&#39;t help because this is wholesale replacement, not editing.</div=
>


<div><br></div><div>Most uses of this attack are for puerile purposes. I do=
n&#39;t think it&#39;s a big concern. Then again I could be very, very wron=
g.</div></div></blockquote><div><br></div><div>We certainly know about this=
 form of attack. For instance, you can just</div>

<div>pretend to operate a calling service but instead of showing the remote=
</div><div>video you can just embed a YouTube video. There are effectively =
two</div><div>cases here:</div><div><br></div><div>- The call is not using =
isolated streams in which case you didn&#39;t have</div>

<div>=A0 any protection for the media from the site anyway.</div><div><br><=
/div><div>- The call is using isolated streams in which case the site will =
have</div><div>=A0 no access to the true media and so can only blindly inje=
ct their</div>

<div>=A0 own.</div><div><br></div><div>Obviously this isn&#39;t ideal but i=
t&#39;s pretty hard to get rid of since the Web</div><div>already includes =
video and even without WebRTC, it&#39;s trivial for a site</div><div>to pla=
y any video it wants to you. We&#39;ve talked about having some sort</div>

<div>of UI for showing each video element and its provenance, but it&#39;s =
a hard</div><div>UX problem.</div><div><br></div><div><br></div><div>=A0</d=
iv><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bord=
er-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:soli=
d;padding-left:1ex">

<div dir=3D"ltr"><div>Problem 6: General UI issues</div><div><br>
</div><div>Currently quite a bit of the security model depends on UI presen=
tations of subtly different states. I don&#39;t know how well users will un=
derstand this, or how well the UI will work. Past experience suggests not w=
ell. Luckily we can mostly change this later/it isn&#39;t part of standardi=
sation.</div>

</div></blockquote><div><br></div><div>Yes, we are aware of this. Not sure =
there&#39;s much to do about it.</div><div><br></div><div>=A0</div><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wid=
th:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-l=
eft:1ex">

<div dir=3D"ltr"><div>Problem 7: VBR privacy leaks</div><div><br></div><div=
>I can do no better then to refer to the paper.=A0<a href=3D"http://www.inf=
sec.cs.uni-saarland.de/teaching/WS08/Seminar/reports/yes-we-can.pdf" target=
=3D"_blank">http://www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/repo=
rts/yes-we-can.pdf</a> Spoken phrases could be identified from encrypted da=
ta alone, using a Hidden Markov Model when the length of packets was preser=
ved by the encryption.</div>

</div></blockquote><div><br></div><div>Yeah, this is also a known issue. Se=
e RFC 6562 for some recommendations</div><div>on how to address this. If yo=
u want to work on this issue, you should probably</div><div>take it to the =
AVTCORE WG.</div>

<div><br></div><div>-Ekr</div><div><br></div></div></div></div>

--00248c0d7914c28bcb04f47e70f0--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 13 Mar 2014 09:15:15 -0700
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On Thu, Mar 13, 2014 at 5:40 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-12 17:05, Justin Uberti wrote:
> >
> > On Wed, Mar 12, 2014 at 1:03 AM, Magnus Westerlund
> > <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> > wrote:
> >
>
> >
> >     Sorry, I don't understand your reasoning here. Is it to provide the
> >     lowest possible drop probability, i.e. using the AFx class that has
> >     least drop probability?
> >
> >
> > The idea is to have the same drop probability as the most critical media
> > that is flowing (e.g. voice), so that we don't end up in a situation
> > where media flows fine, but the media crowds out STUN consent check
> > packets, leading to call failure.
>
> Okay, that makes a certain sense. First of all this is not likely to
> have the "highest" DSCP value. In fact it might not have the lowest drop
> probability either. But, yes it might be prioritized over BE. This is
> assuming that this DSCP marking do work and doesn't prevent the 6-tuple
> to work. But lets take this as a separate issue.
>
> >
> >
> >     If there exist nodes that stomp on some DSCP markings, and not only
> >     remark them to Best Effort (BE), and instead drop then. Then, it is
> not
> >     obvious that that an AFx class is the most appropriate to use. In
> that
> >     case using BE might be more suitable for the consent freshness.
> >
> >
> > If nodes stomp on certain AF-marked packets, and this is widespread,
> > this essentially dooms use of DSCP at all, unless we want to build some
> > sophisticated detection and fallback mechanism. IOW, it's not clear to
> > me that marking packets as BE confers any value over not marking at all.
>
> BE is no marking at all, i.e. DSCP = 0. But, I wanted to be clear that I
> really meant best effort.
>
> But, the question is what we do about nodes that not simple down mark
> DSCP code points !=0. I see some potential solutions.
>
> 1) We recommend that one track the loss rates the different DSCP
> markings individually and in cases when loss rates for some markings
> doesn't match the expected relative behavior between different markings,
> then one stops using DSCP markings and send everything as BE.
>
> 2) One expands the initial STUN exchange to a second round after
> nomination to verify the DSCP code points that one intended to use is
> not being blocked.
>
> The alternative to do something is to ignore the issue. It might be
> small enough that it isn't significant. I don't have data on this. If
> someone has any data source it would be appreciated if they share.
>
>
We could perform an experiment to determine how often this occurs in real
life. I was thinking about having clients send STUN requests with different
DSCP markings to a STUN server, and then analyzing the success rate. Open
to other ideas though.

--20cf302d497a617a9204f47f4393
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Mar 13, 2014 at 5:40 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>On 2014-03-12 17:05, Justin Uberti wrot=
e:<br>
&gt;<br>
&gt; On Wed, Mar 12, 2014 at 1:03 AM, Magnus Westerlund<br>
</div>&gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"=
_blank">magnus.westerlund@ericsson.com</a> &lt;mailto:<a href=3D"mailto:mag=
nus.westerlund@ericsson.com" target=3D"_blank">magnus.westerlund@ericsson.c=
om</a>&gt;&gt;<br>


<div>&gt; wrote:<br>
&gt;<br>
<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Sorry, I don&#39;t understand your reasoning here. Is it=
 to provide the<br>
&gt; =C2=A0 =C2=A0 lowest possible drop probability, i.e. using the AFx cla=
ss that has<br>
&gt; =C2=A0 =C2=A0 least drop probability?<br>
&gt;<br>
&gt;<br>
&gt; The idea is to have the same drop probability as the most critical med=
ia<br>
&gt; that is flowing (e.g. voice), so that we don&#39;t end up in a situati=
on<br>
&gt; where media flows fine, but the media crowds out STUN consent check<br=
>
&gt; packets, leading to call failure.<br>
<br>
</div>Okay, that makes a certain sense. First of all this is not likely to<=
br>
have the &quot;highest&quot; DSCP value. In fact it might not have the lowe=
st drop<br>
probability either. But, yes it might be prioritized over BE. This is<br>
assuming that this DSCP marking do work and doesn&#39;t prevent the 6-tuple=
<br>
to work. But lets take this as a separate issue.<br>
<div><br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 If there exist nodes that stomp on some DSCP markings, a=
nd not only<br>
&gt; =C2=A0 =C2=A0 remark them to Best Effort (BE), and instead drop then. =
Then, it is not<br>
&gt; =C2=A0 =C2=A0 obvious that that an AFx class is the most appropriate t=
o use. In that<br>
&gt; =C2=A0 =C2=A0 case using BE might be more suitable for the consent fre=
shness.<br>
&gt;<br>
&gt;<br>
&gt; If nodes stomp on certain AF-marked packets, and this is widespread,<b=
r>
&gt; this essentially dooms use of DSCP at all, unless we want to build som=
e<br>
&gt; sophisticated detection and fallback mechanism. IOW, it&#39;s not clea=
r to<br>
&gt; me that marking packets as BE confers any value over not marking at al=
l.<br>
<br>
</div>BE is no marking at all, i.e. DSCP =3D 0. But, I wanted to be clear t=
hat I<br>
really meant best effort.<br>
<br>
But, the question is what we do about nodes that not simple down mark<br>
DSCP code points !=3D0. I see some potential solutions.<br>
<br>
1) We recommend that one track the loss rates the different DSCP<br>
markings individually and in cases when loss rates for some markings<br>
doesn&#39;t match the expected relative behavior between different markings=
,<br>
then one stops using DSCP markings and send everything as BE.<br>
<br>
2) One expands the initial STUN exchange to a second round after<br>
nomination to verify the DSCP code points that one intended to use is<br>
not being blocked.<br>
<br>
The alternative to do something is to ignore the issue. It might be<br>
small enough that it isn&#39;t significant. I don&#39;t have data on this. =
If<br>
someone has any data source it would be appreciated if they share.<br>
<div><div><br></div></div></blockquote><div><br></div><div>We could perform=
 an experiment to determine how often this occurs in real life. I was think=
ing about having clients send STUN requests with different DSCP markings to=
 a STUN server, and then analyzing the success rate. Open to other ideas th=
ough.</div>


</div></div></div>

--20cf302d497a617a9204f47f4393--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 13 Mar 2014 09:16:57 -0700
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On Thu, Mar 13, 2014 at 5:52 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi Justin,
>
> > I think rtcp-mux has to be a MUST use when BUNDLE is used between the
> two endpoints. (If the endpoint doesn't support
> > BUNDLE, rtcp-mux is not required.)
> >
> > Otherwise, you get into weird edge cases, such as where you have a data
> channel (no RTCP component), and then want
> > to add audio (with RTP and RTCP components). You could bundle the audio
> RTP over the existing data channel transport,
> > but it's not clear what you should do with the RTCP component. Mandating
> rtcp-mux avoids this issue completely.
>
> According to section 4 in RFC 5761, if you multiplex RTP and RTCP on the
> same port, you need to assign dedicated PT values to the RTCP packets,
> which means that at least PT values 64-95 cannot be used for RTP.
>
> As you have been one of those talking about the risk of running out of PT
> values, I just want to make sure you have taken this into consideration :)
>
>
Yes. The upside of rtcp-mux far outweighs this minor limitation.

--001a11c308d67f312904f47f49f6
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Mar 13, 2014 at 5:52 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Justin,<br>
<div class=3D""><br>
&gt; I think rtcp-mux has to be a MUST use when BUNDLE is used between the =
two endpoints. (If the endpoint doesn&#39;t support<br>
&gt; BUNDLE, rtcp-mux is not required.)<br>
&gt;<br>
&gt; Otherwise, you get into weird edge cases, such as where you have a dat=
a channel (no RTCP component), and then want<br>
&gt; to add audio (with RTP and RTCP components). You could bundle the audi=
o RTP over the existing data channel transport,<br>
&gt; but it&#39;s not clear what you should do with the RTCP component. Man=
dating rtcp-mux avoids this issue completely.<br>
<br>
</div>According to section 4 in RFC 5761, if you multiplex RTP and RTCP on =
the same port, you need to assign dedicated PT values to the RTCP packets, =
which means that at least PT values 64-95 cannot be used for RTP.<br>
<br>
As you have been one of those talking about the risk of running out of PT v=
alues, I just want to make sure you have taken this into consideration :)<b=
r>
<br></blockquote><div><br></div><div>Yes. The upside of rtcp-mux far outwei=
ghs this minor limitation.</div></div></div></div>

--001a11c308d67f312904f47f49f6--


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From: Martin Thomson <martin.thomson@gmail.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 12 March 2014 01:29, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> I like to point out that the agreement and what is documented in
> rtp-usage is that WebRTC endpoint will have to make forwarded streams
> appear as locally originated. However, this as currently written does
> not apply to RTP middleboxes that interconnects multiple PeerConnections
> to form a multi-party session. This is deliberate to ensure that RTP
> topologies like RTP mixer and SFM do work on the RTP/RTCP level.

I'm still not clear on whether a middlebox interoperating with a
WebRTC endpoint is obligated to respect these rules or not.  I had
assumed that WebRTC is such a bully that they would be forced to.

> I am especially interested to know how you will "easy to apply manually"
> the synchronization. Can you please describe that. Because, that either
> requires an API call to tell the media framework, please consider the
> following CNAMES as equivalent, or some other method of telling the
> media framework that these different MST are actually originating from a
> common clock base.
In WebRTC, this would be:

var audio = pc1.getRemoteStreams()[0].getAudioTracks()[0];
var video = pc2.getRemoteStreams()[0].getVideoTracks()[0];
var synchronizedStream = new MediaStream(audio, video);

We'd have to say that this implies a statement about the clock base of
the tracks being the same.  ... and that this statement could be
wrong.

As far as it goes, I'm sure that other systems could build a similar
function, but none of those are standardized.

> I protested about this, not to lower the users privacy, but over the
> concern that this was raised without providing a case where it was
> obvious that the user's privacy was impacted. Martin, you said that you
> would think about it, and the next statement was lets change it, without
> providing any motivation why the concern was significant.

I'm sorry about that, I thought about it, but didn't want to waste
everyone's time (mine included) with an essay on the subject.


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From: Watson Ladd <watsonbladd@gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Areas of security concern
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On Thu, Mar 13, 2014 at 8:16 AM, Eric Rescorla <ekr@rtfm.com> wrote:
> Watson,
>
> Thanks for your comments. Responses below.
>
>
> On Tue, Mar 11, 2014 at 10:11 PM, Watson Ladd <watsonbladd@gmail.com> wrote:
>>
>>
>> Problem 1: Which John Smith?
>>
>> Currently IdPs link attributes to identities.
>
>
> The identities are phrased as RFC822-style names subordinated beneath
> the domain name of the IdP (or if the IdP is a universal trust point, then
> not subordinated). See:
>
> http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09#section-5.6.5.3.3.1
>
> There used to be a human-readable display-name property as well but that's
> gone as of this version, precisely because of concerns of this type. Perhaps
> you are reading a pre-09 version?

No, I am reading the most recent version. Ask someone named John Smith
how much email they get not addressed to them. Address-book style
limitations might be a useful UI feature, but I don't see how to do it
here without custom integration. We can always punt on this until
later: there is nothing stopping a more extended set of interactions
with IdP in their verification process.

>
>
>>
>> Problem 2: IdP pointy bits
>>
>>
>> An IdP has to cryptographically bind a username to some data. It would be
>> nice if we could come up with a secure method to do this, and have the
>> browser take care of it for the identity provider as much as possible,
>> instead of everyone having to do it themselves in likely wrong ways. The
>> webcrypto API has been roundly critiqued as being too low-level.
>
>
> We certainly considered this, but that would basically amount to choosing
> a specific identity system or standardizing our own, neither of which seems
> totally awesome. Incidentally, cryptography isn't actually required here
> (though it's probably the method of choice). The IdP could also just
> store all the identity assertions it generates in a database and retrieve
> it on demand.

As it stands you have an open invitation for everyone to try to roll
their own, and fail miserably. Even a higher-level sign/verify API
would
assuage my complaint. Probably a W3C thing though/educating the people
implementing/writing the higher-level API.

>
>
>
>
>
>> Problem 3: Renegotiation
>>
>> The DTLS handshake being used is specified in RFC 4347. In particular, I
>> don't see anything about the renegotiation bug being fixed. This enables an
>> unknown key-share attack in which A thinks they are transmitting video to B,
>> but really are transmitting it to C who knows that they are talking to A.
>
>
> I'm certainly aware of this paper --- though as you know it just came
> out recently --- but it's not clear to me how applicable it is,
> since the client-authentication version of the attack (which I believe
> is the appropriate one here) depends on the TLS server changing
> its identity during the session, and RFC5763 requires that the
> certificates presented during the DTLS handshake match the
> fingerprints in the SDP:
>
>    o  The certificate presented during the DTLS handshake MUST match the
>       fingerprint exchanged via the signaling path in the SDP.  The
>       security properties of this mechanism are described in Section 8.
>
> RFC 5763 doesn't explicitly talk about renegotiation, so we probably
> could add something about handling renegotiation.
>
> If you think of some way in which the mechanisms above aren't sufficient
> please drop me a private e-mail so we can discuss.

If the client and server stick to the flow in RFC 5764, then all is
good, because the client's signature verify message signs the
handshake and there is only one hello.

However, it needs to be made explicit that you have to stick exactly
to this message sequence. If a client lets the server trick it into
renegotiating, we have a problem.

Because you never put a state machine into the TLS 1.1 spec, I've had
a devil of a time figuring out if this is possible. In particular, it
could be that the transition to SRTP after the finished stops
renegotiation from being possible.

How does rekeying work in DTLS+SRTP?

>
>
>> Problem 4: HMAC-SHA1 in SRTP
>>
>> If I've chased the chain of references correctly this is the sole MAC
>> provided. Is it okay? I have no idea: SHA-1 has been significantly weakened
>> in recent years.
>
>
> As far as I know, HMAC-SHA1 is sufficiently secure for this purpose. Three
> notes about this:
>
> 1. We are discussing audio and video packets so there are a lot of them
> but modifying any particular one doesn't generally get you much.

Video codecs are infamous for having exploitable bugs. Sending in a
maliciously formed packet from outside could lead to fun.
But yes, I agree that HMAC-SHA1 is probably fine, and given that we
are getting CCM I'm not too worried. (Rekeying is a necessity with all
of these because of the small MACs).
>
> 2. The MACs are truncated to 80 or 32 bits.
>
> 3. There are GCM ciphersuites in the pipeline but they haven't landed yet.
>
> This is really an issue for the IETF AVTCORE WG, however, since it applies
> to all of SRTP, not just rtcweb.
>
>
>
>
>
>>
>> Problem 5: What you see is not what I sent
>>
>> This is an area that is somewhat underspecified, depending as it does on
>> the media API's capabilities. The video frame could be silently replaced by
>> a different video frame with rather different contents. Any authentication
>> of the contents of the video frame would not carry over, but if the contents
>> were sufficiently shocking/subtle this might not be noticed. Authentication
>> independent of the website offering the video chat makes this attack worse,
>> because users will trust what they see more. Isolation doesn't help because
>> this is wholesale replacement, not editing.
>>
>> Most uses of this attack are for puerile purposes. I don't think it's a
>> big concern. Then again I could be very, very wrong.
>
>
> We certainly know about this form of attack. For instance, you can just
> pretend to operate a calling service but instead of showing the remote
> video you can just embed a YouTube video. There are effectively two
> cases here:
>
> - The call is not using isolated streams in which case you didn't have
>   any protection for the media from the site anyway.
>
> - The call is using isolated streams in which case the site will have
>   no access to the true media and so can only blindly inject their
>   own.
>
> Obviously this isn't ideal but it's pretty hard to get rid of since the Web
> already includes video and even without WebRTC, it's trivial for a site
> to play any video it wants to you. We've talked about having some sort
> of UI for showing each video element and its provenance, but it's a hard
> UX problem.

Yeah, punting is perfectly fine on this one. I fully expect heavy exploitation.
>
>
>
>>
>> Problem 6: General UI issues
>>
>> Currently quite a bit of the security model depends on UI presentations of
>> subtly different states. I don't know how well users will understand this,
>> or how well the UI will work. Past experience suggests not well. Luckily we
>> can mostly change this later/it isn't part of standardisation.
>
>
> Yes, we are aware of this. Not sure there's much to do about it.
>
>
>>
>> Problem 7: VBR privacy leaks
>>
>> I can do no better then to refer to the paper.
>> http://www.infsec.cs.uni-saarland.de/teaching/WS08/Seminar/reports/yes-we-can.pdf
>> Spoken phrases could be identified from encrypted data alone, using a Hidden
>> Markov Model when the length of packets was preserved by the encryption.
>
>
> Yeah, this is also a known issue. See RFC 6562 for some recommendations
> on how to address this. If you want to work on this issue, you should
> probably
> take it to the AVTCORE WG.

Why is RFC 6562 not cited in the security architecture document? My
method was to work back from the security architecture RFC, so I
missed that this was fixed. Hopefully an implementor doesn't do the
same.

Sincerely,
Watson Ladd

>
> -Ekr
>



-- 
"Those who would give up Essential Liberty to purchase a little
Temporary Safety deserve neither  Liberty nor Safety."
-- Benjamin Franklin


From nobody Thu Mar 13 14:26:18 2014
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On 03/12/2014 09:35 AM, Dave Taht wrote:
> On Wed, Mar 12, 2014 at 3:55 AM, Justin Uberti <juberti@google.com> wrote:
>>
>>
>> On Tue, Mar 11, 2014 at 10:17 AM, Dave Taht <dave.taht@gmail.com> wrote:
>
>>>> Whether that always translates to "the highest DSCP code point" is ....
>>>> a good question.
>>> It doesn't.  Keith added af42 support to mosh last year and had several
>>> reports of packets being dropped with that marking. Worse the drops happened
>>> after 10 seconds of connectivity.
>>>
>>> Ecn markings on the other hand have thus far survived (if occasionally
>>> stomped on) across the open internet in that protocol.
>>
>> This is a concern of mine as well, which is why we haven't yet flipped the
>> switch to turn DSCP on by default in Chrome. I think we'll need to do some
>> experimentation to see how often DSCP makes things worse.
> Well, while I'm at this, I note that how linux handles DSCP in WIFI 802.11e
> is generally suboptimal. the CS6 and CS7 bit patterns map to the VO
> queue which is not aggregatable in wireless-n, CS4 and CS5
> map to the VI queue which has a lot of good properties for videoconference
> and voice traffic (but limited aggregation), and CS1 and CS2, which map
> to the background queue, which has good aggregation but limited txops.
That sounds like a Linux bug.... I would expect this translation to be 
done via lookup tables, not copying one field into another with 
different semantics (with or without bit-shift).

>
> I no longer have the bit patterns for AFxx memorized, but basically
> all that was returned to the 802.11e classifier was dscp >> 5 up until very
> recently. Lastly, there really is no interpretation whatsoever of the AF
> classes equating to "drop probability" in wifi (at least in lnux), they merely
> control queue selection and
>
> it is generally better with wireless-n to aim for better aggregation in
> one queue rather than using multiple queues as the cost of acquiring
> the media dominates.
>
>
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Date: Fri, 14 Mar 2014 05:56:38 +0000
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

Just to clarify: you would still use the a=3Drtcp-mux attribute, right?

It is e.g. needed to inform intermediaries (that may not understand BUNDLE)=
 about the usage of RTCP mux.

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Justin Uberti <juberti@google.com> wrote:





On Thu, Mar 13, 2014 at 5:52 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi Justin,

> I think rtcp-mux has to be a MUST use when BUNDLE is used between the two=
 endpoints. (If the endpoint doesn't support
> BUNDLE, rtcp-mux is not required.)
>
> Otherwise, you get into weird edge cases, such as where you have a data c=
hannel (no RTCP component), and then want
> to add audio (with RTP and RTCP components). You could bundle the audio R=
TP over the existing data channel transport,
> but it's not clear what you should do with the RTCP component. Mandating =
rtcp-mux avoids this issue completely.

According to section 4 in RFC 5761, if you multiplex RTP and RTCP on the sa=
me port, you need to assign dedicated PT values to the RTCP packets, which =
means that at least PT values 64-95 cannot be used for RTP.

As you have been one of those talking about the risk of running out of PT v=
alues, I just want to make sure you have taken this into consideration :)


Yes. The upside of rtcp-mux far outweighs this minor limitation.

--_000_cm0fhueueaioe4gj0u8m7vqa1394776596681emailandroidcom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body>
<pre style=3D"word-wrap:break-word; font-size:10.0pt; font-family:Tahoma; c=
olor:black">Hi,=0A=
=0A=
Just to clarify: you would still use the a=3Drtcp-mux attribute, right? =0A=
=0A=
It is e.g. needed to inform intermediaries (that may not understand BUNDLE)=
 about the usage of RTCP mux.=0A=
=0A=
Regards,=0A=
=0A=
Christer=0A=
=0A=
Sent from my Sony Ericsson Xperia arc S=0A=
=0A=
Justin Uberti &lt;juberti@google.com&gt; wrote:=0A=
=0A=
</pre>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Thu, Mar 13, 2014 at 5:52 AM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
Hi Justin,<br>
<div class=3D""><br>
&gt; I think rtcp-mux has to be a MUST use when BUNDLE is used between the =
two endpoints. (If the endpoint doesn't support<br>
&gt; BUNDLE, rtcp-mux is not required.)<br>
&gt;<br>
&gt; Otherwise, you get into weird edge cases, such as where you have a dat=
a channel (no RTCP component), and then want<br>
&gt; to add audio (with RTP and RTCP components). You could bundle the audi=
o RTP over the existing data channel transport,<br>
&gt; but it's not clear what you should do with the RTCP component. Mandati=
ng rtcp-mux avoids this issue completely.<br>
<br>
</div>
According to section 4 in RFC 5761, if you multiplex RTP and RTCP on the sa=
me port, you need to assign dedicated PT values to the RTCP packets, which =
means that at least PT values 64-95 cannot be used for RTP.<br>
<br>
As you have been one of those talking about the risk of running out of PT v=
alues, I just want to make sure you have taken this into consideration :)<b=
r>
<br>
</blockquote>
<div><br>
</div>
<div>Yes. The upside of rtcp-mux far outweighs this minor limitation.</div>
</div>
</div>
</div>
</div>
</body>
</html>

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On 2014-03-13 17:53, Martin Thomson wrote:
> On 12 March 2014 01:29, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> I like to point out that the agreement and what is documented in
>> rtp-usage is that WebRTC endpoint will have to make forwarded streams
>> appear as locally originated. However, this as currently written does
>> not apply to RTP middleboxes that interconnects multiple PeerConnections
>> to form a multi-party session. This is deliberate to ensure that RTP
>> topologies like RTP mixer and SFM do work on the RTP/RTCP level.
> 
> I'm still not clear on whether a middlebox interoperating with a
> WebRTC endpoint is obligated to respect these rules or not.  I had
> assumed that WebRTC is such a bully that they would be forced to.

Well, I know Colin and I did put in quite some thought to make the
WebRTC RTP usage rules such that you can use the common RTP middleboxes
as long as you have the right front-end, i.e. DTLS-SRTP and signalling
translation.

> 
>> I am especially interested to know how you will "easy to apply manually"
>> the synchronization. Can you please describe that. Because, that either
>> requires an API call to tell the media framework, please consider the
>> following CNAMES as equivalent, or some other method of telling the
>> media framework that these different MST are actually originating from a
>> common clock base.
> In WebRTC, this would be:
> 
> var audio = pc1.getRemoteStreams()[0].getAudioTracks()[0];
> var video = pc2.getRemoteStreams()[0].getVideoTracks()[0];
> var synchronizedStream = new MediaStream(audio, video);
> 
> We'd have to say that this implies a statement about the clock base of
> the tracks being the same.  ... and that this statement could be
> wrong.

I think this looks dangerous. If you default the assumption that
different CNAMEs of the MST you add are actually sharing clock base,
then a lot of basic programs that may group MST from different PCs into
one MS for convenience are going to cause serious issue in the media
frameworks, when they attempt to synchronize things.

> 
> As far as it goes, I'm sure that other systems could build a similar
> function, but none of those are standardized.
> 
>> I protested about this, not to lower the users privacy, but over the
>> concern that this was raised without providing a case where it was
>> obvious that the user's privacy was impacted. Martin, you said that you
>> would think about it, and the next statement was lets change it, without
>> providing any motivation why the concern was significant.
> 
> I'm sorry about that, I thought about it, but didn't want to waste
> everyone's time (mine included) with an essay on the subject.
> 

Having considered this a bit more, I do think the risk to privacy out
weighs the other concerns, as long as the API and its handling can be
sorted out, I don't think the above is sufficient to get good
functionality.

My understanding is that the risk to privacy exist when one have one JS
application enabling communication in different contexts during the same
application session. In cases where there might be participants in the
different contexts that could link the same end-point and thus human
user to different participant IDs (that otherwise are anonymous). Thus
revealing privacy concerns. As it would take the application designer to
take this risk into consideration to avoid it, I feel safer if the
application designer would need to take active measurements to perform
linkage.

I will wait an see if there are other inputs on this within the next
week. If nothing arises I will propose a text change to the rtp-usage to
address this.

Cheers

Magnus Westerlund
(As individual)

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Hi,
(as individual)

Although this discussion really should be on MMUSIC, I am still
following up here. Anything we come to conclude here really need to go
to MMUSIC for confirmation.

Justin, I do agree that it appear unnecessarily of having to include an
RTCP transport in an OFFER when the offerer know the answerer is BUNDLE
capable. I am fine with that and think the rule needed here is something
along the line of the below:

A BUNDLE supporting endpoint MUST implement a=rtcp-mux (that is already
required). In any answer by a BUNDLE capable endpoint, it MUST NOT
change the usage of a=rtcp-mux. If it is present in the offer, it must
be confirmed to be used in the answer, and if not present in the offer,
it MUST NOT be added in the answer. An BUNDLE capable endpoint is
RECOMMENDED to use a=rtcp-mux whenever suitable. Cases when it might not
be suitable include, lack RTP payload types, or cases when RTCP traffic
is needed to be on its own transport flow (5-tuple) to enable QoS or
other traffic handling functionalities.

The reason for writing the rule like the above, is that then an endpoint
can choose when creating an offer to renegotiate the use of a=rtcp-mux.
For example due to it running out of available PTs.

Feedback on this proposal?

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Justin Uberti <juberti@google.com>, Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Should we reference the pause/resume I-D?
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Date: Fri, 14 Mar 2014 11:25:51 +0000
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Subject: Re: [rtcweb] Should we reference the pause/resume I-D?
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On 2014-03-12 21:56, Justin Uberti wrote:=0A=
> Agreed.=0A=
=0A=
I may well be that the pause/resume doc will not be ready in time, so it =
=0A=
from that aspect is a bad idea. What about just stating that TMMBN/TMMBR =
=0A=
0 meaning pause should be supported?=0A=
=0A=
> Aren't there also patent declarations against this doc from=0A=
> multiple holders?=0A=
=0A=
I've not looked into that, but I deliberately chose to propose the "old" =
=0A=
TMMBN 0 way to signal that is mentioned in [1] (and not any of the "new" =
=0A=
signaling in [1]).=0A=
=0A=
>=0A=
> While SDP will likely be removed from the API in the future, the=0A=
> replacement would be a app-specific message sent over websockets, which=
=0A=
> seems like it would work just fine.=0A=
=0A=
s/would/could/: I don't think this has been discussed or agreed (and =0A=
websocket is of course just an example).=0A=
=0A=
I think it is worthwhile to push this kind of signaling into the media =0A=
plane if possible. For example, this would enable the UA to save =0A=
transmission/battery even if the app is badly written.=0A=
=0A=
As an example: if a MST sent over a PC is not used in any way at the =0A=
receiving end, the receiving end UA could signal TMMBR 0 to tell the =0A=
sending UA to not send. Sure, the receiving UA could also fire a =0A=
"negotiationneeded" event, but maybe the app doesn't listen.=0A=
=0A=
>=0A=
>=0A=
> On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba <bernard.aboba@gmail.com=
=0A=
> <mailto:bernard.aboba@gmail.com>> wrote:=0A=
>=0A=
>     While I do like the pause/resume draft, having core RTCWEB WG=0A=
>     documents (such as RTP Usage) depend on it seems like a bit of a=0A=
>     stretch. After all, the document was only adopted last week, and it=
=0A=
>     is a rare IETF WG document that can go from a -00 WG draft to=0A=
>     publication as an RFC in under a year.=0A=
>=0A=
>=0A=
>     On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=E5kansson LK=0A=
>     <stefan.lk.hakansson@ericsson.com=0A=
>     <mailto:stefan.lk.hakansson@ericsson.com>> wrote:=0A=
>=0A=
>         Hi,=0A=
>=0A=
>         at the IETF last week there was consensus in the AVTEXT WG=0A=
>         meeting to=0A=
>         adopt the pause/resume draft [1] as a WG draft.=0A=
>=0A=
>         In rtcweb/webrtc we're have the situation that we're discussing s=
o=0A=
>         called "doo-hickeys" as an API surface where the web app=0A=
>         (amongst other=0A=
>         things) can pause and resume the sending of a track. This can=0A=
>         be signaled with the direction attribute and a SDP O/A exchange=
=0A=
>         (and the=0A=
>         app pausing/resuming sending of a track would presumably lead to =
a=0A=
>         "negotiationneeded" event being fired).=0A=
>=0A=
>         But I think we should in addition require the browser to signal i=
t=0A=
>         according to one of the methods in [1] (e.g. TMMBN =3D 0), and al=
so=0A=
>         understand that signaling (a browser receiving TMMBN =3D 0 must=
=0A=
>         know that=0A=
>         the other end-point will pause sending).=0A=
>=0A=
>         My argument is that we know that many dislike SDP in rtcweb, and =
a=0A=
>         likely development is that it will be removed in a later version.=
 My=0A=
>         speculation is that signaling as outlined in [1] will then be=0A=
>         used for=0A=
>         pause/resume. If we support this from the beginning earlier=0A=
>         implementations could more easily interop with those later versio=
ns.=0A=
>=0A=
>=0A=
>         Stefan=0A=
>=0A=
>         [1]=0A=
>         https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-paus=
e-05.txt=0A=
>=0A=
>         _______________________________________________=0A=
>         rtcweb mailing list=0A=
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>         https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
>=0A=
>=0A=
>     _______________________________________________=0A=
>     rtcweb mailing list=0A=
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>=0A=
>     https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
>=0A=
=0A=


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From: Dave Taht <dave.taht@gmail.com>
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Subject: Re: [rtcweb] DSCP marking for STUN packets
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On Thu, Mar 13, 2014 at 2:25 PM, Harald Alvestrand <harald@alvestrand.no> w=
rote:
> On 03/12/2014 09:35 AM, Dave Taht wrote:
>>
>> On Wed, Mar 12, 2014 at 3:55 AM, Justin Uberti <juberti@google.com> wrot=
e:
>>>
>>>
>>>
>>> On Tue, Mar 11, 2014 at 10:17 AM, Dave Taht <dave.taht@gmail.com> wrote=
:
>>
>>
>>>>> Whether that always translates to "the highest DSCP code point" is ..=
..
>>>>> a good question.
>>>>
>>>> It doesn't.  Keith added af42 support to mosh last year and had severa=
l
>>>> reports of packets being dropped with that marking. Worse the drops
>>>> happened
>>>> after 10 seconds of connectivity.
>>>>
>>>> Ecn markings on the other hand have thus far survived (if occasionally
>>>> stomped on) across the open internet in that protocol.
>>>
>>>
>>> This is a concern of mine as well, which is why we haven't yet flipped
>>> the
>>> switch to turn DSCP on by default in Chrome. I think we'll need to do
>>> some
>>> experimentation to see how often DSCP makes things worse.
>>
>> Well, while I'm at this, I note that how linux handles DSCP in WIFI
>> 802.11e
>> is generally suboptimal. the CS6 and CS7 bit patterns map to the VO
>> queue which is not aggregatable in wireless-n, CS4 and CS5
>> map to the VI queue which has a lot of good properties for videoconferen=
ce
>> and voice traffic (but limited aggregation), and CS1 and CS2, which map
>> to the background queue, which has good aggregation but limited txops.
>
> That sounds like a Linux bug.... I would expect this translation to be do=
ne

That has existed since 802.11e was standardized in 2005.

> via lookup tables, not copying one field into another with different
> semantics (with or without bit-shift).

The diffserv codepoint IS preserved however it is classified into an 802.11=
e
queue based on the top 3 bits up until recently.

In linux 3.14 and later there is an optional classifier that does a diffser=
v
lookup to 802.11e queue translation.

These queues determine txop scheduling and bandwidth characteristics,
not drop probability.

http://en.wikipedia.org/wiki/IEEE_802.11e-2005#802.11e_MAC_protocol_operati=
on

In some versions of cerowrt I have obsoleted the VO queue entirely,
pushed several AFxx markings into the VI queue, and respected
background markings to push them into the BK queue, and am working
on something that will try to maximize txops and aggregation by
"pushing" traffic into better queues when needed.

>
>>
>> I no longer have the bit patterns for AFxx memorized, but basically
>> all that was returned to the 802.11e classifier was dscp >> 5 up until
>> very
>> recently. Lastly, there really is no interpretation whatsoever of the AF
>> classes equating to "drop probability" in wifi (at least in lnux), they
>> merely
>> control queue selection and

I am not aware of any OS that translates AFxx markings into drop
probability directly on wifi.


>> it is generally better with wireless-n to aim for better aggregation in
>> one queue rather than using multiple queues as the cost of acquiring
>> the media dominates.


>>
>>
>>
>



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html


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Hi,

I did review the JSEP -06 prior to the meeting. Now I have had time to
write-up my comments.

1) Section 1):

[W3C.WD-webrtc-20111027]
This is very old version of the API, please update to the currently
relevant version.

2) Section 3.4.1.1:
The m-line index is a zero-based index, referring to the Nth m-line in
the SDP.

a) I think the above is a bit confusing. As a zero-based index would
mean that the first m= section would have index 0. Thus, the referring
to the Nth m-line in the SDP is unclear. Is N, the provided index, in
which case it would be the N+1th m-line. Please rewrite.

b) I also wonder if one has to be clearer on what "the SDP" is. I assume
it is the one that has been generated by the agent, rather than the
latest sent or received?

3) Section 3.4.1.1:
"The MID uses the "media stream identification", as defined in
   [RFC5888] , to identify the m-line."

a spurious space after ref.

4) JSEP applications typically inform the browser to begin ICE gathering
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams to gather candidates
   for.

I find the statement that ICE candidate gathering starts first with
setLocalDescription. Doesn't the created offer already contain local
candidates, thus one must have already gathered some candidates? To me
it appears reasonable to start gathering earlier, rather than later.
Thus, as soon as one create the PeerConnection object and have added any
MST to it or a data channel, one can start gathering.

5) Section 3.5.2:
   In the parallel forking case where the Javascript application wishes
   to simultaneously exchange media with multiple peers, the flow is
   slightly more complex, but the Javascript application can follow the
   strategy that [RFC3960] describes using UPDATE.  (It is worth noting
   that use cases where this is the desired behavior are very unusual.)

I mostly react to the last sentence within the paragraph. I think the
poster child for semi-parallel forking was the MMORG that like to
establish PCs with anyone that is coming close in the virtual world.
Thus either needs to have a number of PCs on standby or have just one,
but treat that as parallel forking. And if you like to discourage the
later, then I think you might need to at least inform about the
possibility to have stand-by PCs.

6. Section 3.6:

   In the event that the local application state is reinitialized,
   either due to a user reload of the page, or a decision within the
   application to reload itself (perhaps to update to a new version), it
   is possible to keep an existing session alive, via a process called
   "rehydration".  The explicit goal of rehydration is to carry out this
   session resumption with no interaction with the remote side other
   than normal call signaling messages.

I think this paragraph makes two erronous statements:
a) "keep an existing session alive"
I think "resume" an existing session would be more correct. Because it
would be broken when the reload happens.

b) "with no interaction with the remote side"
My understanding is that due to DTLS and key handling a DTLS
renegotiation will be required with the peer. Isn't also a signalling
exchange likely to be required due to change of local DTLS cert as well
as ICE candidates as parameters? Any other parameters that will be
required to exchanges when rehydrating?

7. Section 3.6:
Next, a new offer is
   generated by the new PeerConnection; this offer will have new ICE and
   possibly new DTLS-SRTP certificate fingerprints (since the old ICE
   and SRTP state has been lost).

I think this may require quite a bit more details. A forward pointer to
where you will specify that?

8. Section 3.6:
   [OPEN ISSUE: EKR proposed an alternative rehydration approach where
   the actual internal PeerConnection object in the browser was kept
   alive for some time after the web page was killed and provided some
   way for a new page to acquire the old PeerConnection object.]

Can't we declare this open issue closed? I have seen no details on the
alternative proposal. Thus, lets work with the one that is present.

9. Section 4.1.1:
There is no discussion of how the bundle policies relate to the Data
Channels? Are they included or do they have a specific set of policies.
Please extend this to discuss this.

10. Section 4.1.3:

This text doesn't appear to contain a requirement on that one MUST have
called setRemote prior to being able to call it.

11. Section 4.1.4.1:
   While this could also be done with an inactive "pranswer", followed
   by a sendrecv "answer", the initial "pranswer" leaves the offer-
   answer exchange open, which means the caller cannot send an updated
   offer during this time.

Should one mention that also the callee can't send an offer without
doing a "final" answer?

12. Section 4.1.4.1

By the time the human has accepted the call and
   triggered the new offer, it is likely that the ICE and DTLS
   handshaking for all the channels will already be set up.

Wouldn't the handshaking have "finished" rather than "set up"?

13. Section 4.1.4.2:
   Rollback can only be used to cancel proposed changes; there is no
   support for rolling back from a stable state to a previous stable
   state.

What is meant with "proposed changes"? From my side it is not clear how
much you can do after having performed a SetLocal or SetRemote. I think
these needs to be treated separately.

For an offerer, they can createOffer, SetLocal, send to peer, receive
reject and then undo the SetLocal. That is straight forward, and if they
done the SetRemote, they would be consider in a new Stable state and
can't roll back anymore?

For the answering side, it receive an offer, does SetRemote,
CreateAnswer, determines that it doesn't like it, the can undo the
SetRemote and send a reject message to offerer.

But what about this case? it receive an offer, does SetRemote,
CreateAnswer, SetLocal and sends the message, then receive a reject from
the peer, because it didn't like the answer? Are the answerer then in a
stable state that can't be unrolled, or?

14. Section 4.1.5:
   This API indirectly controls the candidate gathering process.  When a
   local description is supplied, and the number of transports currently
   in use does not match the number of transports needed by the local
   description, the PeerConnection will create transports as needed and
   begin gathering candidates for them.

Another part of an earlier question regarding what triggers ICE gathering?

15. Section 4.1.7:
   TODO: Do we need to expose accessors for both the current and
   proposed local description?

This needs to be resolved, I don't know if I can answer this question.

16. Section 4.1.9:

Should this section discuss that it might be browser that has overriding
controls regarding if other candidates then relay ones may be provided,
i.e. privacy mode?

17. Section 4.1.9:
   Regardless of the configuration, the gathering process collects all
   available candidates, but excluded candidates will not be surfaced in
   onicecandidate callback or used for connectivity checks.

Doesn't this result in both resource consumption and that the configured
TURN server will learn the external IP address and port for the client?

18. Section 5.2.1:

Structure in relation between RTP media and DATA channel. Maybe you
should try to find a structure that makes the difference between the two
types of transport more clear, and what applies to the perspective.

19. Section 5.2.1:


   o  For the current default candidate, a "c=" line, as specific in
      [RFC5245], Section 4.3., paragraph 6.  If no candidates have been
      gathered yet, the default candidate should be set to the 'null'
      value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1.

What about host candidates?

20. Section 5.2.1:
The CNAME must be generated
      in accordance with draft-rescorla-random-cname-00.  [OPEN ISSUE:
      How are CNAMEs specified for MSTs?  Are they randomly generated
      for each MediaStream?  If so, can two MediaStreams be synced?]

I think you should point to the RTP usage for this particular piece.
Because we do have text there both for the CNAME generation and how
CNAMES among MSTs and between PeerConnection. Likely to be changed soon
based on the mailing list discussion.

21. Section 5.2.1:
   o  [OPEN ISSUE: Handling of a=imageattr]

Yes, I think imageattr should be supported to give guidance on suitable
resolutions that an endpoint like to receive for a particular MST.

22. Section 5.2.1:
   Once all m= sections have been generated, a session-level "a=group"
   attribute MUST be added as specified in [RFC5888].  This attribute
   MUST have semantics "BUNDLE", and MUST include the mid identifiers of
   each m= section.

Isn't this overriding BUNDLE as currently specified, forcing everything
into a single bundle group. Thus, preventing the bundle policies to work.

23. Section 5.2.2:

   o  If any MediaStreamTracks have been added, and there exist m=
      sections of the appropriate media type with no associated
      MediaStreamTracks

Isnt' the last a "MediaStreamTrack" rather than plural ones?

24. 5.2.3.3.  VoiceActivityDetection

   If the "VoiceActivityDetection" constraint is specified, with a value
   of "true", the offer MUST indicate support for silence suppression by
   including comfort noise ("CN") codecs for each supported clock rate,
   as specified in [RFC3389], Section 5.1.

As currently specified this text requires the usage of a comfort noise
codec, currently there is no such on required. I think the "MUST" needs
to be rewritten.

25. Section 5.3.1:
                                                                  Note
      that for simplicity, the answerer MAY use different payload types
      for codecs than the offerer, as it is not prohibited by
      Section 6.1.

I react to the word simplicity here. I never consider usage of different
PTs per direction anything simple. But, yes it is allowed by RFC 3264.
But I kind of protest to encourage the usage.

26. Section 5.3.2 etc.

When will we see a first draft of the text in the sections:
5.3.2, 5.3.3., 5.4, 5.5, 5.6, 5.7?

I really like to see it prior to the Interim meeting.

27. Section 6:
   Note: This section is still very early and is likely to significantly
   change as we get a better understanding of a) the use cases for this
   b) the implications at the protocol level c) feedback from
   implementors on what they can do.

Shouldn't this mention W3C discussion also?

28. Section 7 - Security Considerations:

First of all there are no references to the Security Considerations or
architecture document.

Secondly, I would like to have some security discussion on significant
issues with JSEP itself. Resource attacks on the local endpoint for
example?

29. section 10

There are a couple of draft references that are either published RFCs or
at least WG versions. Please update the reference list.

30. Section 10.2:
Looking at the spec, I think the following references are normative:

   [I-D.ietf-mmusic-trickle-ice]
              Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
              Incremental Provisioning of Candidates for the Interactive
              Connectivity Establishment (ICE) Protocol", draft-ietf-
              mmusic-trickle-ice-00 (work in progress), March 2013.

   [I-D.ietf-rtcweb-data-protocol]
              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
              Protocol", draft-ietf-rtcweb-data-protocol-04 (work in
              progress), February 2013.


Cheers

Magnus Westerlund

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Färögatan 6                 | Mobile +46 73 0949079
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From: Justin Uberti <juberti@google.com>
Date: Fri, 14 Mar 2014 09:15:20 -0700
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Subject: Re: [rtcweb] Should we reference the pause/resume I-D?
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On Fri, Mar 14, 2014 at 4:25 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 2014-03-12 21:56, Justin Uberti wrote:
> > Agreed.
>
> I may well be that the pause/resume doc will not be ready in time, so it
> from that aspect is a bad idea. What about just stating that TMMBN/TMMBR
> 0 meaning pause should be supported?
>

It's not clear to me what the interactions are between the signaling and
media plane when this occurs. That is, if I do doohickey.active =3D false,
should that cause a=3Drecvonly signaling and a TMMBN notification that the
track is paused? I am concerned that things can get out of sync (i.e. what
happens if we get TMMBR indicating resume, but signaling is still
a=3Drecvonly)?


> > Aren't there also patent declarations against this doc from
> > multiple holders?
>
> I've not looked into that, but I deliberately chose to propose the "old"
> TMMBN 0 way to signal that is mentioned in [1] (and not any of the "new"
> signaling in [1]).
>

OK.


>
> >
> > While SDP will likely be removed from the API in the future, the
> > replacement would be a app-specific message sent over websockets, which
> > seems like it would work just fine.
>
> s/would/could/: I don't think this has been discussed or agreed (and
> websocket is of course just an example).
>

Right, the whole thing is an example, but the point is that the new API
directions don't really affect this particular problem.

>
> I think it is worthwhile to push this kind of signaling into the media
> plane if possible. For example, this would enable the UA to save
> transmission/battery even if the app is badly written.
>
> As an example: if a MST sent over a PC is not used in any way at the
> receiving end, the receiving end UA could signal TMMBR 0 to tell the
> sending UA to not send. Sure, the receiving UA could also fire a
> "negotiationneeded" event, but maybe the app doesn't listen.
>

True, if the app doesn't perform the necessary signaling, things are not
going to work, but that applies to a lot of operations on the
PeerConnection.

Overall this feels to me like something that needs some thought to figure
out all the interactions between the API, the signaling plane, and the
media plane. As such I am inclined to leave this out of 1.0.

>
> >
> >
> > On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba <bernard.aboba@gmail.co=
m
> > <mailto:bernard.aboba@gmail.com>> wrote:
> >
> >     While I do like the pause/resume draft, having core RTCWEB WG
> >     documents (such as RTP Usage) depend on it seems like a bit of a
> >     stretch. After all, the document was only adopted last week, and it
> >     is a rare IETF WG document that can go from a -00 WG draft to
> >     publication as an RFC in under a year.
> >
> >
> >     On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=C3=A5kansson LK
> >     <stefan.lk.hakansson@ericsson.com
> >     <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
> >
> >         Hi,
> >
> >         at the IETF last week there was consensus in the AVTEXT WG
> >         meeting to
> >         adopt the pause/resume draft [1] as a WG draft.
> >
> >         In rtcweb/webrtc we're have the situation that we're discussing
> so
> >         called "doo-hickeys" as an API surface where the web app
> >         (amongst other
> >         things) can pause and resume the sending of a track. This can
> >         be signaled with the direction attribute and a SDP O/A exchange
> >         (and the
> >         app pausing/resuming sending of a track would presumably lead t=
o
> a
> >         "negotiationneeded" event being fired).
> >
> >         But I think we should in addition require the browser to signal
> it
> >         according to one of the methods in [1] (e.g. TMMBN =3D 0), and =
also
> >         understand that signaling (a browser receiving TMMBN =3D 0 must
> >         know that
> >         the other end-point will pause sending).
> >
> >         My argument is that we know that many dislike SDP in rtcweb, an=
d
> a
> >         likely development is that it will be removed in a later
> version. My
> >         speculation is that signaling as outlined in [1] will then be
> >         used for
> >         pause/resume. If we support this from the beginning earlier
> >         implementations could more easily interop with those later
> versions.
> >
> >
> >         Stefan
> >
> >         [1]
> >
> https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.txt
> >
> >         _______________________________________________
> >         rtcweb mailing list
> >         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> >         https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >
> >
> >     _______________________________________________
> >     rtcweb mailing list
> >     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> >     https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >
>
>

--089e013a26e68ee4bb04f493615f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Mar 14, 2014 at 4:25 AM, Stefan H=C3=A5kansson LK <span dir=
=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"=
_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">On 2014-03-12 21:56, Justin Uberti wrote:<br=
>
&gt; Agreed.<br>
<br>
I may well be that the pause/resume doc will not be ready in time, so it<br=
>
from that aspect is a bad idea. What about just stating that TMMBN/TMMBR<br=
>
0 meaning pause should be supported?<br></blockquote><div><br></div><div>It=
&#39;s not clear to me what the interactions are between the signaling and =
media plane when this occurs. That is, if I do doohickey.active =3D false, =
should that cause a=3Drecvonly signaling and a TMMBN notification that the =
track is paused? I am concerned that things can get out of sync (i.e. what =
happens if we get TMMBR indicating resume, but signaling is still a=3Drecvo=
nly)?</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D""><br>
&gt; Aren&#39;t there also patent declarations against this doc from<br>
&gt; multiple holders?<br>
<br>
</div>I&#39;ve not looked into that, but I deliberately chose to propose th=
e &quot;old&quot;<br>
TMMBN 0 way to signal that is mentioned in [1] (and not any of the &quot;ne=
w&quot;<br>
signaling in [1]).<br></blockquote><div><br></div><div>OK.</div><div>=C2=A0=
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex">
<div class=3D""><br>
&gt;<br>
&gt; While SDP will likely be removed from the API in the future, the<br>
&gt; replacement would be a app-specific message sent over websockets, whic=
h<br>
&gt; seems like it would work just fine.<br>
<br>
</div>s/would/could/: I don&#39;t think this has been discussed or agreed (=
and<br>
websocket is of course just an example).<br></blockquote><div><br></div><di=
v>Right, the whole thing is an example, but the point is that the new API d=
irections don&#39;t really affect this particular problem.=C2=A0</div><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex">


<br>
I think it is worthwhile to push this kind of signaling into the media<br>
plane if possible. For example, this would enable the UA to save<br>
transmission/battery even if the app is badly written.<br>
<br>
As an example: if a MST sent over a PC is not used in any way at the<br>
receiving end, the receiving end UA could signal TMMBR 0 to tell the<br>
sending UA to not send. Sure, the receiving UA could also fire a<br>
&quot;negotiationneeded&quot; event, but maybe the app doesn&#39;t listen.<=
br></blockquote><div><br></div><div>True, if the app doesn&#39;t perform th=
e necessary signaling, things are not going to work, but that applies to a =
lot of operations on the PeerConnection.</div>

<div><br></div><div>Overall this feels to me like something that needs some=
 thought to figure out all the interactions between the API, the signaling =
plane, and the media plane. As such I am inclined to leave this out of 1.0.=
=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D""><br>
&gt;<br>
&gt;<br>
&gt; On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba &lt;<a href=3D"mailto:=
bernard.aboba@gmail.com">bernard.aboba@gmail.com</a><br>
</div><div class=3D"">&gt; &lt;mailto:<a href=3D"mailto:bernard.aboba@gmail=
.com">bernard.aboba@gmail.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 While I do like the pause/resume draft, having core RTCW=
EB WG<br>
&gt; =C2=A0 =C2=A0 documents (such as RTP Usage) depend on it seems like a =
bit of a<br>
&gt; =C2=A0 =C2=A0 stretch. After all, the document was only adopted last w=
eek, and it<br>
&gt; =C2=A0 =C2=A0 is a rare IETF WG document that can go from a -00 WG dra=
ft to<br>
&gt; =C2=A0 =C2=A0 publication as an RFC in under a year.<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=C3=A5kansson =
LK<br>
&gt; =C2=A0 =C2=A0 &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com">=
stefan.lk.hakansson@ericsson.com</a><br>
</div><div><div class=3D"h5">&gt; =C2=A0 =C2=A0 &lt;mailto:<a href=3D"mailt=
o:stefan.lk.hakansson@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt=
;&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Hi,<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 at the IETF last week there was consensus =
in the AVTEXT WG<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 meeting to<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 adopt the pause/resume draft [1] as a WG d=
raft.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 In rtcweb/webrtc we&#39;re have the situat=
ion that we&#39;re discussing so<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 called &quot;doo-hickeys&quot; as an API s=
urface where the web app<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 (amongst other<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 things) can pause and resume the sending o=
f a track. This can<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 be signaled with the direction attribute a=
nd a SDP O/A exchange<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 (and the<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 app pausing/resuming sending of a track wo=
uld presumably lead to a<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 &quot;negotiationneeded&quot; event being =
fired).<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 But I think we should in addition require =
the browser to signal it<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 according to one of the methods in [1] (e.=
g. TMMBN =3D 0), and also<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 understand that signaling (a browser recei=
ving TMMBN =3D 0 must<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 know that<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 the other end-point will pause sending).<b=
r>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 My argument is that we know that many disl=
ike SDP in rtcweb, and a<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 likely development is that it will be remo=
ved in a later version. My<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 speculation is that signaling as outlined =
in [1] will then be<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 used for<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 pause/resume. If we support this from the =
beginning earlier<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 implementations could more easily interop =
with those later versions.<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Stefan<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 [1]<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"https://tools.ietf.org/id/draft=
-westerlund-avtext-rtp-stream-pause-05.txt" target=3D"_blank">https://tools=
.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.txt</a><br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 __________________________________________=
_____<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 rtcweb mailing list<br>
</div></div>&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"mailto:rtcweb@ietf.=
org">rtcweb@ietf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org">rtcw=
eb@ietf.org</a>&gt;<br>
<div class=3D"">&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"https://www.iet=
f.org/mailman/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailm=
an/listinfo/rtcweb</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 _______________________________________________<br>
&gt; =C2=A0 =C2=A0 rtcweb mailing list<br>
</div>&gt; =C2=A0 =C2=A0 <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<=
br>
&gt; =C2=A0 =C2=A0 <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt;<br>
<br>
</blockquote></div><br></div></div>

--089e013a26e68ee4bb04f493615f--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 14 Mar 2014 09:26:10 -0700
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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--bcaec546950550c86c04f49388f5
Content-Type: text/plain; charset=UTF-8

On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
> (as individual)
>
> Although this discussion really should be on MMUSIC, I am still
> following up here. Anything we come to conclude here really need to go
> to MMUSIC for confirmation.
>
> Justin, I do agree that it appear unnecessarily of having to include an
> RTCP transport in an OFFER when the offerer know the answerer is BUNDLE
> capable. I am fine with that and think the rule needed here is something
> along the line of the below:
>
> A BUNDLE supporting endpoint MUST implement a=rtcp-mux (that is already
> required). In any answer by a BUNDLE capable endpoint, it MUST NOT
> change the usage of a=rtcp-mux. If it is present in the offer, it must
> be confirmed to be used in the answer, and if not present in the offer,
> it MUST NOT be added in the answer. An BUNDLE capable endpoint is
> RECOMMENDED to use a=rtcp-mux whenever suitable. Cases when it might not
> be suitable include, lack RTP payload types, or cases when RTCP traffic
> is needed to be on its own transport flow (5-tuple) to enable QoS or
> other traffic handling functionalities.
>
> The reason for writing the rule like the above, is that then an endpoint
> can choose when creating an offer to renegotiate the use of a=rtcp-mux.
> For example due to it running out of available PTs.
>
> Feedback on this proposal?
>

I am not OK with this. It just gets too complicated to deal with all the
cases, especially if with the provision that rtcp-mux can change during the
session.

I think Christer's current language is what we want - if you are going to
BUNDLE, you MUST also use rtcp-mux for any RTP m-lines. If the answerer
cannot do that, it needs to reject the BUNDLE.

I agree with your initial point though - if you know that both sides are
going to BUNDLE and rtcp-mux, it makes sense to allow them to do that from
the outset, i.e. no need to include a RTCP transport in the offer.

--bcaec546950550c86c04f49388f5
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
(as individual)<br>
<br>
Although this discussion really should be on MMUSIC, I am still<br>
following up here. Anything we come to conclude here really need to go<br>
to MMUSIC for confirmation.<br>
<br>
Justin, I do agree that it appear unnecessarily of having to include an<br>
RTCP transport in an OFFER when the offerer know the answerer is BUNDLE<br>
capable. I am fine with that and think the rule needed here is something<br=
>
along the line of the below:<br>
<br>
A BUNDLE supporting endpoint MUST implement a=3Drtcp-mux (that is already<b=
r>
required). In any answer by a BUNDLE capable endpoint, it MUST NOT<br>
change the usage of a=3Drtcp-mux. If it is present in the offer, it must<br=
>
be confirmed to be used in the answer, and if not present in the offer,<br>
it MUST NOT be added in the answer. An BUNDLE capable endpoint is<br>
RECOMMENDED to use a=3Drtcp-mux whenever suitable. Cases when it might not<=
br>
be suitable include, lack RTP payload types, or cases when RTCP traffic<br>
is needed to be on its own transport flow (5-tuple) to enable QoS or<br>
other traffic handling functionalities.<br>
<br>
The reason for writing the rule like the above, is that then an endpoint<br=
>
can choose when creating an offer to renegotiate the use of a=3Drtcp-mux.<b=
r>
For example due to it running out of available PTs.<br>
<br>
Feedback on this proposal?<br></blockquote><div><br></div><div>I am not OK =
with this. It just gets too complicated to deal with all the cases, especia=
lly if with the provision that rtcp-mux can change during the session.</div=
>

<div><br></div><div>I think Christer&#39;s current language is what we want=
 - if you are going to BUNDLE, you MUST also use rtcp-mux for any RTP m-lin=
es. If the answerer cannot do that, it needs to reject the BUNDLE.</div>

<div><br></div><div>I agree with your initial point though - if you know th=
at both sides are going to BUNDLE and rtcp-mux, it makes sense to allow the=
m to do that from the outset, i.e. no need to include a RTCP transport in t=
he offer.</div>

</div></div></div>

--bcaec546950550c86c04f49388f5--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Date: Fri, 14 Mar 2014 16:30:36 +0000
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi Justin,

What do you mean by "RTCP transport in the offer"? Are you referring to the=
 a=3Drtcp attribute, or?

Even if we mandate rtcp-mux within a BUNDLE group, the a=3Drtcp-mux attribu=
te must still always be present - even if you know the remote endpoint supp=
orts rtcp-mux (or, even if you have already negotiated usage of rtcp-mux).

Regards,

Christer
________________________________________
From: Justin Uberti [juberti@google.com]
Sent: Friday, 14 March 2014 6:26 PM
To: Magnus Westerlund
Cc: Christer Holmberg; rtcweb@ietf.org; Eric Rescorla
Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux

On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund <magnus.westerlund@erics=
son.com<mailto:magnus.westerlund@ericsson.com>> wrote:
Hi,
(as individual)

Although this discussion really should be on MMUSIC, I am still
following up here. Anything we come to conclude here really need to go
to MMUSIC for confirmation.

Justin, I do agree that it appear unnecessarily of having to include an
RTCP transport in an OFFER when the offerer know the answerer is BUNDLE
capable. I am fine with that and think the rule needed here is something
along the line of the below:

A BUNDLE supporting endpoint MUST implement a=3Drtcp-mux (that is already
required). In any answer by a BUNDLE capable endpoint, it MUST NOT
change the usage of a=3Drtcp-mux. If it is present in the offer, it must
be confirmed to be used in the answer, and if not present in the offer,
it MUST NOT be added in the answer. An BUNDLE capable endpoint is
RECOMMENDED to use a=3Drtcp-mux whenever suitable. Cases when it might not
be suitable include, lack RTP payload types, or cases when RTCP traffic
is needed to be on its own transport flow (5-tuple) to enable QoS or
other traffic handling functionalities.

The reason for writing the rule like the above, is that then an endpoint
can choose when creating an offer to renegotiate the use of a=3Drtcp-mux.
For example due to it running out of available PTs.

Feedback on this proposal?

I am not OK with this. It just gets too complicated to deal with all the ca=
ses, especially if with the provision that rtcp-mux can change during the s=
ession.

I think Christer's current language is what we want - if you are going to B=
UNDLE, you MUST also use rtcp-mux for any RTP m-lines. If the answerer cann=
ot do that, it needs to reject the BUNDLE.

I agree with your initial point though - if you know that both sides are go=
ing to BUNDLE and rtcp-mux, it makes sense to allow them to do that from th=
e outset, i.e. no need to include a RTCP transport in the offer.


From nobody Fri Mar 14 09:44:40 2014
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From: Justin Uberti <juberti@google.com>
Date: Fri, 14 Mar 2014 09:44:06 -0700
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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--001a1134a86e6bcbad04f493c89b
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On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-13 17:53, Martin Thomson wrote:
> > On 12 March 2014 01:29, Magnus Westerlund
> > <magnus.westerlund@ericsson.com> wrote:
> >> I like to point out that the agreement and what is documented in
> >> rtp-usage is that WebRTC endpoint will have to make forwarded streams
> >> appear as locally originated. However, this as currently written does
> >> not apply to RTP middleboxes that interconnects multiple PeerConnections
> >> to form a multi-party session. This is deliberate to ensure that RTP
> >> topologies like RTP mixer and SFM do work on the RTP/RTCP level.
> >
> > I'm still not clear on whether a middlebox interoperating with a
> > WebRTC endpoint is obligated to respect these rules or not.  I had
> > assumed that WebRTC is such a bully that they would be forced to.
>
> Well, I know Colin and I did put in quite some thought to make the
> WebRTC RTP usage rules such that you can use the common RTP middleboxes
> as long as you have the right front-end, i.e. DTLS-SRTP and signalling
> translation.
>
> >
> >> I am especially interested to know how you will "easy to apply manually"
> >> the synchronization. Can you please describe that. Because, that either
> >> requires an API call to tell the media framework, please consider the
> >> following CNAMES as equivalent, or some other method of telling the
> >> media framework that these different MST are actually originating from a
> >> common clock base.
> > In WebRTC, this would be:
> >
> > var audio = pc1.getRemoteStreams()[0].getAudioTracks()[0];
> > var video = pc2.getRemoteStreams()[0].getVideoTracks()[0];
> > var synchronizedStream = new MediaStream(audio, video);
> >
> > We'd have to say that this implies a statement about the clock base of
> > the tracks being the same.  ... and that this statement could be
> > wrong.
>
> I think this looks dangerous. If you default the assumption that
> different CNAMEs of the MST you add are actually sharing clock base,
> then a lot of basic programs that may group MST from different PCs into
> one MS for convenience are going to cause serious issue in the media
> frameworks, when they attempt to synchronize things.
>
> >
> > As far as it goes, I'm sure that other systems could build a similar
> > function, but none of those are standardized.
> >
> >> I protested about this, not to lower the users privacy, but over the
> >> concern that this was raised without providing a case where it was
> >> obvious that the user's privacy was impacted. Martin, you said that you
> >> would think about it, and the next statement was lets change it, without
> >> providing any motivation why the concern was significant.
> >
> > I'm sorry about that, I thought about it, but didn't want to waste
> > everyone's time (mine included) with an essay on the subject.
> >
>
> Having considered this a bit more, I do think the risk to privacy out
> weighs the other concerns, as long as the API and its handling can be
> sorted out, I don't think the above is sufficient to get good
> functionality.
>
> My understanding is that the risk to privacy exist when one have one JS
> application enabling communication in different contexts during the same
> application session. In cases where there might be participants in the

different contexts that could link the same end-point and thus human
> user to different participant IDs (that otherwise are anonymous). Thus
> revealing privacy concerns. As it would take the application designer to
> take this risk into consideration to avoid it, I feel safer if the
> application designer would need to take active measurements to perform
> linkage.


> I will wait an see if there are other inputs on this within the next
> week. If nothing arises I will propose a text change to the rtp-usage to
> address this.
>
>
At an implementation level, one could imagine at least 3 policies for
generating CNAMEs:
a) per-session (i.e. per-PeerConnection)
b) per-page (i.e. shared between all PCs on a page)
c) per-page, persistent (i.e. shared between all PCs on a page, including
across page loads)

While we seem to agree that a) is the right solution for CNAMEs, it is
worth pointing out that we (Chrome) are currently doing c) for DTLS
certificates, to avoid performance problems with cert generation at page
load. Ergo, this linkability concern already exists, and I don't think it
is easy to solve it in the default case. There have been some proposals to
allow generation/storage of unique certs to prevent this linkability, but
this will require app input.

Ergo, we might want to match the DTLS behavior (i.e. generate unique CNAMEs
only when the certs are unique), to ensure we treat linkability
consistently.

--001a1134a86e6bcbad04f493c89b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 2014-03-13 17:53, Martin =
Thomson wrote:<br>
&gt; On 12 March 2014 01:29, Magnus Westerlund<br>
&gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlun=
d@ericsson.com</a>&gt; wrote:<br>
&gt;&gt; I like to point out that the agreement and what is documented in<b=
r>
&gt;&gt; rtp-usage is that WebRTC endpoint will have to make forwarded stre=
ams<br>
&gt;&gt; appear as locally originated. However, this as currently written d=
oes<br>
&gt;&gt; not apply to RTP middleboxes that interconnects multiple PeerConne=
ctions<br>
&gt;&gt; to form a multi-party session. This is deliberate to ensure that R=
TP<br>
&gt;&gt; topologies like RTP mixer and SFM do work on the RTP/RTCP level.<b=
r>
&gt;<br>
&gt; I&#39;m still not clear on whether a middlebox interoperating with a<b=
r>
&gt; WebRTC endpoint is obligated to respect these rules or not. =C2=A0I ha=
d<br>
&gt; assumed that WebRTC is such a bully that they would be forced to.<br>
<br>
</div>Well, I know Colin and I did put in quite some thought to make the<br=
>
WebRTC RTP usage rules such that you can use the common RTP middleboxes<br>
as long as you have the right front-end, i.e. DTLS-SRTP and signalling<br>
translation.<br>
<div class=3D""><br>
&gt;<br>
&gt;&gt; I am especially interested to know how you will &quot;easy to appl=
y manually&quot;<br>
&gt;&gt; the synchronization. Can you please describe that. Because, that e=
ither<br>
&gt;&gt; requires an API call to tell the media framework, please consider =
the<br>
&gt;&gt; following CNAMES as equivalent, or some other method of telling th=
e<br>
&gt;&gt; media framework that these different MST are actually originating =
from a<br>
&gt;&gt; common clock base.<br>
&gt; In WebRTC, this would be:<br>
&gt;<br>
&gt; var audio =3D pc1.getRemoteStreams()[0].getAudioTracks()[0];<br>
&gt; var video =3D pc2.getRemoteStreams()[0].getVideoTracks()[0];<br>
&gt; var synchronizedStream =3D new MediaStream(audio, video);<br>
&gt;<br>
&gt; We&#39;d have to say that this implies a statement about the clock bas=
e of<br>
&gt; the tracks being the same. =C2=A0... and that this statement could be<=
br>
&gt; wrong.<br>
<br>
</div>I think this looks dangerous. If you default the assumption that<br>
different CNAMEs of the MST you add are actually sharing clock base,<br>
then a lot of basic programs that may group MST from different PCs into<br>
one MS for convenience are going to cause serious issue in the media<br>
frameworks, when they attempt to synchronize things.<br>
<div class=3D""><br>
&gt;<br>
&gt; As far as it goes, I&#39;m sure that other systems could build a simil=
ar<br>
&gt; function, but none of those are standardized.<br>
&gt;<br>
&gt;&gt; I protested about this, not to lower the users privacy, but over t=
he<br>
&gt;&gt; concern that this was raised without providing a case where it was=
<br>
&gt;&gt; obvious that the user&#39;s privacy was impacted. Martin, you said=
 that you<br>
&gt;&gt; would think about it, and the next statement was lets change it, w=
ithout<br>
&gt;&gt; providing any motivation why the concern was significant.<br>
&gt;<br>
&gt; I&#39;m sorry about that, I thought about it, but didn&#39;t want to w=
aste<br>
&gt; everyone&#39;s time (mine included) with an essay on the subject.<br>
&gt;<br>
<br>
</div>Having considered this a bit more, I do think the risk to privacy out=
<br>
weighs the other concerns, as long as the API and its handling can be<br>
sorted out, I don&#39;t think the above is sufficient to get good<br>
functionality.<br>
<br>
My understanding is that the risk to privacy exist when one have one JS<br>
application enabling communication in different contexts during the same<br=
>
application session. In cases where there might be participants in the</blo=
ckquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border=
-left:1px #ccc solid;padding-left:1ex">
different contexts that could link the same end-point and thus human<br>
user to different participant IDs (that otherwise are anonymous). Thus<br>
revealing privacy concerns. As it would take the application designer to<br=
>
take this risk into consideration to avoid it, I feel safer if the<br>
application designer would need to take active measurements to perform<br>
linkage.</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
I will wait an see if there are other inputs on this within the next<br>
week. If nothing arises I will propose a text change to the rtp-usage to<br=
>
address this.<br><div class=3D"HOEnZb"><div class=3D"h5"><br></div></div></=
blockquote><div><br></div><div>At an implementation level, one could imagin=
e at least 3 policies for generating CNAMEs:<br></div><div>a) per-session (=
i.e. per-PeerConnection)</div>

<div>b) per-page (i.e. shared between all PCs on a page)</div><div>c) per-p=
age, persistent (i.e. shared between all PCs on a page, including across pa=
ge loads)</div><div><br></div><div>While we seem to agree that a) is the ri=
ght solution for CNAMEs, it is worth pointing out that we (Chrome) are curr=
ently doing c) for DTLS certificates, to avoid performance problems with ce=
rt generation at page load. Ergo, this linkability concern already exists, =
and I don&#39;t think it is easy to solve it in the default case. There hav=
e been some proposals to allow generation/storage of unique certs to preven=
t this linkability, but this will require app input.</div>

<div><br></div><div>Ergo, we might want to match the DTLS behavior (i.e. ge=
nerate unique CNAMEs only when the certs are unique), to ensure we treat li=
nkability consistently.</div></div></div></div>

--001a1134a86e6bcbad04f493c89b--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 14 Mar 2014 09:46:27 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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--047d7bdc945ed8002204f493d034
Content-Type: text/plain; charset=UTF-8

By "RTCP transport in the offer", I mean separate ICE candidates generated
for a RTCP ICE component, the original problem that I was proposing to
solve.

I agree the rtcp-mux attribute must always be present if you are going to
do rtcp-mux.


On Fri, Mar 14, 2014 at 9:30 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>
> Hi Justin,
>
> What do you mean by "RTCP transport in the offer"? Are you referring to
> the a=rtcp attribute, or?
>
> Even if we mandate rtcp-mux within a BUNDLE group, the a=rtcp-mux
> attribute must still always be present - even if you know the remote
> endpoint supports rtcp-mux (or, even if you have already negotiated usage
> of rtcp-mux).
>
> Regards,
>
> Christer
> ________________________________________
> From: Justin Uberti [juberti@google.com]
> Sent: Friday, 14 March 2014 6:26 PM
> To: Magnus Westerlund
> Cc: Christer Holmberg; rtcweb@ietf.org; Eric Rescorla
> Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
>
> On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund <
> magnus.westerlund@ericsson.com<mailto:magnus.westerlund@ericsson.com>>
> wrote:
> Hi,
> (as individual)
>
> Although this discussion really should be on MMUSIC, I am still
> following up here. Anything we come to conclude here really need to go
> to MMUSIC for confirmation.
>
> Justin, I do agree that it appear unnecessarily of having to include an
> RTCP transport in an OFFER when the offerer know the answerer is BUNDLE
> capable. I am fine with that and think the rule needed here is something
> along the line of the below:
>
> A BUNDLE supporting endpoint MUST implement a=rtcp-mux (that is already
> required). In any answer by a BUNDLE capable endpoint, it MUST NOT
> change the usage of a=rtcp-mux. If it is present in the offer, it must
> be confirmed to be used in the answer, and if not present in the offer,
> it MUST NOT be added in the answer. An BUNDLE capable endpoint is
> RECOMMENDED to use a=rtcp-mux whenever suitable. Cases when it might not
> be suitable include, lack RTP payload types, or cases when RTCP traffic
> is needed to be on its own transport flow (5-tuple) to enable QoS or
> other traffic handling functionalities.
>
> The reason for writing the rule like the above, is that then an endpoint
> can choose when creating an offer to renegotiate the use of a=rtcp-mux.
> For example due to it running out of available PTs.
>
> Feedback on this proposal?
>
> I am not OK with this. It just gets too complicated to deal with all the
> cases, especially if with the provision that rtcp-mux can change during the
> session.
>
> I think Christer's current language is what we want - if you are going to
> BUNDLE, you MUST also use rtcp-mux for any RTP m-lines. If the answerer
> cannot do that, it needs to reject the BUNDLE.
>
> I agree with your initial point though - if you know that both sides are
> going to BUNDLE and rtcp-mux, it makes sense to allow them to do that from
> the outset, i.e. no need to include a RTCP transport in the offer.
>

--047d7bdc945ed8002204f493d034
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">By &quot;RTCP transport in the offer&quot;, I mean separat=
e ICE candidates generated for a RTCP ICE component, the original problem t=
hat I was proposing to solve.<div><br></div><div>I agree the rtcp-mux attri=
bute must always be present if you are going to do rtcp-mux.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri,=
 Mar 14, 2014 at 9:30 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
Hi Justin,<br>
<br>
What do you mean by &quot;RTCP transport in the offer&quot;? Are you referr=
ing to the a=3Drtcp attribute, or?<br>
<br>
Even if we mandate rtcp-mux within a BUNDLE group, the a=3Drtcp-mux attribu=
te must still always be present - even if you know the remote endpoint supp=
orts rtcp-mux (or, even if you have already negotiated usage of rtcp-mux).<=
br>


<br>
Regards,<br>
<br>
Christer<br>
________________________________________<br>
From: Justin Uberti [<a href=3D"mailto:juberti@google.com">juberti@google.c=
om</a>]<br>
Sent: Friday, 14 March 2014 6:26 PM<br>
To: Magnus Westerlund<br>
Cc: Christer Holmberg; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</=
a>; Eric Rescorla<br>
Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund &lt;<a href=3D"mailto:ma=
gnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a>&lt;mailto:=
<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsso=
n.com</a>&gt;&gt; wrote:<br>


Hi,<br>
(as individual)<br>
<br>
Although this discussion really should be on MMUSIC, I am still<br>
following up here. Anything we come to conclude here really need to go<br>
to MMUSIC for confirmation.<br>
<br>
Justin, I do agree that it appear unnecessarily of having to include an<br>
RTCP transport in an OFFER when the offerer know the answerer is BUNDLE<br>
capable. I am fine with that and think the rule needed here is something<br=
>
along the line of the below:<br>
<br>
A BUNDLE supporting endpoint MUST implement a=3Drtcp-mux (that is already<b=
r>
required). In any answer by a BUNDLE capable endpoint, it MUST NOT<br>
change the usage of a=3Drtcp-mux. If it is present in the offer, it must<br=
>
be confirmed to be used in the answer, and if not present in the offer,<br>
it MUST NOT be added in the answer. An BUNDLE capable endpoint is<br>
RECOMMENDED to use a=3Drtcp-mux whenever suitable. Cases when it might not<=
br>
be suitable include, lack RTP payload types, or cases when RTCP traffic<br>
is needed to be on its own transport flow (5-tuple) to enable QoS or<br>
other traffic handling functionalities.<br>
<br>
The reason for writing the rule like the above, is that then an endpoint<br=
>
can choose when creating an offer to renegotiate the use of a=3Drtcp-mux.<b=
r>
For example due to it running out of available PTs.<br>
<br>
Feedback on this proposal?<br>
<br>
I am not OK with this. It just gets too complicated to deal with all the ca=
ses, especially if with the provision that rtcp-mux can change during the s=
ession.<br>
<br>
I think Christer&#39;s current language is what we want - if you are going =
to BUNDLE, you MUST also use rtcp-mux for any RTP m-lines. If the answerer =
cannot do that, it needs to reject the BUNDLE.<br>
<br>
I agree with your initial point though - if you know that both sides are go=
ing to BUNDLE and rtcp-mux, it makes sense to allow them to do that from th=
e outset, i.e. no need to include a RTCP transport in the offer.<br>
</div></div></blockquote></div><br></div>

--047d7bdc945ed8002204f493d034--


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Thanks for the detailed review. We will work through these comments over
the next few days.


On Fri, Mar 14, 2014 at 6:05 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> I did review the JSEP -06 prior to the meeting. Now I have had time to
> write-up my comments.
>
> 1) Section 1):
>
> [W3C.WD-webrtc-20111027]
> This is very old version of the API, please update to the currently
> relevant version.
>
> 2) Section 3.4.1.1:
> The m-line index is a zero-based index, referring to the Nth m-line in
> the SDP.
>
> a) I think the above is a bit confusing. As a zero-based index would
> mean that the first m=3D section would have index 0. Thus, the referring
> to the Nth m-line in the SDP is unclear. Is N, the provided index, in
> which case it would be the N+1th m-line. Please rewrite.
>
> b) I also wonder if one has to be clearer on what "the SDP" is. I assume
> it is the one that has been generated by the agent, rather than the
> latest sent or received?
>
> 3) Section 3.4.1.1:
> "The MID uses the "media stream identification", as defined in
>    [RFC5888] , to identify the m-line."
>
> a spurious space after ref.
>
> 4) JSEP applications typically inform the browser to begin ICE gathering
>    via the information supplied to setLocalDescription, as this is where
>    the app specifies the number of media streams to gather candidates
>    for.
>
> I find the statement that ICE candidate gathering starts first with
> setLocalDescription. Doesn't the created offer already contain local
> candidates, thus one must have already gathered some candidates? To me
> it appears reasonable to start gathering earlier, rather than later.
> Thus, as soon as one create the PeerConnection object and have added any
> MST to it or a data channel, one can start gathering.
>
> 5) Section 3.5.2:
>    In the parallel forking case where the Javascript application wishes
>    to simultaneously exchange media with multiple peers, the flow is
>    slightly more complex, but the Javascript application can follow the
>    strategy that [RFC3960] describes using UPDATE.  (It is worth noting
>    that use cases where this is the desired behavior are very unusual.)
>
> I mostly react to the last sentence within the paragraph. I think the
> poster child for semi-parallel forking was the MMORG that like to
> establish PCs with anyone that is coming close in the virtual world.
> Thus either needs to have a number of PCs on standby or have just one,
> but treat that as parallel forking. And if you like to discourage the
> later, then I think you might need to at least inform about the
> possibility to have stand-by PCs.
>
> 6. Section 3.6:
>
>    In the event that the local application state is reinitialized,
>    either due to a user reload of the page, or a decision within the
>    application to reload itself (perhaps to update to a new version), it
>    is possible to keep an existing session alive, via a process called
>    "rehydration".  The explicit goal of rehydration is to carry out this
>    session resumption with no interaction with the remote side other
>    than normal call signaling messages.
>
> I think this paragraph makes two erronous statements:
> a) "keep an existing session alive"
> I think "resume" an existing session would be more correct. Because it
> would be broken when the reload happens.
>
> b) "with no interaction with the remote side"
> My understanding is that due to DTLS and key handling a DTLS
> renegotiation will be required with the peer. Isn't also a signalling
> exchange likely to be required due to change of local DTLS cert as well
> as ICE candidates as parameters? Any other parameters that will be
> required to exchanges when rehydrating?
>
> 7. Section 3.6:
> Next, a new offer is
>    generated by the new PeerConnection; this offer will have new ICE and
>    possibly new DTLS-SRTP certificate fingerprints (since the old ICE
>    and SRTP state has been lost).
>
> I think this may require quite a bit more details. A forward pointer to
> where you will specify that?
>
> 8. Section 3.6:
>    [OPEN ISSUE: EKR proposed an alternative rehydration approach where
>    the actual internal PeerConnection object in the browser was kept
>    alive for some time after the web page was killed and provided some
>    way for a new page to acquire the old PeerConnection object.]
>
> Can't we declare this open issue closed? I have seen no details on the
> alternative proposal. Thus, lets work with the one that is present.
>
> 9. Section 4.1.1:
> There is no discussion of how the bundle policies relate to the Data
> Channels? Are they included or do they have a specific set of policies.
> Please extend this to discuss this.
>
> 10. Section 4.1.3:
>
> This text doesn't appear to contain a requirement on that one MUST have
> called setRemote prior to being able to call it.
>
> 11. Section 4.1.4.1:
>    While this could also be done with an inactive "pranswer", followed
>    by a sendrecv "answer", the initial "pranswer" leaves the offer-
>    answer exchange open, which means the caller cannot send an updated
>    offer during this time.
>
> Should one mention that also the callee can't send an offer without
> doing a "final" answer?
>
> 12. Section 4.1.4.1
>
> By the time the human has accepted the call and
>    triggered the new offer, it is likely that the ICE and DTLS
>    handshaking for all the channels will already be set up.
>
> Wouldn't the handshaking have "finished" rather than "set up"?
>
> 13. Section 4.1.4.2:
>    Rollback can only be used to cancel proposed changes; there is no
>    support for rolling back from a stable state to a previous stable
>    state.
>
> What is meant with "proposed changes"? From my side it is not clear how
> much you can do after having performed a SetLocal or SetRemote. I think
> these needs to be treated separately.
>
> For an offerer, they can createOffer, SetLocal, send to peer, receive
> reject and then undo the SetLocal. That is straight forward, and if they
> done the SetRemote, they would be consider in a new Stable state and
> can't roll back anymore?
>
> For the answering side, it receive an offer, does SetRemote,
> CreateAnswer, determines that it doesn't like it, the can undo the
> SetRemote and send a reject message to offerer.
>
> But what about this case? it receive an offer, does SetRemote,
> CreateAnswer, SetLocal and sends the message, then receive a reject from
> the peer, because it didn't like the answer? Are the answerer then in a
> stable state that can't be unrolled, or?
>
> 14. Section 4.1.5:
>    This API indirectly controls the candidate gathering process.  When a
>    local description is supplied, and the number of transports currently
>    in use does not match the number of transports needed by the local
>    description, the PeerConnection will create transports as needed and
>    begin gathering candidates for them.
>
> Another part of an earlier question regarding what triggers ICE gathering=
?
>
> 15. Section 4.1.7:
>    TODO: Do we need to expose accessors for both the current and
>    proposed local description?
>
> This needs to be resolved, I don't know if I can answer this question.
>
> 16. Section 4.1.9:
>
> Should this section discuss that it might be browser that has overriding
> controls regarding if other candidates then relay ones may be provided,
> i.e. privacy mode?
>
> 17. Section 4.1.9:
>    Regardless of the configuration, the gathering process collects all
>    available candidates, but excluded candidates will not be surfaced in
>    onicecandidate callback or used for connectivity checks.
>
> Doesn't this result in both resource consumption and that the configured
> TURN server will learn the external IP address and port for the client?
>
> 18. Section 5.2.1:
>
> Structure in relation between RTP media and DATA channel. Maybe you
> should try to find a structure that makes the difference between the two
> types of transport more clear, and what applies to the perspective.
>
> 19. Section 5.2.1:
>
>
>    o  For the current default candidate, a "c=3D" line, as specific in
>       [RFC5245], Section 4.3., paragraph 6.  If no candidates have been
>       gathered yet, the default candidate should be set to the 'null'
>       value defined in [I-D.ietf-mmusic-trickle-ice], Section 5.1.
>
> What about host candidates?
>
> 20. Section 5.2.1:
> The CNAME must be generated
>       in accordance with draft-rescorla-random-cname-00.  [OPEN ISSUE:
>       How are CNAMEs specified for MSTs?  Are they randomly generated
>       for each MediaStream?  If so, can two MediaStreams be synced?]
>
> I think you should point to the RTP usage for this particular piece.
> Because we do have text there both for the CNAME generation and how
> CNAMES among MSTs and between PeerConnection. Likely to be changed soon
> based on the mailing list discussion.
>
> 21. Section 5.2.1:
>    o  [OPEN ISSUE: Handling of a=3Dimageattr]
>
> Yes, I think imageattr should be supported to give guidance on suitable
> resolutions that an endpoint like to receive for a particular MST.
>
> 22. Section 5.2.1:
>    Once all m=3D sections have been generated, a session-level "a=3Dgroup=
"
>    attribute MUST be added as specified in [RFC5888].  This attribute
>    MUST have semantics "BUNDLE", and MUST include the mid identifiers of
>    each m=3D section.
>
> Isn't this overriding BUNDLE as currently specified, forcing everything
> into a single bundle group. Thus, preventing the bundle policies to work.
>
> 23. Section 5.2.2:
>
>    o  If any MediaStreamTracks have been added, and there exist m=3D
>       sections of the appropriate media type with no associated
>       MediaStreamTracks
>
> Isnt' the last a "MediaStreamTrack" rather than plural ones?
>
> 24. 5.2.3.3.  VoiceActivityDetection
>
>    If the "VoiceActivityDetection" constraint is specified, with a value
>    of "true", the offer MUST indicate support for silence suppression by
>    including comfort noise ("CN") codecs for each supported clock rate,
>    as specified in [RFC3389], Section 5.1.
>
> As currently specified this text requires the usage of a comfort noise
> codec, currently there is no such on required. I think the "MUST" needs
> to be rewritten.
>
> 25. Section 5.3.1:
>                                                                   Note
>       that for simplicity, the answerer MAY use different payload types
>       for codecs than the offerer, as it is not prohibited by
>       Section 6.1.
>
> I react to the word simplicity here. I never consider usage of different
> PTs per direction anything simple. But, yes it is allowed by RFC 3264.
> But I kind of protest to encourage the usage.
>
> 26. Section 5.3.2 etc.
>
> When will we see a first draft of the text in the sections:
> 5.3.2, 5.3.3., 5.4, 5.5, 5.6, 5.7?
>
> I really like to see it prior to the Interim meeting.
>
> 27. Section 6:
>    Note: This section is still very early and is likely to significantly
>    change as we get a better understanding of a) the use cases for this
>    b) the implications at the protocol level c) feedback from
>    implementors on what they can do.
>
> Shouldn't this mention W3C discussion also?
>
> 28. Section 7 - Security Considerations:
>
> First of all there are no references to the Security Considerations or
> architecture document.
>
> Secondly, I would like to have some security discussion on significant
> issues with JSEP itself. Resource attacks on the local endpoint for
> example?
>
> 29. section 10
>
> There are a couple of draft references that are either published RFCs or
> at least WG versions. Please update the reference list.
>
> 30. Section 10.2:
> Looking at the spec, I think the following references are normative:
>
>    [I-D.ietf-mmusic-trickle-ice]
>               Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
>               Incremental Provisioning of Candidates for the Interactive
>               Connectivity Establishment (ICE) Protocol", draft-ietf-
>               mmusic-trickle-ice-00 (work in progress), March 2013.
>
>    [I-D.ietf-rtcweb-data-protocol]
>               Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
>               Protocol", draft-ietf-rtcweb-data-protocol-04 (work in
>               progress), February 2013.
>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
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<div dir=3D"ltr">Thanks for the detailed review. We will work through these=
 comments over the next few days.</div><div class=3D"gmail_extra"><br><br><=
div class=3D"gmail_quote">On Fri, Mar 14, 2014 at 6:05 AM, Magnus Westerlun=
d <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" t=
arget=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
I did review the JSEP -06 prior to the meeting. Now I have had time to<br>
write-up my comments.<br>
<br>
1) Section 1):<br>
<br>
[W3C.WD-webrtc-20111027]<br>
This is very old version of the API, please update to the currently<br>
relevant version.<br>
<br>
2) Section <a href=3D"http://3.4.1.1" target=3D"_blank">3.4.1.1</a>:<br>
The m-line index is a zero-based index, referring to the Nth m-line in<br>
the SDP.<br>
<br>
a) I think the above is a bit confusing. As a zero-based index would<br>
mean that the first m=3D section would have index 0. Thus, the referring<br=
>
to the Nth m-line in the SDP is unclear. Is N, the provided index, in<br>
which case it would be the N+1th m-line. Please rewrite.<br>
<br>
b) I also wonder if one has to be clearer on what &quot;the SDP&quot; is. I=
 assume<br>
it is the one that has been generated by the agent, rather than the<br>
latest sent or received?<br>
<br>
3) Section <a href=3D"http://3.4.1.1" target=3D"_blank">3.4.1.1</a>:<br>
&quot;The MID uses the &quot;media stream identification&quot;, as defined =
in<br>
=C2=A0 =C2=A0[RFC5888] , to identify the m-line.&quot;<br>
<br>
a spurious space after ref.<br>
<br>
4) JSEP applications typically inform the browser to begin ICE gathering<br=
>
=C2=A0 =C2=A0via the information supplied to setLocalDescription, as this i=
s where<br>
=C2=A0 =C2=A0the app specifies the number of media streams to gather candid=
ates<br>
=C2=A0 =C2=A0for.<br>
<br>
I find the statement that ICE candidate gathering starts first with<br>
setLocalDescription. Doesn&#39;t the created offer already contain local<br=
>
candidates, thus one must have already gathered some candidates? To me<br>
it appears reasonable to start gathering earlier, rather than later.<br>
Thus, as soon as one create the PeerConnection object and have added any<br=
>
MST to it or a data channel, one can start gathering.<br>
<br>
5) Section 3.5.2:<br>
=C2=A0 =C2=A0In the parallel forking case where the Javascript application =
wishes<br>
=C2=A0 =C2=A0to simultaneously exchange media with multiple peers, the flow=
 is<br>
=C2=A0 =C2=A0slightly more complex, but the Javascript application can foll=
ow the<br>
=C2=A0 =C2=A0strategy that [RFC3960] describes using UPDATE. =C2=A0(It is w=
orth noting<br>
=C2=A0 =C2=A0that use cases where this is the desired behavior are very unu=
sual.)<br>
<br>
I mostly react to the last sentence within the paragraph. I think the<br>
poster child for semi-parallel forking was the MMORG that like to<br>
establish PCs with anyone that is coming close in the virtual world.<br>
Thus either needs to have a number of PCs on standby or have just one,<br>
but treat that as parallel forking. And if you like to discourage the<br>
later, then I think you might need to at least inform about the<br>
possibility to have stand-by PCs.<br>
<br>
6. Section 3.6:<br>
<br>
=C2=A0 =C2=A0In the event that the local application state is reinitialized=
,<br>
=C2=A0 =C2=A0either due to a user reload of the page, or a decision within =
the<br>
=C2=A0 =C2=A0application to reload itself (perhaps to update to a new versi=
on), it<br>
=C2=A0 =C2=A0is possible to keep an existing session alive, via a process c=
alled<br>
=C2=A0 =C2=A0&quot;rehydration&quot;. =C2=A0The explicit goal of rehydratio=
n is to carry out this<br>
=C2=A0 =C2=A0session resumption with no interaction with the remote side ot=
her<br>
=C2=A0 =C2=A0than normal call signaling messages.<br>
<br>
I think this paragraph makes two erronous statements:<br>
a) &quot;keep an existing session alive&quot;<br>
I think &quot;resume&quot; an existing session would be more correct. Becau=
se it<br>
would be broken when the reload happens.<br>
<br>
b) &quot;with no interaction with the remote side&quot;<br>
My understanding is that due to DTLS and key handling a DTLS<br>
renegotiation will be required with the peer. Isn&#39;t also a signalling<b=
r>
exchange likely to be required due to change of local DTLS cert as well<br>
as ICE candidates as parameters? Any other parameters that will be<br>
required to exchanges when rehydrating?<br>
<br>
7. Section 3.6:<br>
Next, a new offer is<br>
=C2=A0 =C2=A0generated by the new PeerConnection; this offer will have new =
ICE and<br>
=C2=A0 =C2=A0possibly new DTLS-SRTP certificate fingerprints (since the old=
 ICE<br>
=C2=A0 =C2=A0and SRTP state has been lost).<br>
<br>
I think this may require quite a bit more details. A forward pointer to<br>
where you will specify that?<br>
<br>
8. Section 3.6:<br>
=C2=A0 =C2=A0[OPEN ISSUE: EKR proposed an alternative rehydration approach =
where<br>
=C2=A0 =C2=A0the actual internal PeerConnection object in the browser was k=
ept<br>
=C2=A0 =C2=A0alive for some time after the web page was killed and provided=
 some<br>
=C2=A0 =C2=A0way for a new page to acquire the old PeerConnection object.]<=
br>
<br>
Can&#39;t we declare this open issue closed? I have seen no details on the<=
br>
alternative proposal. Thus, lets work with the one that is present.<br>
<br>
9. Section 4.1.1:<br>
There is no discussion of how the bundle policies relate to the Data<br>
Channels? Are they included or do they have a specific set of policies.<br>
Please extend this to discuss this.<br>
<br>
10. Section 4.1.3:<br>
<br>
This text doesn&#39;t appear to contain a requirement on that one MUST have=
<br>
called setRemote prior to being able to call it.<br>
<br>
11. Section <a href=3D"http://4.1.4.1" target=3D"_blank">4.1.4.1</a>:<br>
=C2=A0 =C2=A0While this could also be done with an inactive &quot;pranswer&=
quot;, followed<br>
=C2=A0 =C2=A0by a sendrecv &quot;answer&quot;, the initial &quot;pranswer&q=
uot; leaves the offer-<br>
=C2=A0 =C2=A0answer exchange open, which means the caller cannot send an up=
dated<br>
=C2=A0 =C2=A0offer during this time.<br>
<br>
Should one mention that also the callee can&#39;t send an offer without<br>
doing a &quot;final&quot; answer?<br>
<br>
12. Section 4.1.4.1<br>
<br>
By the time the human has accepted the call and<br>
=C2=A0 =C2=A0triggered the new offer, it is likely that the ICE and DTLS<br=
>
=C2=A0 =C2=A0handshaking for all the channels will already be set up.<br>
<br>
Wouldn&#39;t the handshaking have &quot;finished&quot; rather than &quot;se=
t up&quot;?<br>
<br>
13. Section <a href=3D"http://4.1.4.2" target=3D"_blank">4.1.4.2</a>:<br>
=C2=A0 =C2=A0Rollback can only be used to cancel proposed changes; there is=
 no<br>
=C2=A0 =C2=A0support for rolling back from a stable state to a previous sta=
ble<br>
=C2=A0 =C2=A0state.<br>
<br>
What is meant with &quot;proposed changes&quot;? From my side it is not cle=
ar how<br>
much you can do after having performed a SetLocal or SetRemote. I think<br>
these needs to be treated separately.<br>
<br>
For an offerer, they can createOffer, SetLocal, send to peer, receive<br>
reject and then undo the SetLocal. That is straight forward, and if they<br=
>
done the SetRemote, they would be consider in a new Stable state and<br>
can&#39;t roll back anymore?<br>
<br>
For the answering side, it receive an offer, does SetRemote,<br>
CreateAnswer, determines that it doesn&#39;t like it, the can undo the<br>
SetRemote and send a reject message to offerer.<br>
<br>
But what about this case? it receive an offer, does SetRemote,<br>
CreateAnswer, SetLocal and sends the message, then receive a reject from<br=
>
the peer, because it didn&#39;t like the answer? Are the answerer then in a=
<br>
stable state that can&#39;t be unrolled, or?<br>
<br>
14. Section 4.1.5:<br>
=C2=A0 =C2=A0This API indirectly controls the candidate gathering process. =
=C2=A0When a<br>
=C2=A0 =C2=A0local description is supplied, and the number of transports cu=
rrently<br>
=C2=A0 =C2=A0in use does not match the number of transports needed by the l=
ocal<br>
=C2=A0 =C2=A0description, the PeerConnection will create transports as need=
ed and<br>
=C2=A0 =C2=A0begin gathering candidates for them.<br>
<br>
Another part of an earlier question regarding what triggers ICE gathering?<=
br>
<br>
15. Section 4.1.7:<br>
=C2=A0 =C2=A0TODO: Do we need to expose accessors for both the current and<=
br>
=C2=A0 =C2=A0proposed local description?<br>
<br>
This needs to be resolved, I don&#39;t know if I can answer this question.<=
br>
<br>
16. Section 4.1.9:<br>
<br>
Should this section discuss that it might be browser that has overriding<br=
>
controls regarding if other candidates then relay ones may be provided,<br>
i.e. privacy mode?<br>
<br>
17. Section 4.1.9:<br>
=C2=A0 =C2=A0Regardless of the configuration, the gathering process collect=
s all<br>
=C2=A0 =C2=A0available candidates, but excluded candidates will not be surf=
aced in<br>
=C2=A0 =C2=A0onicecandidate callback or used for connectivity checks.<br>
<br>
Doesn&#39;t this result in both resource consumption and that the configure=
d<br>
TURN server will learn the external IP address and port for the client?<br>
<br>
18. Section 5.2.1:<br>
<br>
Structure in relation between RTP media and DATA channel. Maybe you<br>
should try to find a structure that makes the difference between the two<br=
>
types of transport more clear, and what applies to the perspective.<br>
<br>
19. Section 5.2.1:<br>
<br>
<br>
=C2=A0 =C2=A0o =C2=A0For the current default candidate, a &quot;c=3D&quot; =
line, as specific in<br>
=C2=A0 =C2=A0 =C2=A0 [RFC5245], Section 4.3., paragraph 6. =C2=A0If no cand=
idates have been<br>
=C2=A0 =C2=A0 =C2=A0 gathered yet, the default candidate should be set to t=
he &#39;null&#39;<br>
=C2=A0 =C2=A0 =C2=A0 value defined in [I-D.ietf-mmusic-trickle-ice], Sectio=
n 5.1.<br>
<br>
What about host candidates?<br>
<br>
20. Section 5.2.1:<br>
The CNAME must be generated<br>
=C2=A0 =C2=A0 =C2=A0 in accordance with draft-rescorla-random-cname-00. =C2=
=A0[OPEN ISSUE:<br>
=C2=A0 =C2=A0 =C2=A0 How are CNAMEs specified for MSTs? =C2=A0Are they rand=
omly generated<br>
=C2=A0 =C2=A0 =C2=A0 for each MediaStream? =C2=A0If so, can two MediaStream=
s be synced?]<br>
<br>
I think you should point to the RTP usage for this particular piece.<br>
Because we do have text there both for the CNAME generation and how<br>
CNAMES among MSTs and between PeerConnection. Likely to be changed soon<br>
based on the mailing list discussion.<br>
<br>
21. Section 5.2.1:<br>
=C2=A0 =C2=A0o =C2=A0[OPEN ISSUE: Handling of a=3Dimageattr]<br>
<br>
Yes, I think imageattr should be supported to give guidance on suitable<br>
resolutions that an endpoint like to receive for a particular MST.<br>
<br>
22. Section 5.2.1:<br>
=C2=A0 =C2=A0Once all m=3D sections have been generated, a session-level &q=
uot;a=3Dgroup&quot;<br>
=C2=A0 =C2=A0attribute MUST be added as specified in [RFC5888]. =C2=A0This =
attribute<br>
=C2=A0 =C2=A0MUST have semantics &quot;BUNDLE&quot;, and MUST include the m=
id identifiers of<br>
=C2=A0 =C2=A0each m=3D section.<br>
<br>
Isn&#39;t this overriding BUNDLE as currently specified, forcing everything=
<br>
into a single bundle group. Thus, preventing the bundle policies to work.<b=
r>
<br>
23. Section 5.2.2:<br>
<br>
=C2=A0 =C2=A0o =C2=A0If any MediaStreamTracks have been added, and there ex=
ist m=3D<br>
=C2=A0 =C2=A0 =C2=A0 sections of the appropriate media type with no associa=
ted<br>
=C2=A0 =C2=A0 =C2=A0 MediaStreamTracks<br>
<br>
Isnt&#39; the last a &quot;MediaStreamTrack&quot; rather than plural ones?<=
br>
<br>
24. 5.2.3.3. =C2=A0VoiceActivityDetection<br>
<br>
=C2=A0 =C2=A0If the &quot;VoiceActivityDetection&quot; constraint is specif=
ied, with a value<br>
=C2=A0 =C2=A0of &quot;true&quot;, the offer MUST indicate support for silen=
ce suppression by<br>
=C2=A0 =C2=A0including comfort noise (&quot;CN&quot;) codecs for each suppo=
rted clock rate,<br>
=C2=A0 =C2=A0as specified in [RFC3389], Section 5.1.<br>
<br>
As currently specified this text requires the usage of a comfort noise<br>
codec, currently there is no such on required. I think the &quot;MUST&quot;=
 needs<br>
to be rewritten.<br>
<br>
25. Section 5.3.1:<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 Note<br>
=C2=A0 =C2=A0 =C2=A0 that for simplicity, the answerer MAY use different pa=
yload types<br>
=C2=A0 =C2=A0 =C2=A0 for codecs than the offerer, as it is not prohibited b=
y<br>
=C2=A0 =C2=A0 =C2=A0 Section 6.1.<br>
<br>
I react to the word simplicity here. I never consider usage of different<br=
>
PTs per direction anything simple. But, yes it is allowed by RFC 3264.<br>
But I kind of protest to encourage the usage.<br>
<br>
26. Section 5.3.2 etc.<br>
<br>
When will we see a first draft of the text in the sections:<br>
5.3.2, 5.3.3., 5.4, 5.5, 5.6, 5.7?<br>
<br>
I really like to see it prior to the Interim meeting.<br>
<br>
27. Section 6:<br>
=C2=A0 =C2=A0Note: This section is still very early and is likely to signif=
icantly<br>
=C2=A0 =C2=A0change as we get a better understanding of a) the use cases fo=
r this<br>
=C2=A0 =C2=A0b) the implications at the protocol level c) feedback from<br>
=C2=A0 =C2=A0implementors on what they can do.<br>
<br>
Shouldn&#39;t this mention W3C discussion also?<br>
<br>
28. Section 7 - Security Considerations:<br>
<br>
First of all there are no references to the Security Considerations or<br>
architecture document.<br>
<br>
Secondly, I would like to have some security discussion on significant<br>
issues with JSEP itself. Resource attacks on the local endpoint for<br>
example?<br>
<br>
29. section 10<br>
<br>
There are a couple of draft references that are either published RFCs or<br=
>
at least WG versions. Please update the reference list.<br>
<br>
30. Section 10.2:<br>
Looking at the spec, I think the following references are normative:<br>
<br>
=C2=A0 =C2=A0[I-D.ietf-mmusic-trickle-ice]<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Ivov, E., Rescorla, E., an=
d J. Uberti, &quot;Trickle ICE:<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Incremental Provisioning o=
f Candidates for the Interactive<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Connectivity Establishment=
 (ICE) Protocol&quot;, draft-ietf-<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 mmusic-trickle-ice-00 (wor=
k in progress), March 2013.<br>
<br>
=C2=A0 =C2=A0[I-D.ietf-rtcweb-data-protocol]<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Jesup, R., Loreto, S., and=
 M. Tuexen, &quot;WebRTC Data Channel<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Protocol&quot;, draft-ietf=
-rtcweb-data-protocol-04 (work in<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 progress), February 2013.<=
br>
<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0+46 10 7148287<br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile +46 73 0949079<br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

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Date: Fri, 14 Mar 2014 09:53:24 -0700
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From: Watson Ladd <watsonbladd@gmail.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On Mar 14, 2014 9:44 AM, "Justin Uberti" <juberti@google.com> wrote:
>
>
>
>
> On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:
>>
>> On 2014-03-13 17:53, Martin Thomson wrote:
>> > On 12 March 2014 01:29, Magnus Westerlund
>> > <magnus.westerlund@ericsson.com> wrote:
>> >> I like to point out that the agreement and what is documented in
>> >> rtp-usage is that WebRTC endpoint will have to make forwarded streams
>> >> appear as locally originated. However, this as currently written does
>> >> not apply to RTP middleboxes that interconnects multiple
PeerConnections
>> >> to form a multi-party session. This is deliberate to ensure that RTP
>> >> topologies like RTP mixer and SFM do work on the RTP/RTCP level.
>> >
>> > I'm still not clear on whether a middlebox interoperating with a
>> > WebRTC endpoint is obligated to respect these rules or not.  I had
>> > assumed that WebRTC is such a bully that they would be forced to.
>>
>> Well, I know Colin and I did put in quite some thought to make the
>> WebRTC RTP usage rules such that you can use the common RTP middleboxes
>> as long as you have the right front-end, i.e. DTLS-SRTP and signalling
>> translation.
>>
>> >
>> >> I am especially interested to know how you will "easy to apply
manually"
>> >> the synchronization. Can you please describe that. Because, that
either
>> >> requires an API call to tell the media framework, please consider the
>> >> following CNAMES as equivalent, or some other method of telling the
>> >> media framework that these different MST are actually originating
from a
>> >> common clock base.
>> > In WebRTC, this would be:
>> >
>> > var audio = pc1.getRemoteStreams()[0].getAudioTracks()[0];
>> > var video = pc2.getRemoteStreams()[0].getVideoTracks()[0];
>> > var synchronizedStream = new MediaStream(audio, video);
>> >
>> > We'd have to say that this implies a statement about the clock base of
>> > the tracks being the same.  ... and that this statement could be
>> > wrong.
>>
>> I think this looks dangerous. If you default the assumption that
>> different CNAMEs of the MST you add are actually sharing clock base,
>> then a lot of basic programs that may group MST from different PCs into
>> one MS for convenience are going to cause serious issue in the media
>> frameworks, when they attempt to synchronize things.
>>
>> >
>> > As far as it goes, I'm sure that other systems could build a similar
>> > function, but none of those are standardized.
>> >
>> >> I protested about this, not to lower the users privacy, but over the
>> >> concern that this was raised without providing a case where it was
>> >> obvious that the user's privacy was impacted. Martin, you said that
you
>> >> would think about it, and the next statement was lets change it,
without
>> >> providing any motivation why the concern was significant.
>> >
>> > I'm sorry about that, I thought about it, but didn't want to waste
>> > everyone's time (mine included) with an essay on the subject.
>> >
>>
>> Having considered this a bit more, I do think the risk to privacy out
>> weighs the other concerns, as long as the API and its handling can be
>> sorted out, I don't think the above is sufficient to get good
>> functionality.
>>
>> My understanding is that the risk to privacy exist when one have one JS
>> application enabling communication in different contexts during the same
>> application session. In cases where there might be participants in the
>>
>> different contexts that could link the same end-point and thus human
>> user to different participant IDs (that otherwise are anonymous). Thus
>> revealing privacy concerns. As it would take the application designer to
>> take this risk into consideration to avoid it, I feel safer if the
>> application designer would need to take active measurements to perform
>> linkage.
>>
>>
>> I will wait an see if there are other inputs on this within the next
>> week. If nothing arises I will propose a text change to the rtp-usage to
>> address this.
>>
>
> At an implementation level, one could imagine at least 3 policies for
generating CNAMEs:
> a) per-session (i.e. per-PeerConnection)
> b) per-page (i.e. shared between all PCs on a page)
> c) per-page, persistent (i.e. shared between all PCs on a page, including
across page loads)
>
> While we seem to agree that a) is the right solution for CNAMEs, it is
worth pointing out that we (Chrome) are currently doing c) for DTLS
certificates, to avoid performance problems with cert generation at page
load. Ergo, this linkability concern already exists, and I don't think it
is easy to solve it in the default case. There have been some proposals to
allow generation/storage of unique certs to prevent this linkability, but
this will require app input.
>

One exponentiation, fixed base, per certificate. Use an addition
multichain. Cert generation is cheap with ECDSA.

Sincerely,
Watson Ladd
> Ergo, we might want to match the DTLS behavior (i.e. generate unique
CNAMEs only when the certs are unique), to ensure we treat linkability
consistently.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--485b397dce3376a90504f493e84c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr"><br>
On Mar 14, 2014 9:44 AM, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto:ju=
berti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund &lt;<a href=3D"mail=
to:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a>&gt; w=
rote:<br>
&gt;&gt;<br>
&gt;&gt; On 2014-03-13 17:53, Martin Thomson wrote:<br>
&gt;&gt; &gt; On 12 March 2014 01:29, Magnus Westerlund<br>
&gt;&gt; &gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.=
westerlund@ericsson.com</a>&gt; wrote:<br>
&gt;&gt; &gt;&gt; I like to point out that the agreement and what is docume=
nted in<br>
&gt;&gt; &gt;&gt; rtp-usage is that WebRTC endpoint will have to make forwa=
rded streams<br>
&gt;&gt; &gt;&gt; appear as locally originated. However, this as currently =
written does<br>
&gt;&gt; &gt;&gt; not apply to RTP middleboxes that interconnects multiple =
PeerConnections<br>
&gt;&gt; &gt;&gt; to form a multi-party session. This is deliberate to ensu=
re that RTP<br>
&gt;&gt; &gt;&gt; topologies like RTP mixer and SFM do work on the RTP/RTCP=
 level.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I&#39;m still not clear on whether a middlebox interoperating=
 with a<br>
&gt;&gt; &gt; WebRTC endpoint is obligated to respect these rules or not. =
=C2=A0I had<br>
&gt;&gt; &gt; assumed that WebRTC is such a bully that they would be forced=
 to.<br>
&gt;&gt;<br>
&gt;&gt; Well, I know Colin and I did put in quite some thought to make the=
<br>
&gt;&gt; WebRTC RTP usage rules such that you can use the common RTP middle=
boxes<br>
&gt;&gt; as long as you have the right front-end, i.e. DTLS-SRTP and signal=
ling<br>
&gt;&gt; translation.<br>
&gt;&gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&gt; I am especially interested to know how you will &quot;eas=
y to apply manually&quot;<br>
&gt;&gt; &gt;&gt; the synchronization. Can you please describe that. Becaus=
e, that either<br>
&gt;&gt; &gt;&gt; requires an API call to tell the media framework, please =
consider the<br>
&gt;&gt; &gt;&gt; following CNAMES as equivalent, or some other method of t=
elling the<br>
&gt;&gt; &gt;&gt; media framework that these different MST are actually ori=
ginating from a<br>
&gt;&gt; &gt;&gt; common clock base.<br>
&gt;&gt; &gt; In WebRTC, this would be:<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; var audio =3D pc1.getRemoteStreams()[0].getAudioTracks()[0];<=
br>
&gt;&gt; &gt; var video =3D pc2.getRemoteStreams()[0].getVideoTracks()[0];<=
br>
&gt;&gt; &gt; var synchronizedStream =3D new MediaStream(audio, video);<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; We&#39;d have to say that this implies a statement about the =
clock base of<br>
&gt;&gt; &gt; the tracks being the same. =C2=A0... and that this statement =
could be<br>
&gt;&gt; &gt; wrong.<br>
&gt;&gt;<br>
&gt;&gt; I think this looks dangerous. If you default the assumption that<b=
r>
&gt;&gt; different CNAMEs of the MST you add are actually sharing clock bas=
e,<br>
&gt;&gt; then a lot of basic programs that may group MST from different PCs=
 into<br>
&gt;&gt; one MS for convenience are going to cause serious issue in the med=
ia<br>
&gt;&gt; frameworks, when they attempt to synchronize things.<br>
&gt;&gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; As far as it goes, I&#39;m sure that other systems could buil=
d a similar<br>
&gt;&gt; &gt; function, but none of those are standardized.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&gt; I protested about this, not to lower the users privacy, b=
ut over the<br>
&gt;&gt; &gt;&gt; concern that this was raised without providing a case whe=
re it was<br>
&gt;&gt; &gt;&gt; obvious that the user&#39;s privacy was impacted. Martin,=
 you said that you<br>
&gt;&gt; &gt;&gt; would think about it, and the next statement was lets cha=
nge it, without<br>
&gt;&gt; &gt;&gt; providing any motivation why the concern was significant.=
<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I&#39;m sorry about that, I thought about it, but didn&#39;t =
want to waste<br>
&gt;&gt; &gt; everyone&#39;s time (mine included) with an essay on the subj=
ect.<br>
&gt;&gt; &gt;<br>
&gt;&gt;<br>
&gt;&gt; Having considered this a bit more, I do think the risk to privacy =
out<br>
&gt;&gt; weighs the other concerns, as long as the API and its handling can=
 be<br>
&gt;&gt; sorted out, I don&#39;t think the above is sufficient to get good<=
br>
&gt;&gt; functionality.<br>
&gt;&gt;<br>
&gt;&gt; My understanding is that the risk to privacy exist when one have o=
ne JS<br>
&gt;&gt; application enabling communication in different contexts during th=
e same<br>
&gt;&gt; application session. In cases where there might be participants in=
 the<br>
&gt;&gt;<br>
&gt;&gt; different contexts that could link the same end-point and thus hum=
an<br>
&gt;&gt; user to different participant IDs (that otherwise are anonymous). =
Thus<br>
&gt;&gt; revealing privacy concerns. As it would take the application desig=
ner to<br>
&gt;&gt; take this risk into consideration to avoid it, I feel safer if the=
<br>
&gt;&gt; application designer would need to take active measurements to per=
form<br>
&gt;&gt; linkage.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; I will wait an see if there are other inputs on this within the ne=
xt<br>
&gt;&gt; week. If nothing arises I will propose a text change to the rtp-us=
age to<br>
&gt;&gt; address this.<br>
&gt;&gt;<br>
&gt;<br>
&gt; At an implementation level, one could imagine at least 3 policies for =
generating CNAMEs:<br>
&gt; a) per-session (i.e. per-PeerConnection)<br>
&gt; b) per-page (i.e. shared between all PCs on a page)<br>
&gt; c) per-page, persistent (i.e. shared between all PCs on a page, includ=
ing across page loads)<br>
&gt;<br>
&gt; While we seem to agree that a) is the right solution for CNAMEs, it is=
 worth pointing out that we (Chrome) are currently doing c) for DTLS certif=
icates, to avoid performance problems with cert generation at page load. Er=
go, this linkability concern already exists, and I don&#39;t think it is ea=
sy to solve it in the default case. There have been some proposals to allow=
 generation/storage of unique certs to prevent this linkability, but this w=
ill require app input.<br>

&gt;</p>
<p dir=3D"ltr">One exponentiation, fixed base, per certificate. Use an addi=
tion multichain. Cert generation is cheap with ECDSA.</p>
<p dir=3D"ltr">Sincerely, <br>
Watson Ladd<br>
&gt; Ergo, we might want to match the DTLS behavior (i.e. generate unique C=
NAMEs only when the certs are unique), to ensure we treat linkability consi=
stently.<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
</p>

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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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On 14 March 2014 09:44, Justin Uberti <juberti@google.com> wrote:
> While we seem to agree that a) is the right solution for CNAMEs, it is worth
> pointing out that we (Chrome) are currently doing c) for DTLS certificates,
> to avoid performance problems with cert generation at page load.

Firefox does 1-per PC.  There are plans to allow more granular control
over this (following ekr's proposal), but that's currently low
priority.  There are plenty of other places where we can make
performance improvements.


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From: Justin Uberti <juberti@google.com>
Date: Fri, 14 Mar 2014 10:32:02 -0700
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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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--bcaec52997451c4dbd04f4948f03
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On Fri, Mar 14, 2014 at 9:53 AM, Watson Ladd <watsonbladd@gmail.com> wrote:

>
> On Mar 14, 2014 9:44 AM, "Justin Uberti" <juberti@google.com> wrote:
> >
> >
> >
> >
> > On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
> >>
> >> On 2014-03-13 17:53, Martin Thomson wrote:
> >> > On 12 March 2014 01:29, Magnus Westerlund
> >> > <magnus.westerlund@ericsson.com> wrote:
> >> >> I like to point out that the agreement and what is documented in
> >> >> rtp-usage is that WebRTC endpoint will have to make forwarded streams
> >> >> appear as locally originated. However, this as currently written does
> >> >> not apply to RTP middleboxes that interconnects multiple
> PeerConnections
> >> >> to form a multi-party session. This is deliberate to ensure that RTP
> >> >> topologies like RTP mixer and SFM do work on the RTP/RTCP level.
> >> >
> >> > I'm still not clear on whether a middlebox interoperating with a
> >> > WebRTC endpoint is obligated to respect these rules or not.  I had
> >> > assumed that WebRTC is such a bully that they would be forced to.
> >>
> >> Well, I know Colin and I did put in quite some thought to make the
> >> WebRTC RTP usage rules such that you can use the common RTP middleboxes
> >> as long as you have the right front-end, i.e. DTLS-SRTP and signalling
> >> translation.
> >>
> >> >
> >> >> I am especially interested to know how you will "easy to apply
> manually"
> >> >> the synchronization. Can you please describe that. Because, that
> either
> >> >> requires an API call to tell the media framework, please consider the
> >> >> following CNAMES as equivalent, or some other method of telling the
> >> >> media framework that these different MST are actually originating
> from a
> >> >> common clock base.
> >> > In WebRTC, this would be:
> >> >
> >> > var audio = pc1.getRemoteStreams()[0].getAudioTracks()[0];
> >> > var video = pc2.getRemoteStreams()[0].getVideoTracks()[0];
> >> > var synchronizedStream = new MediaStream(audio, video);
> >> >
> >> > We'd have to say that this implies a statement about the clock base of
> >> > the tracks being the same.  ... and that this statement could be
> >> > wrong.
> >>
> >> I think this looks dangerous. If you default the assumption that
> >> different CNAMEs of the MST you add are actually sharing clock base,
> >> then a lot of basic programs that may group MST from different PCs into
> >> one MS for convenience are going to cause serious issue in the media
> >> frameworks, when they attempt to synchronize things.
> >>
> >> >
> >> > As far as it goes, I'm sure that other systems could build a similar
> >> > function, but none of those are standardized.
> >> >
> >> >> I protested about this, not to lower the users privacy, but over the
> >> >> concern that this was raised without providing a case where it was
> >> >> obvious that the user's privacy was impacted. Martin, you said that
> you
> >> >> would think about it, and the next statement was lets change it,
> without
> >> >> providing any motivation why the concern was significant.
> >> >
> >> > I'm sorry about that, I thought about it, but didn't want to waste
> >> > everyone's time (mine included) with an essay on the subject.
> >> >
> >>
> >> Having considered this a bit more, I do think the risk to privacy out
> >> weighs the other concerns, as long as the API and its handling can be
> >> sorted out, I don't think the above is sufficient to get good
> >> functionality.
> >>
> >> My understanding is that the risk to privacy exist when one have one JS
> >> application enabling communication in different contexts during the same
> >> application session. In cases where there might be participants in the
> >>
> >> different contexts that could link the same end-point and thus human
> >> user to different participant IDs (that otherwise are anonymous). Thus
> >> revealing privacy concerns. As it would take the application designer to
> >> take this risk into consideration to avoid it, I feel safer if the
> >> application designer would need to take active measurements to perform
> >> linkage.
> >>
> >>
> >> I will wait an see if there are other inputs on this within the next
> >> week. If nothing arises I will propose a text change to the rtp-usage to
> >> address this.
> >>
> >
> > At an implementation level, one could imagine at least 3 policies for
> generating CNAMEs:
> > a) per-session (i.e. per-PeerConnection)
> > b) per-page (i.e. shared between all PCs on a page)
> > c) per-page, persistent (i.e. shared between all PCs on a page,
> including across page loads)
> >
> > While we seem to agree that a) is the right solution for CNAMEs, it is
> worth pointing out that we (Chrome) are currently doing c) for DTLS
> certificates, to avoid performance problems with cert generation at page
> load. Ergo, this linkability concern already exists, and I don't think it
> is easy to solve it in the default case. There have been some proposals to
> allow generation/storage of unique certs to prevent this linkability, but
> this will require app input.
> >
>
> One exponentiation, fixed base, per certificate. Use an addition
> multichain. Cert generation is cheap with ECDSA.
>
You're going to have to explain a bit more for me to grok this.

Also, define 'cheap', especially on ARM CPUs. Currently RSA generation on
best-in-class ARM CPUs is 500 ms for 1024-bit, 2 seconds for 2048-bit. That
is a problem.

--bcaec52997451c4dbd04f4948f03
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Mar 14, 2014 at 9:53 AM, Watson Ladd <span dir=3D"ltr">&lt;=
<a href=3D"mailto:watsonbladd@gmail.com" target=3D"_blank">watsonbladd@gmai=
l.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div><p dir=3D"ltr"><br>
On Mar 14, 2014 9:44 AM, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto:ju=
berti@google.com" target=3D"_blank">juberti@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund &lt;<a href=3D"mail=
to:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerlund@eric=
sson.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On 2014-03-13 17:53, Martin Thomson wrote:<br>
&gt;&gt; &gt; On 12 March 2014 01:29, Magnus Westerlund<br>
&gt;&gt; &gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=
=3D"_blank">magnus.westerlund@ericsson.com</a>&gt; wrote:<br>
&gt;&gt; &gt;&gt; I like to point out that the agreement and what is docume=
nted in<br>
&gt;&gt; &gt;&gt; rtp-usage is that WebRTC endpoint will have to make forwa=
rded streams<br>
&gt;&gt; &gt;&gt; appear as locally originated. However, this as currently =
written does<br>
&gt;&gt; &gt;&gt; not apply to RTP middleboxes that interconnects multiple =
PeerConnections<br>
&gt;&gt; &gt;&gt; to form a multi-party session. This is deliberate to ensu=
re that RTP<br>
&gt;&gt; &gt;&gt; topologies like RTP mixer and SFM do work on the RTP/RTCP=
 level.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I&#39;m still not clear on whether a middlebox interoperating=
 with a<br>
&gt;&gt; &gt; WebRTC endpoint is obligated to respect these rules or not. =
=C2=A0I had<br>
&gt;&gt; &gt; assumed that WebRTC is such a bully that they would be forced=
 to.<br>
&gt;&gt;<br>
&gt;&gt; Well, I know Colin and I did put in quite some thought to make the=
<br>
&gt;&gt; WebRTC RTP usage rules such that you can use the common RTP middle=
boxes<br>
&gt;&gt; as long as you have the right front-end, i.e. DTLS-SRTP and signal=
ling<br>
&gt;&gt; translation.<br>
&gt;&gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&gt; I am especially interested to know how you will &quot;eas=
y to apply manually&quot;<br>
&gt;&gt; &gt;&gt; the synchronization. Can you please describe that. Becaus=
e, that either<br>
&gt;&gt; &gt;&gt; requires an API call to tell the media framework, please =
consider the<br>
&gt;&gt; &gt;&gt; following CNAMES as equivalent, or some other method of t=
elling the<br>
&gt;&gt; &gt;&gt; media framework that these different MST are actually ori=
ginating from a<br>
&gt;&gt; &gt;&gt; common clock base.<br>
&gt;&gt; &gt; In WebRTC, this would be:<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; var audio =3D pc1.getRemoteStreams()[0].getAudioTracks()[0];<=
br>
&gt;&gt; &gt; var video =3D pc2.getRemoteStreams()[0].getVideoTracks()[0];<=
br>
&gt;&gt; &gt; var synchronizedStream =3D new MediaStream(audio, video);<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; We&#39;d have to say that this implies a statement about the =
clock base of<br>
&gt;&gt; &gt; the tracks being the same. =C2=A0... and that this statement =
could be<br>
&gt;&gt; &gt; wrong.<br>
&gt;&gt;<br>
&gt;&gt; I think this looks dangerous. If you default the assumption that<b=
r>
&gt;&gt; different CNAMEs of the MST you add are actually sharing clock bas=
e,<br>
&gt;&gt; then a lot of basic programs that may group MST from different PCs=
 into<br>
&gt;&gt; one MS for convenience are going to cause serious issue in the med=
ia<br>
&gt;&gt; frameworks, when they attempt to synchronize things.<br>
&gt;&gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; As far as it goes, I&#39;m sure that other systems could buil=
d a similar<br>
&gt;&gt; &gt; function, but none of those are standardized.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&gt; I protested about this, not to lower the users privacy, b=
ut over the<br>
&gt;&gt; &gt;&gt; concern that this was raised without providing a case whe=
re it was<br>
&gt;&gt; &gt;&gt; obvious that the user&#39;s privacy was impacted. Martin,=
 you said that you<br>
&gt;&gt; &gt;&gt; would think about it, and the next statement was lets cha=
nge it, without<br>
&gt;&gt; &gt;&gt; providing any motivation why the concern was significant.=
<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I&#39;m sorry about that, I thought about it, but didn&#39;t =
want to waste<br>
&gt;&gt; &gt; everyone&#39;s time (mine included) with an essay on the subj=
ect.<br>
&gt;&gt; &gt;<br>
&gt;&gt;<br>
&gt;&gt; Having considered this a bit more, I do think the risk to privacy =
out<br>
&gt;&gt; weighs the other concerns, as long as the API and its handling can=
 be<br>
&gt;&gt; sorted out, I don&#39;t think the above is sufficient to get good<=
br>
&gt;&gt; functionality.<br>
&gt;&gt;<br>
&gt;&gt; My understanding is that the risk to privacy exist when one have o=
ne JS<br>
&gt;&gt; application enabling communication in different contexts during th=
e same<br>
&gt;&gt; application session. In cases where there might be participants in=
 the<br>
&gt;&gt;<br>
&gt;&gt; different contexts that could link the same end-point and thus hum=
an<br>
&gt;&gt; user to different participant IDs (that otherwise are anonymous). =
Thus<br>
&gt;&gt; revealing privacy concerns. As it would take the application desig=
ner to<br>
&gt;&gt; take this risk into consideration to avoid it, I feel safer if the=
<br>
&gt;&gt; application designer would need to take active measurements to per=
form<br>
&gt;&gt; linkage.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; I will wait an see if there are other inputs on this within the ne=
xt<br>
&gt;&gt; week. If nothing arises I will propose a text change to the rtp-us=
age to<br>
&gt;&gt; address this.<br>
&gt;&gt;<br>
&gt;<br>
&gt; At an implementation level, one could imagine at least 3 policies for =
generating CNAMEs:<br>
&gt; a) per-session (i.e. per-PeerConnection)<br>
&gt; b) per-page (i.e. shared between all PCs on a page)<br>
&gt; c) per-page, persistent (i.e. shared between all PCs on a page, includ=
ing across page loads)<br>
&gt;<br>
&gt; While we seem to agree that a) is the right solution for CNAMEs, it is=
 worth pointing out that we (Chrome) are currently doing c) for DTLS certif=
icates, to avoid performance problems with cert generation at page load. Er=
go, this linkability concern already exists, and I don&#39;t think it is ea=
sy to solve it in the default case. There have been some proposals to allow=
 generation/storage of unique certs to prevent this linkability, but this w=
ill require app input.<br>




&gt;</p>
</div></div><p dir=3D"ltr">One exponentiation, fixed base, per certificate.=
 Use an addition multichain. Cert generation is cheap with ECDSA.</p></bloc=
kquote><div>You&#39;re going to have to explain a bit more for me to grok t=
his.</div>


<div><br></div><div>Also, define &#39;cheap&#39;, especially on ARM CPUs. C=
urrently RSA generation on best-in-class ARM CPUs is 500 ms for 1024-bit, 2=
 seconds for 2048-bit. That is a problem.<br></div></div><br></div></div>


--bcaec52997451c4dbd04f4948f03--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

> By "RTCP transport in the offer", I mean separate ICE candidates generate=
d for a RTCP ICE component, the original problem that I was proposing to so=
lve.

Ok.

But, again, we do have a decision that we are not going to specify procedur=
es based on pre-knoweldge of BUNDLE support by the answerer. Or, at least t=
hat was the outcome in London, I guess it still has to be verified on the l=
ist.

> I agree the rtcp-mux attribute must always be present if you are going to=
 do rtcp-mux.

Good :)

Regards,

Christer


On Fri, Mar 14, 2014 at 9:30 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:

Hi Justin,

What do you mean by "RTCP transport in the offer"? Are you referring to the=
 a=3Drtcp attribute, or?

Even if we mandate rtcp-mux within a BUNDLE group, the a=3Drtcp-mux attribu=
te must still always be present - even if you know the remote endpoint supp=
orts rtcp-mux (or, even if you have already negotiated usage of rtcp-mux).

Regards,

Christer
________________________________________
From: Justin Uberti [juberti@google.com<mailto:juberti@google.com>]
Sent: Friday, 14 March 2014 6:26 PM
To: Magnus Westerlund
Cc: Christer Holmberg; rtcweb@ietf.org<mailto:rtcweb@ietf.org>; Eric Rescor=
la
Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux

On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund <magnus.westerlund@erics=
son.com<mailto:magnus.westerlund@ericsson.com><mailto:magnus.westerlund@eri=
csson.com<mailto:magnus.westerlund@ericsson.com>>> wrote:
Hi,
(as individual)

Although this discussion really should be on MMUSIC, I am still
following up here. Anything we come to conclude here really need to go
to MMUSIC for confirmation.

Justin, I do agree that it appear unnecessarily of having to include an
RTCP transport in an OFFER when the offerer know the answerer is BUNDLE
capable. I am fine with that and think the rule needed here is something
along the line of the below:

A BUNDLE supporting endpoint MUST implement a=3Drtcp-mux (that is already
required). In any answer by a BUNDLE capable endpoint, it MUST NOT
change the usage of a=3Drtcp-mux. If it is present in the offer, it must
be confirmed to be used in the answer, and if not present in the offer,
it MUST NOT be added in the answer. An BUNDLE capable endpoint is
RECOMMENDED to use a=3Drtcp-mux whenever suitable. Cases when it might not
be suitable include, lack RTP payload types, or cases when RTCP traffic
is needed to be on its own transport flow (5-tuple) to enable QoS or
other traffic handling functionalities.

The reason for writing the rule like the above, is that then an endpoint
can choose when creating an offer to renegotiate the use of a=3Drtcp-mux.
For example due to it running out of available PTs.

Feedback on this proposal?

I am not OK with this. It just gets too complicated to deal with all the ca=
ses, especially if with the provision that rtcp-mux can change during the s=
ession.

I think Christer's current language is what we want - if you are going to B=
UNDLE, you MUST also use rtcp-mux for any RTP m-lines. If the answerer cann=
ot do that, it needs to reject the BUNDLE.

I agree with your initial point though - if you know that both sides are go=
ing to BUNDLE and rtcp-mux, it makes sense to allow them to do that from th=
e outset, i.e. no need to include a RTCP transport in the offer.


From nobody Fri Mar 14 10:57:13 2014
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Date: Fri, 14 Mar 2014 10:56:39 -0700
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Subject: Re: [rtcweb] Should we reference the pause/resume I-D?
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Justin said:

"Overall this feels to me like something that needs some thought to figure
out all the interactions between the API, the signaling plane, and the
media plane. As such I am inclined to leave this out of 1.0."

[BA] I agree.


"It's not clear to me what the interactions are between the signaling and
media plane when this occurs. That is, if I do doohickey.active =3D false,
should that cause a=3Drecvonly signaling and a TMMBN notification that the
track is paused? I am concerned that things can get out of sync (i.e. what
happens if we get TMMBR indicating resume, but signaling is still
a=3Drecvonly)?"

[BA] A few thoughts:

a. Even though TMMBR/TMMBN is mandatory-to-implement in the RTP usage
document, I don't think we can assume that the pause/resume will be
supported by all mixers.. It therefore probably isn't a good idea for a
browser to automatically send a TMMBN 0 indication when setting
rtpsender.active =3D false.  Since the application might or might not use S=
DP
for signaling, this is probably something that needs to be under
application control.  BTW, Section 6.3, isn't clear about the response to
an unsolicited TMMBN (TMMBR 0?).

b. As described in Section 1, pause/resume is for use in scenarios where
signaling might not meet the performance requirements.  Due to the
differing transport characteristics of RTCP and signaling over reliable
transport, trying to use both pause/resume and signaling is quite likely to
result in them being out of sync, so my assumption was that an application
wishing to support pause/resume would not choose to use signaling for the
same purpose.   There is also the potential for conflict between
codec-specific messages (e.g. layer drop in H.264/SVC) and some potential
uses of pause/resume (e.g. unsolicited TMMBN to indicate layer pause in
multi-stream SVC).  Where there are conflicts, my assumption would be that
signaling and codec-specific messages should win.


On Fri, Mar 14, 2014 at 9:15 AM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Fri, Mar 14, 2014 at 4:25 AM, Stefan H=E5kansson LK <
> stefan.lk.hakansson@ericsson.com> wrote:
>
>> On 2014-03-12 21:56, Justin Uberti wrote:
>> > Agreed.
>>
>> I may well be that the pause/resume doc will not be ready in time, so it
>> from that aspect is a bad idea. What about just stating that TMMBN/TMMBR
>> 0 meaning pause should be supported?
>>
>
> It's not clear to me what the interactions are between the signaling and
> media plane when this occurs. That is, if I do doohickey.active =3D false=
,
> should that cause a=3Drecvonly signaling and a TMMBN notification that th=
e
> track is paused? I am concerned that things can get out of sync (i.e. wha=
t
> happens if we get TMMBR indicating resume, but signaling is still
> a=3Drecvonly)?
>
>
>> > Aren't there also patent declarations against this doc from
>> > multiple holders?
>>
>> I've not looked into that, but I deliberately chose to propose the "old"
>> TMMBN 0 way to signal that is mentioned in [1] (and not any of the "new"
>> signaling in [1]).
>>
>
> OK.
>
>
>>
>> >
>> > While SDP will likely be removed from the API in the future, the
>> > replacement would be a app-specific message sent over websockets, whic=
h
>> > seems like it would work just fine.
>>
>> s/would/could/: I don't think this has been discussed or agreed (and
>> websocket is of course just an example).
>>
>
> Right, the whole thing is an example, but the point is that the new API
> directions don't really affect this particular problem.
>
>>
>> I think it is worthwhile to push this kind of signaling into the media
>> plane if possible. For example, this would enable the UA to save
>> transmission/battery even if the app is badly written.
>>
>> As an example: if a MST sent over a PC is not used in any way at the
>> receiving end, the receiving end UA could signal TMMBR 0 to tell the
>> sending UA to not send. Sure, the receiving UA could also fire a
>> "negotiationneeded" event, but maybe the app doesn't listen.
>>
>
> True, if the app doesn't perform the necessary signaling, things are not
> going to work, but that applies to a lot of operations on the
> PeerConnection.
>
> Overall this feels to me like something that needs some thought to figure
> out all the interactions between the API, the signaling plane, and the
> media plane. As such I am inclined to leave this out of 1.0.
>
>>
>> >
>> >
>> > On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba <
>> bernard.aboba@gmail.com
>> > <mailto:bernard.aboba@gmail.com>> wrote:
>> >
>> >     While I do like the pause/resume draft, having core RTCWEB WG
>> >     documents (such as RTP Usage) depend on it seems like a bit of a
>> >     stretch. After all, the document was only adopted last week, and i=
t
>> >     is a rare IETF WG document that can go from a -00 WG draft to
>> >     publication as an RFC in under a year.
>> >
>> >
>> >     On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=E5kansson LK
>> >     <stefan.lk.hakansson@ericsson.com
>> >     <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>> >
>> >         Hi,
>> >
>> >         at the IETF last week there was consensus in the AVTEXT WG
>> >         meeting to
>> >         adopt the pause/resume draft [1] as a WG draft.
>> >
>> >         In rtcweb/webrtc we're have the situation that we're discussin=
g
>> so
>> >         called "doo-hickeys" as an API surface where the web app
>> >         (amongst other
>> >         things) can pause and resume the sending of a track. This can
>> >         be signaled with the direction attribute and a SDP O/A exchang=
e
>> >         (and the
>> >         app pausing/resuming sending of a track would presumably lead
>> to a
>> >         "negotiationneeded" event being fired).
>> >
>> >         But I think we should in addition require the browser to signa=
l
>> it
>> >         according to one of the methods in [1] (e.g. TMMBN =3D 0), and
>> also
>> >         understand that signaling (a browser receiving TMMBN =3D 0 mus=
t
>> >         know that
>> >         the other end-point will pause sending).
>> >
>> >         My argument is that we know that many dislike SDP in rtcweb,
>> and a
>> >         likely development is that it will be removed in a later
>> version. My
>> >         speculation is that signaling as outlined in [1] will then be
>> >         used for
>> >         pause/resume. If we support this from the beginning earlier
>> >         implementations could more easily interop with those later
>> versions.
>> >
>> >
>> >         Stefan
>> >
>> >         [1]
>> >
>> https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.tx=
t
>> >
>> >         _______________________________________________
>> >         rtcweb mailing list
>> >         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> >         https://www.ietf.org/mailman/listinfo/rtcweb
>> >
>> >
>> >
>> >     _______________________________________________
>> >     rtcweb mailing list
>> >     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> >     https://www.ietf.org/mailman/listinfo/rtcweb
>> >
>> >
>>
>>
>

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<div dir=3D"ltr">Justin said:=A0<div><br></div><div>&quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">Overall this feels to me like so=
mething that needs some thought to figure out all the interactions between =
the API, the signaling plane, and the media plane. As such I am inclined to=
 leave this out of 1.0.</span>&quot;</div>

<div><br></div><div>[BA] I agree.=A0</div><div><br></div><div><br></div><di=
v>&quot;<span style=3D"font-family:arial,sans-serif;font-size:13px">It&#39;=
s not clear to me what the interactions are between the signaling and media=
 plane when this occurs. That is, if I do doohickey.active =3D false, shoul=
d that cause a=3Drecvonly signaling and a TMMBN notification that the track=
 is paused? I am concerned that things can get out of sync (i.e. what happe=
ns if we get TMMBR indicating resume, but signaling is still a=3Drecvonly)?=
</span>&quot;</div>

<div><br></div><div>[BA] A few thoughts:=A0</div><div><br></div><div>a. Eve=
n though TMMBR/TMMBN is mandatory-to-implement in the RTP usage document, I=
 don&#39;t think we can assume that the pause/resume will be supported by a=
ll mixers.. It therefore probably isn&#39;t a good idea for a browser to au=
tomatically send a TMMBN 0 indication when setting rtpsender.active =3D fal=
se. =A0Since the application might or might not use SDP for signaling, this=
 is probably something that needs to be under application control. =A0BTW, =
Section 6.3, isn&#39;t clear about the response to an unsolicited TMMBN (TM=
MBR 0?).=A0</div>

<div><br></div><div>b. As described in Section 1, pause/resume is for use i=
n scenarios where signaling might not meet the performance requirements. =
=A0Due to the differing transport characteristics of RTCP and signaling ove=
r reliable transport, trying to use both pause/resume and signaling is quit=
e likely to result in them being out of sync, so my assumption was that an =
application wishing to support pause/resume would not choose to use signali=
ng for the same purpose. =A0 There is also the potential for conflict betwe=
en codec-specific messages (e.g. layer drop in H.264/SVC) and some potentia=
l uses of pause/resume (e.g. unsolicited TMMBN to indicate layer pause in m=
ulti-stream SVC). =A0Where there are conflicts, my assumption would be that=
 signaling and codec-specific messages should win.=A0</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Fri,=
 Mar 14, 2014 at 9:15 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span=
> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div class=3D"">On Fri, Mar 14, 2014=
 at 4:25 AM, Stefan H=E5kansson LK <span dir=3D"ltr">&lt;<a href=3D"mailto:=
stefan.lk.hakansson@ericsson.com" target=3D"_blank">stefan.lk.hakansson@eri=
csson.com</a>&gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">On 2014-03-12 21:56, Justin Uberti wrote:<br=
>
&gt; Agreed.<br>
<br>
I may well be that the pause/resume doc will not be ready in time, so it<br=
>
from that aspect is a bad idea. What about just stating that TMMBN/TMMBR<br=
>
0 meaning pause should be supported?<br></blockquote><div><br></div></div><=
div>It&#39;s not clear to me what the interactions are between the signalin=
g and media plane when this occurs. That is, if I do doohickey.active =3D f=
alse, should that cause a=3Drecvonly signaling and a TMMBN notification tha=
t the track is paused? I am concerned that things can get out of sync (i.e.=
 what happens if we get TMMBR indicating resume, but signaling is still a=
=3Drecvonly)?</div>

<div class=3D"">

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<div><br>
&gt; Aren&#39;t there also patent declarations against this doc from<br>
&gt; multiple holders?<br>
<br>
</div>I&#39;ve not looked into that, but I deliberately chose to propose th=
e &quot;old&quot;<br>
TMMBN 0 way to signal that is mentioned in [1] (and not any of the &quot;ne=
w&quot;<br>
signaling in [1]).<br></blockquote><div><br></div></div><div>OK.</div><div =
class=3D""><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><br>
&gt;<br>
&gt; While SDP will likely be removed from the API in the future, the<br>
&gt; replacement would be a app-specific message sent over websockets, whic=
h<br>
&gt; seems like it would work just fine.<br>
<br>
</div>s/would/could/: I don&#39;t think this has been discussed or agreed (=
and<br>
websocket is of course just an example).<br></blockquote><div><br></div></d=
iv><div>Right, the whole thing is an example, but the point is that the new=
 API directions don&#39;t really affect this particular problem.=A0</div>

<div class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">


<br>
I think it is worthwhile to push this kind of signaling into the media<br>
plane if possible. For example, this would enable the UA to save<br>
transmission/battery even if the app is badly written.<br>
<br>
As an example: if a MST sent over a PC is not used in any way at the<br>
receiving end, the receiving end UA could signal TMMBR 0 to tell the<br>
sending UA to not send. Sure, the receiving UA could also fire a<br>
&quot;negotiationneeded&quot; event, but maybe the app doesn&#39;t listen.<=
br></blockquote><div><br></div></div><div>True, if the app doesn&#39;t perf=
orm the necessary signaling, things are not going to work, but that applies=
 to a lot of operations on the PeerConnection.</div>



<div><br></div><div>Overall this feels to me like something that needs some=
 thought to figure out all the interactions between the API, the signaling =
plane, and the media plane. As such I am inclined to leave this out of 1.0.=
=A0</div>

<div><div class=3D"h5">

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div><br>
&gt;<br>
&gt;<br>
&gt; On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba &lt;<a href=3D"mailto:=
bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a><br>
</div><div>&gt; &lt;mailto:<a href=3D"mailto:bernard.aboba@gmail.com" targe=
t=3D"_blank">bernard.aboba@gmail.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =A0 =A0 While I do like the pause/resume draft, having core RTCWEB WG<=
br>
&gt; =A0 =A0 documents (such as RTP Usage) depend on it seems like a bit of=
 a<br>
&gt; =A0 =A0 stretch. After all, the document was only adopted last week, a=
nd it<br>
&gt; =A0 =A0 is a rare IETF WG document that can go from a -00 WG draft to<=
br>
&gt; =A0 =A0 publication as an RFC in under a year.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=E5kansson LK<br>
&gt; =A0 =A0 &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=
=3D"_blank">stefan.lk.hakansson@ericsson.com</a><br>
</div><div><div>&gt; =A0 =A0 &lt;mailto:<a href=3D"mailto:stefan.lk.hakanss=
on@ericsson.com" target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;=
&gt; wrote:<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 Hi,<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 at the IETF last week there was consensus in the AVTEX=
T WG<br>
&gt; =A0 =A0 =A0 =A0 meeting to<br>
&gt; =A0 =A0 =A0 =A0 adopt the pause/resume draft [1] as a WG draft.<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 In rtcweb/webrtc we&#39;re have the situation that we&=
#39;re discussing so<br>
&gt; =A0 =A0 =A0 =A0 called &quot;doo-hickeys&quot; as an API surface where=
 the web app<br>
&gt; =A0 =A0 =A0 =A0 (amongst other<br>
&gt; =A0 =A0 =A0 =A0 things) can pause and resume the sending of a track. T=
his can<br>
&gt; =A0 =A0 =A0 =A0 be signaled with the direction attribute and a SDP O/A=
 exchange<br>
&gt; =A0 =A0 =A0 =A0 (and the<br>
&gt; =A0 =A0 =A0 =A0 app pausing/resuming sending of a track would presumab=
ly lead to a<br>
&gt; =A0 =A0 =A0 =A0 &quot;negotiationneeded&quot; event being fired).<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 But I think we should in addition require the browser =
to signal it<br>
&gt; =A0 =A0 =A0 =A0 according to one of the methods in [1] (e.g. TMMBN =3D=
 0), and also<br>
&gt; =A0 =A0 =A0 =A0 understand that signaling (a browser receiving TMMBN =
=3D 0 must<br>
&gt; =A0 =A0 =A0 =A0 know that<br>
&gt; =A0 =A0 =A0 =A0 the other end-point will pause sending).<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 My argument is that we know that many dislike SDP in r=
tcweb, and a<br>
&gt; =A0 =A0 =A0 =A0 likely development is that it will be removed in a lat=
er version. My<br>
&gt; =A0 =A0 =A0 =A0 speculation is that signaling as outlined in [1] will =
then be<br>
&gt; =A0 =A0 =A0 =A0 used for<br>
&gt; =A0 =A0 =A0 =A0 pause/resume. If we support this from the beginning ea=
rlier<br>
&gt; =A0 =A0 =A0 =A0 implementations could more easily interop with those l=
ater versions.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 Stefan<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 [1]<br>
&gt; =A0 =A0 =A0 =A0 <a href=3D"https://tools.ietf.org/id/draft-westerlund-=
avtext-rtp-stream-pause-05.txt" target=3D"_blank">https://tools.ietf.org/id=
/draft-westerlund-avtext-rtp-stream-pause-05.txt</a><br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 _______________________________________________<br>
&gt; =A0 =A0 =A0 =A0 rtcweb mailing list<br>
</div></div>&gt; =A0 =A0 =A0 =A0 <a href=3D"mailto:rtcweb@ietf.org" target=
=3D"_blank">rtcweb@ietf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.or=
g" target=3D"_blank">rtcweb@ietf.org</a>&gt;<br>
<div>&gt; =A0 =A0 =A0 =A0 <a href=3D"https://www.ietf.org/mailman/listinfo/=
rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><=
br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 _______________________________________________<br>
&gt; =A0 =A0 rtcweb mailing list<br>
</div>&gt; =A0 =A0 <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtc=
web@ietf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org" target=3D"_b=
lank">rtcweb@ietf.org</a>&gt;<br>
&gt; =A0 =A0 <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" targe=
t=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt;<br>
<br>
</blockquote></div></div></div><br></div></div>
</blockquote></div><br></div>

--047d7bacb502d8f03404f494cb47--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 14 Mar 2014 11:14:23 -0700
Message-ID: <CAOJ7v-2KMa+998T=_ps+518bufDjb1367cQYWpT5JfoNsxtb1w@mail.gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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--047d7b34394043fac004f4950b46
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On Fri, Mar 14, 2014 at 10:40 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>
> Hi,
>
> > By "RTCP transport in the offer", I mean separate ICE candidates
> generated for a RTCP ICE component, the original problem that I was
> proposing to solve.
>
> Ok.
>
> But, again, we do have a decision that we are not going to specify
> procedures based on pre-knoweldge of BUNDLE support by the answerer. Or, at
> least that was the outcome in London, I guess it still has to be verified
> on the list.
>

???

I was there and I don't recall this decision; I spent half an hour
discussing this exact point in rtcweb based on a scenario you suggested
(namely, a new PC created from a fork). The discussion ended with some
interest but no consensus, which is why we are discussing it here in
email...

For sanity's sake, can you make it clear that you support this proposal?

--047d7b34394043fac004f4950b46
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Mar 14, 2014 at 10:40 AM, Christer Holmberg <span dir=3D"lt=
r">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">=
christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
Hi,<br>
<div class=3D""><br>
&gt; By &quot;RTCP transport in the offer&quot;, I mean separate ICE candid=
ates generated for a RTCP ICE component, the original problem that I was pr=
oposing to solve.<br>
<br>
</div>Ok.<br>
<br>
But, again, we do have a decision that we are not going to specify procedur=
es based on pre-knoweldge of BUNDLE support by the answerer. Or, at least t=
hat was the outcome in London, I guess it still has to be verified on the l=
ist.<br>

</blockquote><div><br></div><div>???=C2=A0</div><div><br></div><div>I was t=
here and I don&#39;t recall this decision; I spent half an hour discussing =
this exact point in rtcweb based on a scenario you suggested (namely, a new=
 PC created from a fork). The discussion ended with some interest but no co=
nsensus, which is why we are discussing it here in email...</div>

<div><br></div><div>For sanity&#39;s sake, can you make it clear that you s=
upport this proposal?</div></div></div></div>

--047d7b34394043fac004f4950b46--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 14 Mar 2014 11:28:46 -0700
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--001a1136ac66bd753a04f4953e5c
Content-Type: text/plain; charset=UTF-8

On Fri, Mar 14, 2014 at 10:32 AM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Fri, Mar 14, 2014 at 9:53 AM, Watson Ladd <watsonbladd@gmail.com>wrote:
>
>>
>> On Mar 14, 2014 9:44 AM, "Justin Uberti" <juberti@google.com> wrote:
>> >
>> >
>> >
>> >
>> > On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund <
>> magnus.westerlund@ericsson.com> wrote:
>> >>
>> >> On 2014-03-13 17:53, Martin Thomson wrote:
>> >> > On 12 March 2014 01:29, Magnus Westerlund
>> >> > <magnus.westerlund@ericsson.com> wrote:
>> >> >> I like to point out that the agreement and what is documented in
>> >> >> rtp-usage is that WebRTC endpoint will have to make forwarded
>> streams
>> >> >> appear as locally originated. However, this as currently written
>> does
>> >> >> not apply to RTP middleboxes that interconnects multiple
>> PeerConnections
>> >> >> to form a multi-party session. This is deliberate to ensure that RTP
>> >> >> topologies like RTP mixer and SFM do work on the RTP/RTCP level.
>> >> >
>> >> > I'm still not clear on whether a middlebox interoperating with a
>> >> > WebRTC endpoint is obligated to respect these rules or not.  I had
>> >> > assumed that WebRTC is such a bully that they would be forced to.
>> >>
>> >> Well, I know Colin and I did put in quite some thought to make the
>> >> WebRTC RTP usage rules such that you can use the common RTP middleboxes
>> >> as long as you have the right front-end, i.e. DTLS-SRTP and signalling
>> >> translation.
>> >>
>> >> >
>> >> >> I am especially interested to know how you will "easy to apply
>> manually"
>> >> >> the synchronization. Can you please describe that. Because, that
>> either
>> >> >> requires an API call to tell the media framework, please consider
>> the
>> >> >> following CNAMES as equivalent, or some other method of telling the
>> >> >> media framework that these different MST are actually originating
>> from a
>> >> >> common clock base.
>> >> > In WebRTC, this would be:
>> >> >
>> >> > var audio = pc1.getRemoteStreams()[0].getAudioTracks()[0];
>> >> > var video = pc2.getRemoteStreams()[0].getVideoTracks()[0];
>> >> > var synchronizedStream = new MediaStream(audio, video);
>> >> >
>> >> > We'd have to say that this implies a statement about the clock base
>> of
>> >> > the tracks being the same.  ... and that this statement could be
>> >> > wrong.
>> >>
>> >> I think this looks dangerous. If you default the assumption that
>> >> different CNAMEs of the MST you add are actually sharing clock base,
>> >> then a lot of basic programs that may group MST from different PCs into
>> >> one MS for convenience are going to cause serious issue in the media
>> >> frameworks, when they attempt to synchronize things.
>> >>
>> >> >
>> >> > As far as it goes, I'm sure that other systems could build a similar
>> >> > function, but none of those are standardized.
>> >> >
>> >> >> I protested about this, not to lower the users privacy, but over the
>> >> >> concern that this was raised without providing a case where it was
>> >> >> obvious that the user's privacy was impacted. Martin, you said that
>> you
>> >> >> would think about it, and the next statement was lets change it,
>> without
>> >> >> providing any motivation why the concern was significant.
>> >> >
>> >> > I'm sorry about that, I thought about it, but didn't want to waste
>> >> > everyone's time (mine included) with an essay on the subject.
>> >> >
>> >>
>> >> Having considered this a bit more, I do think the risk to privacy out
>> >> weighs the other concerns, as long as the API and its handling can be
>> >> sorted out, I don't think the above is sufficient to get good
>> >> functionality.
>> >>
>> >> My understanding is that the risk to privacy exist when one have one JS
>> >> application enabling communication in different contexts during the
>> same
>> >> application session. In cases where there might be participants in the
>> >>
>> >> different contexts that could link the same end-point and thus human
>> >> user to different participant IDs (that otherwise are anonymous). Thus
>> >> revealing privacy concerns. As it would take the application designer
>> to
>> >> take this risk into consideration to avoid it, I feel safer if the
>> >> application designer would need to take active measurements to perform
>> >> linkage.
>> >>
>> >>
>> >> I will wait an see if there are other inputs on this within the next
>> >> week. If nothing arises I will propose a text change to the rtp-usage
>> to
>> >> address this.
>> >>
>> >
>> > At an implementation level, one could imagine at least 3 policies for
>> generating CNAMEs:
>> > a) per-session (i.e. per-PeerConnection)
>> > b) per-page (i.e. shared between all PCs on a page)
>> > c) per-page, persistent (i.e. shared between all PCs on a page,
>> including across page loads)
>> >
>> > While we seem to agree that a) is the right solution for CNAMEs, it is
>> worth pointing out that we (Chrome) are currently doing c) for DTLS
>> certificates, to avoid performance problems with cert generation at page
>> load. Ergo, this linkability concern already exists, and I don't think it
>> is easy to solve it in the default case. There have been some proposals to
>> allow generation/storage of unique certs to prevent this linkability, but
>> this will require app input.
>> >
>>
>> One exponentiation, fixed base, per certificate. Use an addition
>> multichain. Cert generation is cheap with ECDSA.
>>
> You're going to have to explain a bit more for me to grok this.
>
> Also, define 'cheap', especially on ARM CPUs. Currently RSA generation on
> best-in-class ARM CPUs is 500 ms for 1024-bit, 2 seconds for 2048-bit. That
> is a problem.
>

As to the choice of RSA as the algorithm... we just mandated use of
TLS_DHE_RSA_WITH_AES_128_GCM_SHA256 and TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
in http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09.

--001a1136ac66bd753a04f4953e5c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Mar 14, 2014 at 10:32 AM, Justin Uberti <span dir=3D"ltr">&=
lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.c=
om</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><d=
iv class=3D"gmail_quote">

<div><div class=3D"h5">On Fri, Mar 14, 2014 at 9:53 AM, Watson Ladd <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:watsonbladd@gmail.com" target=3D"_blank">w=
atsonbladd@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div><div><p dir=3D"ltr"><br>
On Mar 14, 2014 9:44 AM, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto:ju=
berti@google.com" target=3D"_blank">juberti@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund &lt;<a href=3D"mail=
to:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerlund@eric=
sson.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On 2014-03-13 17:53, Martin Thomson wrote:<br>
&gt;&gt; &gt; On 12 March 2014 01:29, Magnus Westerlund<br>
&gt;&gt; &gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=
=3D"_blank">magnus.westerlund@ericsson.com</a>&gt; wrote:<br>
&gt;&gt; &gt;&gt; I like to point out that the agreement and what is docume=
nted in<br>
&gt;&gt; &gt;&gt; rtp-usage is that WebRTC endpoint will have to make forwa=
rded streams<br>
&gt;&gt; &gt;&gt; appear as locally originated. However, this as currently =
written does<br>
&gt;&gt; &gt;&gt; not apply to RTP middleboxes that interconnects multiple =
PeerConnections<br>
&gt;&gt; &gt;&gt; to form a multi-party session. This is deliberate to ensu=
re that RTP<br>
&gt;&gt; &gt;&gt; topologies like RTP mixer and SFM do work on the RTP/RTCP=
 level.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I&#39;m still not clear on whether a middlebox interoperating=
 with a<br>
&gt;&gt; &gt; WebRTC endpoint is obligated to respect these rules or not. =
=C2=A0I had<br>
&gt;&gt; &gt; assumed that WebRTC is such a bully that they would be forced=
 to.<br>
&gt;&gt;<br>
&gt;&gt; Well, I know Colin and I did put in quite some thought to make the=
<br>
&gt;&gt; WebRTC RTP usage rules such that you can use the common RTP middle=
boxes<br>
&gt;&gt; as long as you have the right front-end, i.e. DTLS-SRTP and signal=
ling<br>
&gt;&gt; translation.<br>
&gt;&gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&gt; I am especially interested to know how you will &quot;eas=
y to apply manually&quot;<br>
&gt;&gt; &gt;&gt; the synchronization. Can you please describe that. Becaus=
e, that either<br>
&gt;&gt; &gt;&gt; requires an API call to tell the media framework, please =
consider the<br>
&gt;&gt; &gt;&gt; following CNAMES as equivalent, or some other method of t=
elling the<br>
&gt;&gt; &gt;&gt; media framework that these different MST are actually ori=
ginating from a<br>
&gt;&gt; &gt;&gt; common clock base.<br>
&gt;&gt; &gt; In WebRTC, this would be:<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; var audio =3D pc1.getRemoteStreams()[0].getAudioTracks()[0];<=
br>
&gt;&gt; &gt; var video =3D pc2.getRemoteStreams()[0].getVideoTracks()[0];<=
br>
&gt;&gt; &gt; var synchronizedStream =3D new MediaStream(audio, video);<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; We&#39;d have to say that this implies a statement about the =
clock base of<br>
&gt;&gt; &gt; the tracks being the same. =C2=A0... and that this statement =
could be<br>
&gt;&gt; &gt; wrong.<br>
&gt;&gt;<br>
&gt;&gt; I think this looks dangerous. If you default the assumption that<b=
r>
&gt;&gt; different CNAMEs of the MST you add are actually sharing clock bas=
e,<br>
&gt;&gt; then a lot of basic programs that may group MST from different PCs=
 into<br>
&gt;&gt; one MS for convenience are going to cause serious issue in the med=
ia<br>
&gt;&gt; frameworks, when they attempt to synchronize things.<br>
&gt;&gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; As far as it goes, I&#39;m sure that other systems could buil=
d a similar<br>
&gt;&gt; &gt; function, but none of those are standardized.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&gt; I protested about this, not to lower the users privacy, b=
ut over the<br>
&gt;&gt; &gt;&gt; concern that this was raised without providing a case whe=
re it was<br>
&gt;&gt; &gt;&gt; obvious that the user&#39;s privacy was impacted. Martin,=
 you said that you<br>
&gt;&gt; &gt;&gt; would think about it, and the next statement was lets cha=
nge it, without<br>
&gt;&gt; &gt;&gt; providing any motivation why the concern was significant.=
<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I&#39;m sorry about that, I thought about it, but didn&#39;t =
want to waste<br>
&gt;&gt; &gt; everyone&#39;s time (mine included) with an essay on the subj=
ect.<br>
&gt;&gt; &gt;<br>
&gt;&gt;<br>
&gt;&gt; Having considered this a bit more, I do think the risk to privacy =
out<br>
&gt;&gt; weighs the other concerns, as long as the API and its handling can=
 be<br>
&gt;&gt; sorted out, I don&#39;t think the above is sufficient to get good<=
br>
&gt;&gt; functionality.<br>
&gt;&gt;<br>
&gt;&gt; My understanding is that the risk to privacy exist when one have o=
ne JS<br>
&gt;&gt; application enabling communication in different contexts during th=
e same<br>
&gt;&gt; application session. In cases where there might be participants in=
 the<br>
&gt;&gt;<br>
&gt;&gt; different contexts that could link the same end-point and thus hum=
an<br>
&gt;&gt; user to different participant IDs (that otherwise are anonymous). =
Thus<br>
&gt;&gt; revealing privacy concerns. As it would take the application desig=
ner to<br>
&gt;&gt; take this risk into consideration to avoid it, I feel safer if the=
<br>
&gt;&gt; application designer would need to take active measurements to per=
form<br>
&gt;&gt; linkage.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; I will wait an see if there are other inputs on this within the ne=
xt<br>
&gt;&gt; week. If nothing arises I will propose a text change to the rtp-us=
age to<br>
&gt;&gt; address this.<br>
&gt;&gt;<br>
&gt;<br>
&gt; At an implementation level, one could imagine at least 3 policies for =
generating CNAMEs:<br>
&gt; a) per-session (i.e. per-PeerConnection)<br>
&gt; b) per-page (i.e. shared between all PCs on a page)<br>
&gt; c) per-page, persistent (i.e. shared between all PCs on a page, includ=
ing across page loads)<br>
&gt;<br>
&gt; While we seem to agree that a) is the right solution for CNAMEs, it is=
 worth pointing out that we (Chrome) are currently doing c) for DTLS certif=
icates, to avoid performance problems with cert generation at page load. Er=
go, this linkability concern already exists, and I don&#39;t think it is ea=
sy to solve it in the default case. There have been some proposals to allow=
 generation/storage of unique certs to prevent this linkability, but this w=
ill require app input.<br>





&gt;</p>
</div></div><p dir=3D"ltr">One exponentiation, fixed base, per certificate.=
 Use an addition multichain. Cert generation is cheap with ECDSA.</p></bloc=
kquote></div></div><div>You&#39;re going to have to explain a bit more for =
me to grok this.</div>



<div><br></div><div>Also, define &#39;cheap&#39;, especially on ARM CPUs. C=
urrently RSA generation on best-in-class ARM CPUs is 500 ms for 1024-bit, 2=
 seconds for 2048-bit. That is a problem.<br></div></div></div></div></bloc=
kquote>

<div><br></div><div>As to the choice of RSA as the algorithm... we just man=
dated use of=C2=A0<span style=3D"color:rgb(0,0,0);font-size:1em">TLS_DHE_RS=
A_WITH_AES_128_GCM_SHA256</span>=C2=A0and=C2=A0<span style=3D"color:rgb(0,0=
,0);font-size:1em"> TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256=C2=A0</span>in=C2=
=A0<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09=
">http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09</a>.</div>

</div><br></div></div>

--001a1136ac66bd753a04f4953e5c--


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Content-Type: text/plain; charset=UTF-8

On Mar 14, 2014 11:29 AM, "Justin Uberti" <juberti@google.com> wrote:
>
>
>
>
> On Fri, Mar 14, 2014 at 10:32 AM, Justin Uberti <juberti@google.com>
wrote:
>>
>>
>>
>>
>> On Fri, Mar 14, 2014 at 9:53 AM, Watson Ladd <watsonbladd@gmail.com>
wrote:
>>>
>>>
>>> On Mar 14, 2014 9:44 AM, "Justin Uberti" <juberti@google.com> wrote:
>>> >
>>> >
>>> >
>>> >
>>> > On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:
>>> >>
>>> >> On 2014-03-13 17:53, Martin Thomson wrote:
>>> >> > On 12 March 2014 01:29, Magnus Westerlund
>>> >> > <magnus.westerlund@ericsson.com> wrote:
>>> >> >> I like to point out that the agreement and what is documented in
>>> >> >> rtp-usage is that WebRTC endpoint will have to make forwarded
streams
>>> >> >> appear as locally originated. However, this as currently written
does
>>> >> >> not apply to RTP middleboxes that interconnects multiple
PeerConnections
>>> >> >> to form a multi-party session. This is deliberate to ensure that
RTP
>>> >> >> topologies like RTP mixer and SFM do work on the RTP/RTCP level.
>>> >> >
>>> >> > I'm still not clear on whether a middlebox interoperating with a
>>> >> > WebRTC endpoint is obligated to respect these rules or not.  I had
>>> >> > assumed that WebRTC is such a bully that they would be forced to.
>>> >>
>>> >> Well, I know Colin and I did put in quite some thought to make the
>>> >> WebRTC RTP usage rules such that you can use the common RTP
middleboxes
>>> >> as long as you have the right front-end, i.e. DTLS-SRTP and
signalling
>>> >> translation.
>>> >>
>>> >> >
>>> >> >> I am especially interested to know how you will "easy to apply
manually"
>>> >> >> the synchronization. Can you please describe that. Because, that
either
>>> >> >> requires an API call to tell the media framework, please consider
the
>>> >> >> following CNAMES as equivalent, or some other method of telling
the
>>> >> >> media framework that these different MST are actually originating
from a
>>> >> >> common clock base.
>>> >> > In WebRTC, this would be:
>>> >> >
>>> >> > var audio = pc1.getRemoteStreams()[0].getAudioTracks()[0];
>>> >> > var video = pc2.getRemoteStreams()[0].getVideoTracks()[0];
>>> >> > var synchronizedStream = new MediaStream(audio, video);
>>> >> >
>>> >> > We'd have to say that this implies a statement about the clock
base of
>>> >> > the tracks being the same.  ... and that this statement could be
>>> >> > wrong.
>>> >>
>>> >> I think this looks dangerous. If you default the assumption that
>>> >> different CNAMEs of the MST you add are actually sharing clock base,
>>> >> then a lot of basic programs that may group MST from different PCs
into
>>> >> one MS for convenience are going to cause serious issue in the media
>>> >> frameworks, when they attempt to synchronize things.
>>> >>
>>> >> >
>>> >> > As far as it goes, I'm sure that other systems could build a
similar
>>> >> > function, but none of those are standardized.
>>> >> >
>>> >> >> I protested about this, not to lower the users privacy, but over
the
>>> >> >> concern that this was raised without providing a case where it was
>>> >> >> obvious that the user's privacy was impacted. Martin, you said
that you
>>> >> >> would think about it, and the next statement was lets change it,
without
>>> >> >> providing any motivation why the concern was significant.
>>> >> >
>>> >> > I'm sorry about that, I thought about it, but didn't want to waste
>>> >> > everyone's time (mine included) with an essay on the subject.
>>> >> >
>>> >>
>>> >> Having considered this a bit more, I do think the risk to privacy out
>>> >> weighs the other concerns, as long as the API and its handling can be
>>> >> sorted out, I don't think the above is sufficient to get good
>>> >> functionality.
>>> >>
>>> >> My understanding is that the risk to privacy exist when one have one
JS
>>> >> application enabling communication in different contexts during the
same
>>> >> application session. In cases where there might be participants in
the
>>> >>
>>> >> different contexts that could link the same end-point and thus human
>>> >> user to different participant IDs (that otherwise are anonymous).
Thus
>>> >> revealing privacy concerns. As it would take the application
designer to
>>> >> take this risk into consideration to avoid it, I feel safer if the
>>> >> application designer would need to take active measurements to
perform
>>> >> linkage.
>>> >>
>>> >>
>>> >> I will wait an see if there are other inputs on this within the next
>>> >> week. If nothing arises I will propose a text change to the
rtp-usage to
>>> >> address this.
>>> >>
>>> >
>>> > At an implementation level, one could imagine at least 3 policies for
generating CNAMEs:
>>> > a) per-session (i.e. per-PeerConnection)
>>> > b) per-page (i.e. shared between all PCs on a page)
>>> > c) per-page, persistent (i.e. shared between all PCs on a page,
including across page loads)
>>> >
>>> > While we seem to agree that a) is the right solution for CNAMEs, it
is worth pointing out that we (Chrome) are currently doing c) for DTLS
certificates, to avoid performance problems with cert generation at page
load. Ergo, this linkability concern already exists, and I don't think it
is easy to solve it in the default case. There have been some proposals to
allow generation/storage of unique certs to prevent this linkability, but
this will require app input.
>>> >
>>>
>>> One exponentiation, fixed base, per certificate. Use an addition
multichain. Cert generation is cheap with ECDSA.
>>
>> You're going to have to explain a bit more for me to grok this.
>>
>> Also, define 'cheap', especially on ARM CPUs. Currently RSA generation
on best-in-class ARM CPUs is 500 ms for 1024-bit, 2 seconds for 2048-bit.
That is a problem.
>
>
> As to the choice of RSA as the algorithm... we just mandated use
of TLS_DHE_RSA_WITH_AES_128_GCM_SHA256 and
TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 in
http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09.
>

That's ridiculous. It should be changed.  I'll email the list to explain
why.

--20cf3040e42efd8ecd04f4954e09
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr"><br>
On Mar 14, 2014 11:29 AM, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto:j=
uberti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Mar 14, 2014 at 10:32 AM, Justin Uberti &lt;<a href=3D"mailto:=
juberti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Fri, Mar 14, 2014 at 9:53 AM, Watson Ladd &lt;<a href=3D"mailto=
:watsonbladd@gmail.com">watsonbladd@gmail.com</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On Mar 14, 2014 9:44 AM, &quot;Justin Uberti&quot; &lt;<a href=
=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt; On Fri, Mar 14, 2014 at 1:34 AM, Magnus Westerlund &lt;<a=
 href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.=
com</a>&gt; wrote:<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; On 2014-03-13 17:53, Martin Thomson wrote:<br>
&gt;&gt;&gt; &gt;&gt; &gt; On 12 March 2014 01:29, Magnus Westerlund<br>
&gt;&gt;&gt; &gt;&gt; &gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson=
.com">magnus.westerlund@ericsson.com</a>&gt; wrote:<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; I like to point out that the agreement and w=
hat is documented in<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; rtp-usage is that WebRTC endpoint will have =
to make forwarded streams<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; appear as locally originated. However, this =
as currently written does<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; not apply to RTP middleboxes that interconne=
cts multiple PeerConnections<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; to form a multi-party session. This is delib=
erate to ensure that RTP<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; topologies like RTP mixer and SFM do work on=
 the RTP/RTCP level.<br>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt; I&#39;m still not clear on whether a middlebox i=
nteroperating with a<br>
&gt;&gt;&gt; &gt;&gt; &gt; WebRTC endpoint is obligated to respect these ru=
les or not. =C2=A0I had<br>
&gt;&gt;&gt; &gt;&gt; &gt; assumed that WebRTC is such a bully that they wo=
uld be forced to.<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; Well, I know Colin and I did put in quite some though=
t to make the<br>
&gt;&gt;&gt; &gt;&gt; WebRTC RTP usage rules such that you can use the comm=
on RTP middleboxes<br>
&gt;&gt;&gt; &gt;&gt; as long as you have the right front-end, i.e. DTLS-SR=
TP and signalling<br>
&gt;&gt;&gt; &gt;&gt; translation.<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; I am especially interested to know how you w=
ill &quot;easy to apply manually&quot;<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; the synchronization. Can you please describe=
 that. Because, that either<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; requires an API call to tell the media frame=
work, please consider the<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; following CNAMES as equivalent, or some othe=
r method of telling the<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; media framework that these different MST are=
 actually originating from a<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; common clock base.<br>
&gt;&gt;&gt; &gt;&gt; &gt; In WebRTC, this would be:<br>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt; var audio =3D pc1.getRemoteStreams()[0].getAudio=
Tracks()[0];<br>
&gt;&gt;&gt; &gt;&gt; &gt; var video =3D pc2.getRemoteStreams()[0].getVideo=
Tracks()[0];<br>
&gt;&gt;&gt; &gt;&gt; &gt; var synchronizedStream =3D new MediaStream(audio=
, video);<br>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt; We&#39;d have to say that this implies a stateme=
nt about the clock base of<br>
&gt;&gt;&gt; &gt;&gt; &gt; the tracks being the same. =C2=A0... and that th=
is statement could be<br>
&gt;&gt;&gt; &gt;&gt; &gt; wrong.<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; I think this looks dangerous. If you default the assu=
mption that<br>
&gt;&gt;&gt; &gt;&gt; different CNAMEs of the MST you add are actually shar=
ing clock base,<br>
&gt;&gt;&gt; &gt;&gt; then a lot of basic programs that may group MST from =
different PCs into<br>
&gt;&gt;&gt; &gt;&gt; one MS for convenience are going to cause serious iss=
ue in the media<br>
&gt;&gt;&gt; &gt;&gt; frameworks, when they attempt to synchronize things.<=
br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt; As far as it goes, I&#39;m sure that other syste=
ms could build a similar<br>
&gt;&gt;&gt; &gt;&gt; &gt; function, but none of those are standardized.<br=
>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; I protested about this, not to lower the use=
rs privacy, but over the<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; concern that this was raised without providi=
ng a case where it was<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; obvious that the user&#39;s privacy was impa=
cted. Martin, you said that you<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; would think about it, and the next statement=
 was lets change it, without<br>
&gt;&gt;&gt; &gt;&gt; &gt;&gt; providing any motivation why the concern was=
 significant.<br>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt; &gt; I&#39;m sorry about that, I thought about it, bu=
t didn&#39;t want to waste<br>
&gt;&gt;&gt; &gt;&gt; &gt; everyone&#39;s time (mine included) with an essa=
y on the subject.<br>
&gt;&gt;&gt; &gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; Having considered this a bit more, I do think the ris=
k to privacy out<br>
&gt;&gt;&gt; &gt;&gt; weighs the other concerns, as long as the API and its=
 handling can be<br>
&gt;&gt;&gt; &gt;&gt; sorted out, I don&#39;t think the above is sufficient=
 to get good<br>
&gt;&gt;&gt; &gt;&gt; functionality.<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; My understanding is that the risk to privacy exist wh=
en one have one JS<br>
&gt;&gt;&gt; &gt;&gt; application enabling communication in different conte=
xts during the same<br>
&gt;&gt;&gt; &gt;&gt; application session. In cases where there might be pa=
rticipants in the<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; different contexts that could link the same end-point=
 and thus human<br>
&gt;&gt;&gt; &gt;&gt; user to different participant IDs (that otherwise are=
 anonymous). Thus<br>
&gt;&gt;&gt; &gt;&gt; revealing privacy concerns. As it would take the appl=
ication designer to<br>
&gt;&gt;&gt; &gt;&gt; take this risk into consideration to avoid it, I feel=
 safer if the<br>
&gt;&gt;&gt; &gt;&gt; application designer would need to take active measur=
ements to perform<br>
&gt;&gt;&gt; &gt;&gt; linkage.<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;&gt; I will wait an see if there are other inputs on this =
within the next<br>
&gt;&gt;&gt; &gt;&gt; week. If nothing arises I will propose a text change =
to the rtp-usage to<br>
&gt;&gt;&gt; &gt;&gt; address this.<br>
&gt;&gt;&gt; &gt;&gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt; At an implementation level, one could imagine at least 3 =
policies for generating CNAMEs:<br>
&gt;&gt;&gt; &gt; a) per-session (i.e. per-PeerConnection)<br>
&gt;&gt;&gt; &gt; b) per-page (i.e. shared between all PCs on a page)<br>
&gt;&gt;&gt; &gt; c) per-page, persistent (i.e. shared between all PCs on a=
 page, including across page loads)<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt; While we seem to agree that a) is the right solution for =
CNAMEs, it is worth pointing out that we (Chrome) are currently doing c) fo=
r DTLS certificates, to avoid performance problems with cert generation at =
page load. Ergo, this linkability concern already exists, and I don&#39;t t=
hink it is easy to solve it in the default case. There have been some propo=
sals to allow generation/storage of unique certs to prevent this linkabilit=
y, but this will require app input.<br>

&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; One exponentiation, fixed base, per certificate. Use an additi=
on multichain. Cert generation is cheap with ECDSA.<br>
&gt;&gt;<br>
&gt;&gt; You&#39;re going to have to explain a bit more for me to grok this=
.<br>
&gt;&gt;<br>
&gt;&gt; Also, define &#39;cheap&#39;, especially on ARM CPUs. Currently RS=
A generation on best-in-class ARM CPUs is 500 ms for 1024-bit, 2 seconds fo=
r 2048-bit. That is a problem.<br>
&gt;<br>
&gt;<br>
&gt; As to the choice of RSA as the algorithm... we just mandated use of=C2=
=A0TLS_DHE_RSA_WITH_AES_128_GCM_SHA256=C2=A0and=C2=A0 TLS_ECDHE_RSA_WITH_AE=
S_128_GCM_SHA256=C2=A0in=C2=A0<a href=3D"http://tools.ietf.org/html/draft-i=
etf-rtcweb-security-arch-09">http://tools.ietf.org/html/draft-ietf-rtcweb-s=
ecurity-arch-09</a>.<br>

&gt;</p>
<p dir=3D"ltr">That&#39;s ridiculous. It should be changed.=C2=A0 I&#39;ll =
email the list to explain why.</p>

--20cf3040e42efd8ecd04f4954e09--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] BUNDLE with implicit rtcp-mux
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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Hi,

>>> By "RTCP transport in the offer", I mean separate ICE candidates genera=
ted for a RTCP ICE component, the original problem that I was proposing to =
solve.
>>
>> Ok.
>>
>>But, again, we do have a decision that we are not going to specify proced=
ures based on pre-knoweldge of BUNDLE support by the answerer. Or, at least=
 that was the outcome in London, I guess it still has to be verified on the=
 list.
>
> ???
>
> I was there and I don't recall this decision; I spent half an hour discus=
sing this exact point in rtcweb based on a scenario you suggested (namely, =
a new PC created from a fork). The discussion ended with some interest but =
no consensus, which is why we are discussing it here in email...

Ok. My recall was that there was no support to do the "optimization" forkin=
g, i.e. allowing identical port values (and, in this case, also rtcp-mux) a=
lready in the initial offer for a PC, and therefor we were not going to do =
it.=20

I'm glad if I'm wrong :)

> For sanity's sake, can you make it clear that you support this proposal?

Yes, I do.

Regards,

Christer


From nobody Fri Mar 14 17:07:16 2014
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Subject: [rtcweb] Unlinkability and RSA
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Dear all,

draft-ietf-rtcweb-security-arch-09 asks "[[OPEN ISSUE: Are these the
right cipher suites?]]" and then describes two RSA suites. They are
secure, but slow, and certificates need to be regenerated frequently
to preserve unlinkability. ECDSA is much more performant, and this
would ameliorate issues relating to certificate regeneration
performance on low-power devices, as well as enable lower-power CPUs
on video devices that intend to interoperate, via custom signalling,
with webRTC.

Secondly, ditch DHE. If you have ECDSA, you have everything for ECDHE,
and it avoids slow, frequently skipped, and necessary checks in the
protocol. I can think of no reason to use it.

Thanks to Justin Uberti for pointing out this issue and the
unlinkability consequences on constrained devices.
Sincerely,
Watson Ladd


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On 2014-03-14 17:44, Justin Uberti wrote:

> 
> At an implementation level, one could imagine at least 3 policies for
> generating CNAMEs:
> a) per-session (i.e. per-PeerConnection)
> b) per-page (i.e. shared between all PCs on a page)
> c) per-page, persistent (i.e. shared between all PCs on a page,
> including across page loads)
> 
> While we seem to agree that a) is the right solution for CNAMEs, it is
> worth pointing out that we (Chrome) are currently doing c) for DTLS
> certificates, to avoid performance problems with cert generation at page
> load. Ergo, this linkability concern already exists, and I don't think
> it is easy to solve it in the default case. There have been some
> proposals to allow generation/storage of unique certs to prevent this
> linkability, but this will require app input.
> 
> Ergo, we might want to match the DTLS behavior (i.e. generate unique
> CNAMEs only when the certs are unique), to ensure we treat linkability
> consistently.

Actually, the DTLS cert and the CNAME is actually not equivalent when it
comes to visibility scope. The DTLS is show only to the DTLS peer, i.e.
the address at the other end of a peer connection. The CNAME in cases of
SFM or RTP mixer based using CSRC lists type of RTP middleboxes, can
result in the CNAME being forward to all participants in the same
multi-party conference.

Cheers

Magnus Westerlund

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On 2014-03-14 17:26, Justin Uberti wrote:
> 
> 
> 
> On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     Hi,
>     (as individual)
> 
>     Although this discussion really should be on MMUSIC, I am still
>     following up here. Anything we come to conclude here really need to go
>     to MMUSIC for confirmation.
> 
>     Justin, I do agree that it appear unnecessarily of having to include an
>     RTCP transport in an OFFER when the offerer know the answerer is BUNDLE
>     capable. I am fine with that and think the rule needed here is something
>     along the line of the below:
> 
>     A BUNDLE supporting endpoint MUST implement a=rtcp-mux (that is already
>     required). In any answer by a BUNDLE capable endpoint, it MUST NOT
>     change the usage of a=rtcp-mux. If it is present in the offer, it must
>     be confirmed to be used in the answer, and if not present in the offer,
>     it MUST NOT be added in the answer. An BUNDLE capable endpoint is
>     RECOMMENDED to use a=rtcp-mux whenever suitable. Cases when it might not
>     be suitable include, lack RTP payload types, or cases when RTCP traffic
>     is needed to be on its own transport flow (5-tuple) to enable QoS or
>     other traffic handling functionalities.
> 
>     The reason for writing the rule like the above, is that then an endpoint
>     can choose when creating an offer to renegotiate the use of a=rtcp-mux.
>     For example due to it running out of available PTs.
> 
>     Feedback on this proposal?
> 
> 
> I am not OK with this. It just gets too complicated to deal with all the
> cases, especially if with the provision that rtcp-mux can change during
> the session.

(still as individual)

Lets be clear what we are discussing here. I don't think this is lot of
cases. And if you don't want to implement this in your code generating
offer, that is fine. What this describes is that your code handling
reception of offers and generating answers must be able to handle the
lack of a=rtcp-mux in any offer, rather than just the initial one.

I would question if even that assumption holds considering what happens
if people start implementing session resumption or session mobility
between devices using rehydration style establishment procedures. What
one sides consider an initial offer might not be the state the answering
party is in. Thus, I am worried about cases that are first offer/answer
exchange related.

To my understanding you will have code to handle the following cases
when offers contains:

1) No bundle, no a=rtcp-mux
2) No bundle, with a=rtcp-mux
3) Bundle, with a=rtcp-mux

Considering that you support bundle, and you support usage or not of
a=rtcp-mux. Then I don't see how the code handling the SDP logic for
Bundle, without a=rtcp-mux is that significant.

First of all, to be robust, you have to be capable of handling
significant reconfigurations of the Peer Connection session in an Offer.
If someone moves the peer of one peer connection to another device
anything can happen, and everything can change.

Secondly, to be able to turn off a=rtcp-mux is one method which can free
up PTs if one run out. The other would be first unbundle the media
types, then to completely unbundle the media sources. The later would be
far worse when it comes to requiring providing ICE candidates and
dealing with new transports. I however think unbundling the media types
is likely to resolve many of the PT shortage situations.

Just to confirm, you are also against allowing any initial offer request
with bundle but without a=rtcp-mux within that bundle-group?

This, I would question, as it prevents anyone from treating the RTCP
packets differently from the RTP packets within one RTP session. I know
that this has been done in some cases, including IMS.

>From a principal point of view I think this should be either be use of
BUNDLE groups results in MUST use a=rtcp-mux with RTP or that any offer
may change the usage of a=rtcp-mux. Anything else depending on initial
offer appears very messy and prone to errors.

I would appreciate if someone else would provide their input and views
here!

First on the question if they need to separate the RTP and RTCP at all
within a BUNDLE group?

Secondly if they see that splitting a bundle group into different groups
if one runs into issue is a sufficient and more preferred method for
dealing with any PT exhaustion issue?


Cheers

Magnus Westerlund

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Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/kDP84R1VCPto80woR5memXBCTU8
Subject: Re: [rtcweb] [tram] Payload Types assignments
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On 03/13/2014 10:02 AM, Karl Stahl wrote:
>
> For the"mission to bring quality to real-time traffic over our best 
> effort Internet" I have started 
> http://www.ietf.org/mail-archive/web/tram/current/msg00331.html
>
> [tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5 
> IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff [1] [2]
>
> Hi Magnus,
>
> With the response just sent to Pål 
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html, 
> skipping all the QoS methods and functions that really does not belong 
> here when creating an "Interface to QoS", I only see the suggestion of 
> "data channel information", i.e. also marking each data channel packet 
> with traffic bandwidth and type information  to be relevant and 
> interesting. I checked with one of our developers and I think the same 
> traffic information as for RTP can be transferred in data channel 
> packets as follows:
>
> The RTP header has the following format (RFC 3550):
>
> 0                   1                   2 3
>
>     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>
> +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
> |V=2|P|X|  CC   |M|     PT      |       sequence number         |
>
> +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
> | timestamp                           |
>
> +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
> |           synchronization source (SSRC) identifier            |
>
> +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>
> |            contributing source (CSRC) identifiers             |
>
> | ....                              |
>
> +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>
> |            RFC5285 header extension ...                      |
>
> |     2 bytes Max Bandwidth     |   2 bytes Traffic Type        |
>
> | ....                              |
>
> RTP Payload (often encrypted)
>
> A Data Channel header could have the following format:
>
> 0                   1                   2 3
>
>     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>
> +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
>    | Data Channel Magic Cookie   |      (sequence number) |
>
> +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
> | timestamp                           |
>
> +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
> |           synchronization source (SSRC) identifier            |
>
> +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>
> |            contributing source (CSRC) identifiers             |
>
> | ....                              |
>
> +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>
> |            RFC5285 header extension ...                      |
>
> |     2 bytes Max Bandwidth     |   2 bytes Traffic Type        |
>
> | ....                              |
>
>   DTLS Header 13 bytes
>
>   SCTP
>
> The RFC5285 header extension is not in a fixed position in any of 
> these cases, but still trivial to find (especially when in  a TURN 
> flow). This could be introduced in 
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-5
>
> by making room for these bytes after the UDP header, but before the 
> DTLS header.
>
> What do you think?
>


No.

We've chosen to make Data Channel a protocol within DTLS, and not use 
any unencrypted headers for it. Neither does the data channel have any 
concept resembling SSRC, CSRC or timestamp. Changing to an RTP-based 
format such as the one you propose would be a drastic revision of the spec.

There's no reason to believe such a revision at this time has any 
traction in the WG.


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    <div class="moz-cite-prefix">On 03/13/2014 10:02 AM, Karl Stahl
      wrote:<br>
    </div>
    <blockquote
      cite="mid:025d01cf3e9a$f4267340$dc7359c0$@stahl@intertex.se"
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        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">For the</span><span
            style="font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"> </span><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">&#8220;</span><span lang="EN-US">mission to bring
            quality to real-time traffic over our best effort Internet&#8221;</span><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">&nbsp;I have started </span><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:black"><a
              moz-do-not-send="true"
              href="http://www.ietf.org/mail-archive/web/tram/current/msg00331.html"><span
                style="color:blue" lang="EN-US">http://www.ietf.org/mail-archive/web/tram/current/msg00331.html</span></a></span><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:black"
            lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:black"
            lang="EN-US">[tram] [rtcweb] The way to "Interfacing to
            QoS", A level 3-5 IP/IETF/WebRTC-thing how to interface to
            lower level's QoS-stuff </span><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">[1] [2]</span><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:black"
            lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">Hi Magnus,<o:p></o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">With the response just sent to P&aring;l <a
              moz-do-not-send="true"
              href="http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html</a>,
            skipping all the QoS methods and functions that really does
            not belong here when creating an &#8220;Interface to QoS&#8221;, I only
            see the suggestion of &#8220;</span><span lang="EN-US">data
            channel information</span><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">&#8221;, i.e. also marking each data channel packet
            with traffic bandwidth and type information &nbsp;to be relevant
            and interesting. I checked with one of our developers and I
            think the same traffic information as for RTP can be
            transferred in data channel packets as follows:<o:p></o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">The
            RTP header has the following format (RFC 3550):<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;&nbsp;
            0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            3<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;&nbsp; 0
            1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
            1<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |V=2|P|X|&nbsp; CC&nbsp;&nbsp; |M|&nbsp;&nbsp;&nbsp;&nbsp; PT&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; sequence
            number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            timestamp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synchronization source (SSRC)
            identifier&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; contributing source (CSRC)
            identifiers&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            ....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC5285 header extension
            ...&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp; 2 bytes Max Bandwidth&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; 2 bytes Traffic
            Type&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            ....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">RTP
            Payload (often encrypted)<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:black" lang="EN-US">&nbsp;</span><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">A Data
            Channel header could have the following format:<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;&nbsp;
            0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            3<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;&nbsp; 0
            1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
            1<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp; |&nbsp;&nbsp;
            Data Channel Magic Cookie&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(sequence number)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            timestamp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synchronization source (SSRC)
            identifier&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; contributing source (CSRC)
            identifiers&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            ....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC5285 header extension
            ...&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp; 2 bytes Max Bandwidth&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; 2 bytes Traffic
            Type&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp;&nbsp;
            |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            ....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp; DTLS
            Header 13 bytes<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;,&quot;serif&quot;;color:black" lang="EN-US">&nbsp; SCTP<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:black" lang="EN-US">&nbsp;<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">The RFC5285 header extension is not in a fixed
            position in any of these cases, but still trivial to find
            (especially when in&nbsp; a TURN flow). This could be introduced
            in <a moz-do-not-send="true"
href="http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-5">http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-5</a>
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">by making room for these bytes after the UDP
            header, but before the DTLS header.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">What do you think?</span><br>
        </p>
      </div>
    </blockquote>
    <br>
    <br>
    No.<br>
    <br>
    We've chosen to make Data Channel a protocol within DTLS, and not
    use any unencrypted headers for it. Neither does the data channel
    have any concept resembling SSRC, CSRC or timestamp. Changing to an
    RTP-based format such as the one you propose would be a drastic
    revision of the spec.<br>
    <br>
    There's no reason to believe such a revision at this time has any
    traction in the WG.<br>
    <br>
  </body>
</html>

--------------000209070002030103090505--


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On 3/12/2014 4:56 PM, Justin Uberti wrote:
> Agreed. Aren't there also patent declarations against this doc from 
> multiple holders?
>
> While SDP will likely be removed from the API in the future, the 
> replacement would be a app-specific message sent over websockets, 
> which seems like it would work just fine.

Except that WebSockets doesn't work peer2peer, which means both delay 
and issues with signaling configurations without a server sitting 
between the users.

However, an app-specific message sent over a DataChannel should work 
just fine.

The downside would be that everything becomes app-specific, which would 
be a problem for any sort of federation or peering setup, and likely 
gatewaying to legacy.


-- 
Randell Jesup -- rjesup a t mozilla d o t com


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Subject: [rtcweb] Transports: RFC 4941 support?
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--089e013a12f060ceae04f4eb2d43
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https://tools.ietf.org/html/rfc4941 defines the concept of temporary IPv6
addresses. For, example, as enumerated on my local system:

inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf
inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf *temporary *

As indicated in the RFC, the temporary addresses expire after hours or
days, and therefore could be used to prevent long-term linkability of
sessions. Expiration shouldn't be an issue for WebRTC, since we can simply
do ICE restart if this occurs during a session.

Is this something we want to recommend in the transports doc?

--089e013a12f060ceae04f4eb2d43
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<div dir=3D"ltr"><a href=3D"https://tools.ietf.org/html/rfc4941">https://to=
ols.ietf.org/html/rfc4941</a> defines the concept of temporary IPv6 address=
es. For, example, as enumerated on my local system:<br><div><br></div><div>=
<span style=3D"font-family:&#39;courier new&#39;,monospace;font-size:13px">=
inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255</span><br style=
=3D"font-family:&#39;courier new&#39;,monospace;font-size:13px">

<span style=3D"font-family:&#39;courier new&#39;,monospace;font-size:13px">=
inet6 2620::1008:100b:e2f8:47ff:wwww</span><span style=3D"font-family:&#39;=
courier new&#39;,monospace;font-size:13px">:xxxx prefixlen 64 autoconf=C2=
=A0</span><br style=3D"font-family:&#39;courier new&#39;,monospace;font-siz=
e:13px">

<span style=3D"font-family:&#39;courier new&#39;,monospace;font-size:13px">=
inet6 2620::1008:100b:819e:1d3f:yyyy</span><span style=3D"font-family:&#39;=
courier new&#39;,monospace;font-size:13px">:zzzz prefixlen 64 autoconf <b>t=
emporary=C2=A0</b></span><br>

</div><div><span style=3D"font-family:&#39;courier new&#39;,monospace;font-=
size:13px"><br></span></div><div>As indicated in the RFC, the temporary add=
resses expire after hours or days, and therefore could be used to prevent l=
ong-term linkability of sessions. Expiration shouldn&#39;t be an issue for =
WebRTC, since we can simply do ICE restart if this occurs during a session.=
</div>

<div><br></div><div>Is this something we want to recommend in the transport=
s doc?<br></div></div>

--089e013a12f060ceae04f4eb2d43--


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On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com> wrote:

> https://tools.ietf.org/html/rfc4941 defines the concept of temporary =
IPv6 addresses. For, example, as enumerated on my local system:
>=20
> inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
> inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf=20
> inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf =
temporary=20
>=20
> As indicated in the RFC, the temporary addresses expire after hours or =
days, and therefore could be used to prevent long-term linkability of =
sessions. Expiration shouldn't be an issue for WebRTC, since we can =
simply do ICE restart if this occurs during a session.
>=20
> Is this something we want to recommend in the transports doc?

Yes.

And it should be as frequent as every new 'call', if IPv6 privacy =
addresses are to provide the same privacy as a NAPT44, as a NAPT44 =
should be assigning random ports per reasons described in RFC6056.

The transport document should also recommend doing port randomization =
(RFC6056), as that was found pretty useful for DNS, but randomizing =
ports is still not popular with many OS's TCP stacks for whatever =
reason.

-d



--Apple-Mail=_B31ED087-23B5-4C3D-BF23-16861A720391
Content-Transfer-Encoding: 7bit
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	charset=us-ascii

<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><br><div><div>On Mar 18, 2014, at 6:00 PM, Justin Uberti &lt;<a href="mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><meta http-equiv="Content-Type" content="text/html; charset=utf-8"><div dir="ltr"><a href="https://tools.ietf.org/html/rfc4941">https://tools.ietf.org/html/rfc4941</a> defines the concept of temporary IPv6 addresses. For, example, as enumerated on my local system:<br><div><br></div><div><span style="font-family:'courier new',monospace;font-size:13px">inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255</span><br style="font-family:'courier new',monospace;font-size:13px">

<span style="font-family:'courier new',monospace;font-size:13px">inet6 2620::1008:100b:e2f8:47ff:wwww</span><span style="font-family:'courier new',monospace;font-size:13px">:xxxx prefixlen 64 autoconf&nbsp;</span><br style="font-family:'courier new',monospace;font-size:13px">

<span style="font-family:'courier new',monospace;font-size:13px">inet6 2620::1008:100b:819e:1d3f:yyyy</span><span style="font-family:'courier new',monospace;font-size:13px">:zzzz prefixlen 64 autoconf <b>temporary&nbsp;</b></span><br>

</div><div><span style="font-family:'courier new',monospace;font-size:13px"><br></span></div><div>As indicated in the RFC, the temporary addresses expire after hours or days, and therefore could be used to prevent long-term linkability of sessions. Expiration shouldn't be an issue for WebRTC, since we can simply do ICE restart if this occurs during a session.</div>

<div><br></div><div>Is this something we want to recommend in the transports doc?<br></div></div></blockquote><br></div><div>Yes.</div><div><br></div><div>And it should be as frequent as every new 'call', if IPv6 privacy addresses are to provide the same privacy as a NAPT44, as a NAPT44 should be assigning random ports per reasons described in RFC6056.</div><div><br></div><div>The transport document should also recommend doing port randomization (RFC6056), as that was found pretty useful for DNS, but randomizing ports is still not popular with many OS's TCP stacks for whatever reason.</div><div><br></div><div>-d</div><div><br></div><br></body></html>
--Apple-Mail=_B31ED087-23B5-4C3D-BF23-16861A720391--


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I'd like to be silent on the issue, since which IPv6 addresses to prefer is
likely to be a matter of system policy. Trying to override system policy in
an application specific profile usually leads to sadness.



On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com> wrote:

>
> On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com> wrote:
>
> https://tools.ietf.org/html/rfc4941 defines the concept of temporary IPv6
> addresses. For, example, as enumerated on my local system:
>
> inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
> inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf
> inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf
> *temporary *
>
> As indicated in the RFC, the temporary addresses expire after hours or
> days, and therefore could be used to prevent long-term linkability of
> sessions. Expiration shouldn't be an issue for WebRTC, since we can simply
> do ICE restart if this occurs during a session.
>
> Is this something we want to recommend in the transports doc?
>
>
> Yes.
>
> And it should be as frequent as every new 'call', if IPv6 privacy
> addresses are to provide the same privacy as a NAPT44, as a NAPT44 should
> be assigning random ports per reasons described in RFC6056.
>
> The transport document should also recommend doing port randomization
> (RFC6056), as that was found pretty useful for DNS, but randomizing ports
> is still not popular with many OS's TCP stacks for whatever reason.
>
> -d
>
>
>

--001a11c210ccb43d2004f4f03de4
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<div dir=3D"ltr">I&#39;d like to be silent on the issue, since which IPv6 a=
ddresses to prefer is likely to be a matter of system policy. Trying to ove=
rride system policy in an application specific profile usually leads to sad=
ness.<div>

<br></div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quot=
e">On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <span dir=3D"ltr">&lt;<a href=
=3D"mailto:dwing@cisco.com" target=3D"_blank">dwing@cisco.com</a>&gt;</span=
> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word"><div><di=
v class=3D"h5"><br><div><div>On Mar 18, 2014, at 6:00 PM, Justin Uberti &lt=
;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com=
</a>&gt; wrote:</div>

<br><blockquote type=3D"cite"><div dir=3D"ltr"><a href=3D"https://tools.iet=
f.org/html/rfc4941" target=3D"_blank">https://tools.ietf.org/html/rfc4941</=
a> defines the concept of temporary IPv6 addresses. For, example, as enumer=
ated on my local system:<br>

<div><br></div><div><span style=3D"font-family:&#39;courier new&#39;,monosp=
ace;font-size:13px">inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.2=
55</span><br style=3D"font-family:&#39;courier new&#39;,monospace;font-size=
:13px">



<span style=3D"font-family:&#39;courier new&#39;,monospace;font-size:13px">=
inet6 2620::1008:100b:e2f8:47ff:wwww</span><span style=3D"font-family:&#39;=
courier new&#39;,monospace;font-size:13px">:xxxx prefixlen 64 autoconf=C2=
=A0</span><br style=3D"font-family:&#39;courier new&#39;,monospace;font-siz=
e:13px">



<span style=3D"font-family:&#39;courier new&#39;,monospace;font-size:13px">=
inet6 2620::1008:100b:819e:1d3f:yyyy</span><span style=3D"font-family:&#39;=
courier new&#39;,monospace;font-size:13px">:zzzz prefixlen 64 autoconf <b>t=
emporary=C2=A0</b></span><br>



</div><div><span style=3D"font-family:&#39;courier new&#39;,monospace;font-=
size:13px"><br></span></div><div>As indicated in the RFC, the temporary add=
resses expire after hours or days, and therefore could be used to prevent l=
ong-term linkability of sessions. Expiration shouldn&#39;t be an issue for =
WebRTC, since we can simply do ICE restart if this occurs during a session.=
</div>



<div><br></div><div>Is this something we want to recommend in the transport=
s doc?<br></div></div></blockquote><br></div></div></div><div>Yes.</div><di=
v><br></div><div>And it should be as frequent as every new &#39;call&#39;, =
if IPv6 privacy addresses are to provide the same privacy as a NAPT44, as a=
 NAPT44 should be assigning random ports per reasons described in RFC6056.<=
/div>

<div><br></div><div>The transport document should also recommend doing port=
 randomization (RFC6056), as that was found pretty useful for DNS, but rand=
omizing ports is still not popular with many OS&#39;s TCP stacks for whatev=
er reason.</div>

<span class=3D"HOEnZb"><font color=3D"#888888"><div><br></div><div>-d</div>=
<div><br></div><br></font></span></div></blockquote></div><br></div>

--001a11c210ccb43d2004f4f03de4--


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Should consent checks be optimized further?
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Though the WG's decision is to go with the STUN consent mechanism (draft-ie=
tf-rtcweb-stun-consent-freshness) instead of the DTLS consent mechanism (dr=
aft-thomson-rtcweb-consent), I think the later raises a question that we ha=
ven't discussed in the context of the former, yet:

Should reception of authenticated traffic from the peer on the inverted 5-t=
uple be considered as peer granting consent to send traffic to it? Should t=
he browser refrain from performing consent freshness when it continues to r=
eceive such traffic from the peer?

Here, authenticated traffic could be DTLS-SRTP or DTLS-SRTCP or data-channe=
l or STUN binding requests.

Some of the reasons why we may not want to do such optimization:
- To keep the mechanism simple.

- To not introduce layering violations.

- Might help network elements to piggyback metadata about change in network

  conditions to other network elements and endpoints (in line with what

  Matthew in the meeting).
- The overhead of performing consent freshness compared to processing actua=
l
  traffic is generally minimal.
- Might help in calculating RTT for the data channel (that doesn't have RTC=
P).

- Introduces a security weakness: SRTP is encrypted and authenticated with

  symmetric keys; that is, both sender and receiver know the keys. With two

  party sessions, receipt of an authenticated packet from the single remote

  party is a strong assurance the packet came from that party. However, whe=
n

  a session involves more than two parties, all of whom know each other's

  keys, any of those parties could have sent the packet. Such shared key

  distributions are possible with some MIKEY [RFC3830] modes,

  Security Descriptions [RFC4568], and EKT [I-D.ietf-avtcore-srtp-ekt].

On the other hand, there might be some constrained networks that allocate f=
ixed bit rate for traffic and the occasional consent checks could break tha=
t fixed bit rate. This looks the only benefit of not performing consent fre=
shness when receiving authenticated traffic.

It should be noted that draft-ietf-rtcweb-stun-consent-freshness already ha=
s the following:

   When not actively sending traffic on a nominated candidate pair,
   performing consent freshness does not serve any purpose from a
   security perspective.

Which itself is good optimization, I think.

Comments welcome..

Muthu


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</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Though the WG's decision is to go with the STUN consent me=
chanism (draft-ietf-rtcweb-stun-consent-freshness) instead of the DTLS cons=
ent mechanism (draft-thomson-rtcweb-consent),
 I think the later raises a question that we haven't discussed in the conte=
xt of the former, yet:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Should reception of authenticated traffic from the peer on=
 the inverted 5-tuple be considered as peer granting consent to send traffi=
c to it? Should the browser refrain from performing
 consent freshness when it continues to receive such traffic from the peer?=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Here, authenticated traffic could be DTLS-SRTP or DTLS-SRT=
CP or data-channel or STUN binding requests.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Some of the reasons why we may not want to do such optimiz=
ation:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- To keep the mechanism simple.<o:p></o:p></span></p>
<p class=3D"MsoPlainText">- To not introduce layering violations.<o:p></o:p=
></p>
<p class=3D"MsoPlainText">- Might help network elements to piggyback metada=
ta about change in network
<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;conditions to other network elements =
and endpoints (in line with what<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp; Matthew in the meeting).<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- The overhead of performing consent freshness compared to=
 processing actual
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp;traffic is generally minimal.<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">- Might help in calculating RTT for the data channel (that=
 doesn't have RTCP).<o:p></o:p></span></p>
<p class=3D"MsoPlainText">- Introduces a security weakness: SRTP is encrypt=
ed and authenticated with
<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;symmetric keys; that is, both sender =
and receiver know the keys. With two
<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;party sessions, receipt of an authent=
icated packet from the single remote
<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;party is a strong assurance the packe=
t came from that party. However, when
<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;a session involves more than two part=
ies, all of whom know each other's<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp; keys, any of those parties could have sent=
 the packet. Such shared key
<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;distributions are possible with some =
MIKEY [RFC3830] modes,
<o:p></o:p></p>
<p class=3D"MsoPlainText">&nbsp;&nbsp;Security Descriptions [RFC4568], and =
EKT [I-D.ietf-avtcore-srtp-ekt].<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">On the other hand, there might be some constrained network=
s that allocate fixed bit rate for traffic and the occasional consent check=
s could break that fixed bit rate. This looks
 the only benefit of not performing consent freshness when receiving authen=
ticated traffic.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">It should be noted that draft-ietf-rtcweb-stun-consent-fre=
shness already has the following:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; When not actively sending traffic on a nomina=
ted candidate pair,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; performing consent freshness does not serve a=
ny purpose from a<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; security perspective.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Which itself is good optimization, I think.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Comments welcome..<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Muthu<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_E721D8C6A2E1544DB2DEBC313AF54DE22E319CE5xmbrcdx02ciscoc_--


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From: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com> wrote:

> I'd like to be silent on the issue, since which IPv6 addresses to =
prefer is likely to be a matter of system policy. Trying to override =
system policy in an application specific profile usually leads to =
sadness.

The application needs to indicate if it wants a temporary address. If =
the host's policy (or configuration or network) does not support =
temporary addresses, the application won't get a temporary address.  I =
don't see why being silent helps?

-d


>=20
>=20
>=20
> On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com> wrote:
>=20
> On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com> wrote:
>=20
>> https://tools.ietf.org/html/rfc4941 defines the concept of temporary =
IPv6 addresses. For, example, as enumerated on my local system:
>>=20
>> inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
>> inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf=20
>> inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf =
temporary=20
>>=20
>> As indicated in the RFC, the temporary addresses expire after hours =
or days, and therefore could be used to prevent long-term linkability of =
sessions. Expiration shouldn't be an issue for WebRTC, since we can =
simply do ICE restart if this occurs during a session.
>>=20
>> Is this something we want to recommend in the transports doc?
>=20
> Yes.
>=20
> And it should be as frequent as every new 'call', if IPv6 privacy =
addresses are to provide the same privacy as a NAPT44, as a NAPT44 =
should be assigning random ports per reasons described in RFC6056.
>=20
> The transport document should also recommend doing port randomization =
(RFC6056), as that was found pretty useful for DNS, but randomizing =
ports is still not popular with many OS's TCP stacks for whatever =
reason.
>=20
> -d
>=20
>=20
>=20


--Apple-Mail=_FA6711E7-B681-4D2D-828D-DB5F6FB573FB
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><br><div><div>On Mar 19, 2014, at 12:02 AM, Harald =
Alvestrand &lt;<a href=3D"mailto:hta@google.com">hta@google.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"><div dir=3D"ltr">I'd like to be silent on the issue, =
since which IPv6 addresses to prefer is likely to be a matter of system =
policy. Trying to override system policy in an application specific =
profile usually leads to =
sadness.</div></blockquote><div><br></div><div>The application needs to =
indicate if it wants a temporary address. If the host's policy (or =
configuration or network) does not support temporary addresses, the =
application won't get a temporary address. &nbsp;I don't see why being =
silent =
helps?</div><div><br></div><div>-d</div><div><br></div><br><blockquote =
type=3D"cite"><div dir=3D"ltr"><div>

<br></div></div><div class=3D"gmail_extra"><br><br><div =
class=3D"gmail_quote">On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <span =
dir=3D"ltr">&lt;<a href=3D"mailto:dwing@cisco.com" =
target=3D"_blank">dwing@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div =
style=3D"word-wrap:break-word"><div><div class=3D"h5"><br><div><div>On =
Mar 18, 2014, at 6:00 PM, Justin Uberti &lt;<a =
href=3D"mailto:juberti@google.com" =
target=3D"_blank">juberti@google.com</a>&gt; wrote:</div>

<br><blockquote type=3D"cite"><div dir=3D"ltr"><a =
href=3D"https://tools.ietf.org/html/rfc4941" =
target=3D"_blank">https://tools.ietf.org/html/rfc4941</a> defines the =
concept of temporary IPv6 addresses. For, example, as enumerated on my =
local system:<br>

<div><br></div><div><span style=3D"font-family:'courier =
new',monospace;font-size:13px">inet 172.31.x.y netmask 0xfffffe00 =
broadcast 172.31.x.255</span><br style=3D"font-family:'courier =
new',monospace;font-size:13px">



<span style=3D"font-family:'courier new',monospace;font-size:13px">inet6 =
2620::1008:100b:e2f8:47ff:wwww</span><span style=3D"font-family:'courier =
new',monospace;font-size:13px">:xxxx prefixlen 64 =
autoconf&nbsp;</span><br style=3D"font-family:'courier =
new',monospace;font-size:13px">



<span style=3D"font-family:'courier new',monospace;font-size:13px">inet6 =
2620::1008:100b:819e:1d3f:yyyy</span><span style=3D"font-family:'courier =
new',monospace;font-size:13px">:zzzz prefixlen 64 autoconf =
<b>temporary&nbsp;</b></span><br>



</div><div><span style=3D"font-family:'courier =
new',monospace;font-size:13px"><br></span></div><div>As indicated in the =
RFC, the temporary addresses expire after hours or days, and therefore =
could be used to prevent long-term linkability of sessions. Expiration =
shouldn't be an issue for WebRTC, since we can simply do ICE restart if =
this occurs during a session.</div>



<div><br></div><div>Is this something we want to recommend in the =
transports =
doc?<br></div></div></blockquote><br></div></div></div><div>Yes.</div><div=
><br></div><div>And it should be as frequent as every new 'call', if =
IPv6 privacy addresses are to provide the same privacy as a NAPT44, as a =
NAPT44 should be assigning random ports per reasons described in =
RFC6056.</div>

<div><br></div><div>The transport document should also recommend doing =
port randomization (RFC6056), as that was found pretty useful for DNS, =
but randomizing ports is still not popular with many OS's TCP stacks for =
whatever reason.</div>

<span class=3D"HOEnZb"><font =
color=3D"#888888"><div><br></div><div>-d</div><div><br></div><br></font></=
span></div></blockquote></div><br></div>
</blockquote></div><br></body></html>=

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WG,

We will get this announce the official way (IETF announce list) also.

Our joint f2f meeting of the RTCWEB WG and
the W3C WebRTC WG and Media Capture TF on

*May 19th to 21st is in Washington DC USA*.

The meeting will take place in the Google office at 1101 New York Ave
NW, Washington, DC 20005.

Agenda and time split between RTCWEB WG, W3C WebRTC and Media Capture
TF will be decided later.

Cheers

Magnus Westerlund
(As chair)

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On 03/19/2014 04:08 PM, Dan Wing wrote:
>
> On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com 
> <mailto:hta@google.com>> wrote:
>
>> I'd like to be silent on the issue, since which IPv6 addresses to 
>> prefer is likely to be a matter of system policy. Trying to override 
>> system policy in an application specific profile usually leads to 
>> sadness.
>
> The application needs to indicate if it wants a temporary address. If 
> the host's policy (or configuration or network) does not support 
> temporary addresses, the application won't get a temporary address.  I 
> don't see why being silent helps?

What API is it using?

With a little Googling, the system policy I was thinking of was the 
policy in RFC 6724 ("Default Address Selection for IPv6"), in particular 
section 5 rule 7: "Prefer  temporary addresses".

I'm happy to say "it is a good idea for systems to implement the 
recommendations of RFC 6724" (or some more 2119-like language). I 
wouldn't want to claim that if a system has chosen to prefer 
non-temporary addresses, it would have to change its non-conformance to 
RFC 6724 in order to be conformant with RTCWEB specifications.

>
> -d
>
>
>>
>>
>>
>> On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com 
>> <mailto:dwing@cisco.com>> wrote:
>>
>>
>>     On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com
>>     <mailto:juberti@google.com>> wrote:
>>
>>>     https://tools.ietf.org/html/rfc4941 defines the concept of
>>>     temporary IPv6 addresses. For, example, as enumerated on my
>>>     local system:
>>>
>>>     inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
>>>     inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf
>>>     inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf
>>>     *temporary *
>>>
>>>     As indicated in the RFC, the temporary addresses expire after
>>>     hours or days, and therefore could be used to prevent long-term
>>>     linkability of sessions. Expiration shouldn't be an issue for
>>>     WebRTC, since we can simply do ICE restart if this occurs during
>>>     a session.
>>>
>>>     Is this something we want to recommend in the transports doc?
>>
>>     Yes.
>>
>>     And it should be as frequent as every new 'call', if IPv6 privacy
>>     addresses are to provide the same privacy as a NAPT44, as a
>>     NAPT44 should be assigning random ports per reasons described in
>>     RFC6056.
>>
>>     The transport document should also recommend doing port
>>     randomization (RFC6056), as that was found pretty useful for DNS,
>>     but randomizing ports is still not popular with many OS's TCP
>>     stacks for whatever reason.
>>
>>     -d
>>
>>
>>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 03/19/2014 04:08 PM, Dan Wing wrote:<br>
    </div>
    <blockquote
      cite="mid:CD747481-EBDA-4FFC-A31D-618E6E217420@cisco.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <br>
      <div>
        <div>On Mar 19, 2014, at 12:02 AM, Harald Alvestrand &lt;<a
            moz-do-not-send="true" href="mailto:hta@google.com">hta@google.com</a>&gt;
          wrote:</div>
        <br class="Apple-interchange-newline">
        <blockquote type="cite">
          <meta http-equiv="Content-Type" content="text/html;
            charset=ISO-8859-1">
          <div dir="ltr">I'd like to be silent on the issue, since which
            IPv6 addresses to prefer is likely to be a matter of system
            policy. Trying to override system policy in an application
            specific profile usually leads to sadness.</div>
        </blockquote>
        <div><br>
        </div>
        <div>The application needs to indicate if it wants a temporary
          address. If the host's policy (or configuration or network)
          does not support temporary addresses, the application won't
          get a temporary address. &nbsp;I don't see why being silent helps?</div>
      </div>
    </blockquote>
    <br>
    What API is it using?<br>
    <br>
    With a little Googling, the system policy I was thinking of was the
    policy in RFC 6724 ("Default Address Selection for IPv6"), in
    particular section 5 rule 7: "Prefer&nbsp; temporary addresses".<br>
    <br>
    I'm happy to say "it is a good idea for systems to implement the
    recommendations of RFC 6724" (or some more 2119-like language). I
    wouldn't want to claim that if a system has chosen to prefer
    non-temporary addresses, it would have to change its non-conformance
    to RFC 6724 in order to be conformant with RTCWEB specifications.<br>
    <br>
    <blockquote
      cite="mid:CD747481-EBDA-4FFC-A31D-618E6E217420@cisco.com"
      type="cite">
      <div>
        <div><br>
        </div>
        <div>-d</div>
        <div><br>
        </div>
        <br>
        <blockquote type="cite">
          <div dir="ltr">
            <div>
              <br>
            </div>
          </div>
          <div class="gmail_extra"><br>
            <br>
            <div class="gmail_quote">On Wed, Mar 19, 2014 at 6:14 AM,
              Dan Wing <span dir="ltr">&lt;<a moz-do-not-send="true"
                  href="mailto:dwing@cisco.com" target="_blank">dwing@cisco.com</a>&gt;</span>
              wrote:<br>
              <blockquote class="gmail_quote" style="margin:0 0 0
                .8ex;border-left:1px #ccc solid;padding-left:1ex">
                <div style="word-wrap:break-word">
                  <div>
                    <div class="h5"><br>
                      <div>
                        <div>On Mar 18, 2014, at 6:00 PM, Justin Uberti
                          &lt;<a moz-do-not-send="true"
                            href="mailto:juberti@google.com"
                            target="_blank">juberti@google.com</a>&gt;
                          wrote:</div>
                        <br>
                        <blockquote type="cite">
                          <div dir="ltr"><a moz-do-not-send="true"
                              href="https://tools.ietf.org/html/rfc4941"
                              target="_blank">https://tools.ietf.org/html/rfc4941</a>
                            defines the concept of temporary IPv6
                            addresses. For, example, as enumerated on my
                            local system:<br>
                            <div><br>
                            </div>
                            <div><span style="font-family:'courier
                                new',monospace;font-size:13px">inet
                                172.31.x.y netmask 0xfffffe00 broadcast
                                172.31.x.255</span><br
                                style="font-family:'courier
                                new',monospace;font-size:13px">
                              <span style="font-family:'courier
                                new',monospace;font-size:13px">inet6
                                2620::1008:100b:e2f8:47ff:wwww</span><span
                                style="font-family:'courier
                                new',monospace;font-size:13px">:xxxx
                                prefixlen 64 autoconf&nbsp;</span><br
                                style="font-family:'courier
                                new',monospace;font-size:13px">
                              <span style="font-family:'courier
                                new',monospace;font-size:13px">inet6
                                2620::1008:100b:819e:1d3f:yyyy</span><span
                                style="font-family:'courier
                                new',monospace;font-size:13px">:zzzz
                                prefixlen 64 autoconf <b>temporary&nbsp;</b></span><br>
                            </div>
                            <div><span style="font-family:'courier
                                new',monospace;font-size:13px"><br>
                              </span></div>
                            <div>As indicated in the RFC, the temporary
                              addresses expire after hours or days, and
                              therefore could be used to prevent
                              long-term linkability of sessions.
                              Expiration shouldn't be an issue for
                              WebRTC, since we can simply do ICE restart
                              if this occurs during a session.</div>
                            <div><br>
                            </div>
                            <div>Is this something we want to recommend
                              in the transports doc?<br>
                            </div>
                          </div>
                        </blockquote>
                        <br>
                      </div>
                    </div>
                  </div>
                  <div>Yes.</div>
                  <div><br>
                  </div>
                  <div>And it should be as frequent as every new 'call',
                    if IPv6 privacy addresses are to provide the same
                    privacy as a NAPT44, as a NAPT44 should be assigning
                    random ports per reasons described in RFC6056.</div>
                  <div><br>
                  </div>
                  <div>The transport document should also recommend
                    doing port randomization (RFC6056), as that was
                    found pretty useful for DNS, but randomizing ports
                    is still not popular with many OS's TCP stacks for
                    whatever reason.</div>
                  <span class="HOEnZb"><font color="#888888">
                      <div><br>
                      </div>
                      <div>-d</div>
                      <div><br>
                      </div>
                      <br>
                    </font></span></div>
              </blockquote>
            </div>
            <br>
          </div>
        </blockquote>
      </div>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------050300090907080509030004--


From nobody Wed Mar 19 08:39:17 2014
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Le 2014-03-19 03:02, Harald Alvestrand a écrit :
> I'd like to be silent on the issue, since which IPv6 addresses to prefer
> is likely to be a matter of system policy. Trying to override system
> policy in an application specific profile usually leads to sadness.

What would be useful to mention IMHO is that ICE restart should be used
to gracefully handle the issue of IP address expiry, of which temporary
IPv6 addresses are a frequently-encountered example.

Simon
-- 
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Le 2014-03-19 11:21, Harald Alvestrand a écrit :
>> The application needs to indicate if it wants a temporary address. If
>> the host's policy (or configuration or network) does not support
>> temporary addresses, the application won't get a temporary address.  I
>> don't see why being silent helps?
> 
> What API is it using?
> 
> With a little Googling, the system policy I was thinking of was the
> policy in RFC 6724 ("Default Address Selection for IPv6"), in particular
> section 5 rule 7: "Prefer  temporary addresses".
> 
> I'm happy to say "it is a good idea for systems to implement the
> recommendations of RFC 6724" (or some more 2119-like language). I
> wouldn't want to claim that if a system has chosen to prefer
> non-temporary addresses, it would have to change its non-conformance to
> RFC 6724 in order to be conformant with RTCWEB specifications.

IMHO it would be perfectly sensible to recommend the use of
IPV6_PREFER_SRC_TMP [RFC5014].

Simon
-- 
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From: Justin Uberti <juberti@google.com>
Date: Wed, 19 Mar 2014 09:08:49 -0700
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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ICE by default is going to gather host candidates from all interfaces,
which means that the non-temporary addresses are going to be used,
regardless of priority, for connectivity checks at least.

As a result, if we don't do anything, it means 2x the v6 interfaces, and 4x
the v6-v6 ICE checks, in addition to any linkability concerns.

We could say that if a system has chosen to prefer non-temporary-addresses,
or privacy is not a concern, then non-temporary-addresses can be used.


On Wed, Mar 19, 2014 at 8:21 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  On 03/19/2014 04:08 PM, Dan Wing wrote:
>
>
>  On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com> wrote:
>
>  I'd like to be silent on the issue, since which IPv6 addresses to prefer
> is likely to be a matter of system policy. Trying to override system policy
> in an application specific profile usually leads to sadness.
>
>
>  The application needs to indicate if it wants a temporary address. If
> the host's policy (or configuration or network) does not support temporary
> addresses, the application won't get a temporary address.  I don't see why
> being silent helps?
>
>
> What API is it using?
>
> With a little Googling, the system policy I was thinking of was the policy
> in RFC 6724 ("Default Address Selection for IPv6"), in particular section 5
> rule 7: "Prefer  temporary addresses".
>
> I'm happy to say "it is a good idea for systems to implement the
> recommendations of RFC 6724" (or some more 2119-like language). I wouldn't
> want to claim that if a system has chosen to prefer non-temporary
> addresses, it would have to change its non-conformance to RFC 6724 in order
> to be conformant with RTCWEB specifications.
>
>
>  -d
>
>
>
>
>
> On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com> wrote:
>
>>
>>  On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com> wrote:
>>
>>  https://tools.ietf.org/html/rfc4941 defines the concept of temporary
>> IPv6 addresses. For, example, as enumerated on my local system:
>>
>>  inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
>> inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf
>> inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf
>> *temporary *
>>
>>  As indicated in the RFC, the temporary addresses expire after hours or
>> days, and therefore could be used to prevent long-term linkability of
>> sessions. Expiration shouldn't be an issue for WebRTC, since we can simply
>> do ICE restart if this occurs during a session.
>>
>>  Is this something we want to recommend in the transports doc?
>>
>>
>>   Yes.
>>
>>  And it should be as frequent as every new 'call', if IPv6 privacy
>> addresses are to provide the same privacy as a NAPT44, as a NAPT44 should
>> be assigning random ports per reasons described in RFC6056.
>>
>>  The transport document should also recommend doing port randomization
>> (RFC6056), as that was found pretty useful for DNS, but randomizing ports
>> is still not popular with many OS's TCP stacks for whatever reason.
>>
>>  -d
>>
>>
>>
>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--f46d0444ee91932c7704f4f7df79
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">ICE by default is going to gather host candidates from all=
 interfaces, which means that the non-temporary addresses are going to be u=
sed, regardless of priority, for connectivity checks at least.<div><br></di=
v>

<div>As a result, if we don&#39;t do anything, it means 2x the v6 interface=
s, and 4x the v6-v6 ICE checks, in addition to any linkability concerns.=C2=
=A0</div><div><br></div><div>We could say that if a system has chosen to pr=
efer non-temporary-addresses, or privacy is not a concern, then non-tempora=
ry-addresses can be used.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed,=
 Mar 19, 2014 at 8:21 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=
=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>=
&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"">
    <div>On 03/19/2014 04:08 PM, Dan Wing wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <br>
      <div>
        <div>On Mar 19, 2014, at 12:02 AM, Harald Alvestrand &lt;<a href=3D=
"mailto:hta@google.com" target=3D"_blank">hta@google.com</a>&gt;
          wrote:</div>
        <br>
        <blockquote type=3D"cite">
         =20
          <div dir=3D"ltr">I&#39;d like to be silent on the issue, since wh=
ich
            IPv6 addresses to prefer is likely to be a matter of system
            policy. Trying to override system policy in an application
            specific profile usually leads to sadness.</div>
        </blockquote>
        <div><br>
        </div>
        <div>The application needs to indicate if it wants a temporary
          address. If the host&#39;s policy (or configuration or network)
          does not support temporary addresses, the application won&#39;t
          get a temporary address. =C2=A0I don&#39;t see why being silent h=
elps?</div>
      </div>
    </blockquote>
    <br></div>
    What API is it using?<br>
    <br>
    With a little Googling, the system policy I was thinking of was the
    policy in RFC 6724 (&quot;Default Address Selection for IPv6&quot;), in
    particular section 5 rule 7: &quot;Prefer=C2=A0 temporary addresses&quo=
t;.<br>
    <br>
    I&#39;m happy to say &quot;it is a good idea for systems to implement t=
he
    recommendations of RFC 6724&quot; (or some more 2119-like language). I
    wouldn&#39;t want to claim that if a system has chosen to prefer
    non-temporary addresses, it would have to change its non-conformance
    to RFC 6724 in order to be conformant with RTCWEB specifications.<br>
    <br>
    <blockquote type=3D"cite"><div class=3D"">
      <div>
        <div><br>
        </div>
        <div>-d</div>
        <div><br>
        </div>
        <br>
        <blockquote type=3D"cite">
          <div dir=3D"ltr">
            <div>
              <br>
            </div>
          </div>
          <div class=3D"gmail_extra"><br>
            <br>
            <div class=3D"gmail_quote">On Wed, Mar 19, 2014 at 6:14 AM,
              Dan Wing <span dir=3D"ltr">&lt;<a href=3D"mailto:dwing@cisco.=
com" target=3D"_blank">dwing@cisco.com</a>&gt;</span>
              wrote:<br>
              <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
                <div style=3D"word-wrap:break-word">
                  <div>
                    <div><br>
                      <div>
                        <div>On Mar 18, 2014, at 6:00 PM, Justin Uberti
                          &lt;<a href=3D"mailto:juberti@google.com" target=
=3D"_blank">juberti@google.com</a>&gt;
                          wrote:</div>
                        <br>
                        <blockquote type=3D"cite">
                          <div dir=3D"ltr"><a href=3D"https://tools.ietf.or=
g/html/rfc4941" target=3D"_blank">https://tools.ietf.org/html/rfc4941</a>
                            defines the concept of temporary IPv6
                            addresses. For, example, as enumerated on my
                            local system:<br>
                            <div><br>
                            </div>
                            <div><span>inet
                                172.31.x.y netmask 0xfffffe00 broadcast
                                172.31.x.255</span><br>
                              <span>inet6
                                2620::1008:100b:e2f8:47ff:wwww</span><span>=
:xxxx
                                prefixlen 64 autoconf=C2=A0</span><br>
                              <span>inet6
                                2620::1008:100b:819e:1d3f:yyyy</span><span>=
:zzzz
                                prefixlen 64 autoconf <b>temporary=C2=A0</b=
></span><br>
                            </div>
                            <div><span><br>
                              </span></div>
                            <div>As indicated in the RFC, the temporary
                              addresses expire after hours or days, and
                              therefore could be used to prevent
                              long-term linkability of sessions.
                              Expiration shouldn&#39;t be an issue for
                              WebRTC, since we can simply do ICE restart
                              if this occurs during a session.</div>
                            <div><br>
                            </div>
                            <div>Is this something we want to recommend
                              in the transports doc?<br>
                            </div>
                          </div>
                        </blockquote>
                        <br>
                      </div>
                    </div>
                  </div>
                  <div>Yes.</div>
                  <div><br>
                  </div>
                  <div>And it should be as frequent as every new &#39;call&=
#39;,
                    if IPv6 privacy addresses are to provide the same
                    privacy as a NAPT44, as a NAPT44 should be assigning
                    random ports per reasons described in RFC6056.</div>
                  <div><br>
                  </div>
                  <div>The transport document should also recommend
                    doing port randomization (RFC6056), as that was
                    found pretty useful for DNS, but randomizing ports
                    is still not popular with many OS&#39;s TCP stacks for
                    whatever reason.</div>
                  <span><font color=3D"#888888">
                      <div><br>
                      </div>
                      <div>-d</div>
                      <div><br>
                      </div>
                      <br>
                    </font></span></div>
              </blockquote>
            </div>
            <br>
          </div>
        </blockquote>
      </div>
      <br>
      <br>
      <fieldset></fieldset>
      <br>
      </div><pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--f46d0444ee91932c7704f4f7df79--


From nobody Wed Mar 19 11:18:35 2014
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From: Martin Thomson <martin.thomson@gmail.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Subject: Re: [rtcweb] Should consent checks be optimized further?
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On 19 March 2014 08:01, Muthu Arul Mozhi Perumal (mperumal)
<mperumal@cisco.com> wrote:
> Should reception of authenticated traffic from the peer on the inverted
> 5-tuple be considered as peer granting consent to send traffic to it? Should
> the browser refrain from performing consent freshness when it continues to
> receive such traffic from the peer?


You forgot to mention another concern here, which is that the receipt
of an authenticated packet is not sufficient if the path has changed.
You need to verify that the peer on the new path has both consented to
receive data AND is in possession of the session keys.  Otherwise, in
combination with source IP/port spoofing, you get an attack that a
victim cannot opt out of.


From nobody Wed Mar 19 13:44:21 2014
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From: Justin Uberti <juberti@google.com>
Date: Wed, 19 Mar 2014 13:43:47 -0700
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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(and, to be clear, that by default, non-temporary addresses SHOULD NOT be
used by clients)


On Wed, Mar 19, 2014 at 9:08 AM, Justin Uberti <juberti@google.com> wrote:

> ICE by default is going to gather host candidates from all interfaces,
> which means that the non-temporary addresses are going to be used,
> regardless of priority, for connectivity checks at least.
>
> As a result, if we don't do anything, it means 2x the v6 interfaces, and
> 4x the v6-v6 ICE checks, in addition to any linkability concerns.
>
> We could say that if a system has chosen to prefer
> non-temporary-addresses, or privacy is not a concern, then
> non-temporary-addresses can be used.
>
>
> On Wed, Mar 19, 2014 at 8:21 AM, Harald Alvestrand <harald@alvestrand.no>wrote:
>
>>  On 03/19/2014 04:08 PM, Dan Wing wrote:
>>
>>
>>  On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com> wrote:
>>
>>  I'd like to be silent on the issue, since which IPv6 addresses to
>> prefer is likely to be a matter of system policy. Trying to override system
>> policy in an application specific profile usually leads to sadness.
>>
>>
>>  The application needs to indicate if it wants a temporary address. If
>> the host's policy (or configuration or network) does not support temporary
>> addresses, the application won't get a temporary address.  I don't see why
>> being silent helps?
>>
>>
>> What API is it using?
>>
>> With a little Googling, the system policy I was thinking of was the
>> policy in RFC 6724 ("Default Address Selection for IPv6"), in particular
>> section 5 rule 7: "Prefer  temporary addresses".
>>
>> I'm happy to say "it is a good idea for systems to implement the
>> recommendations of RFC 6724" (or some more 2119-like language). I wouldn't
>> want to claim that if a system has chosen to prefer non-temporary
>> addresses, it would have to change its non-conformance to RFC 6724 in order
>> to be conformant with RTCWEB specifications.
>>
>>
>>  -d
>>
>>
>>
>>
>>
>> On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com> wrote:
>>
>>>
>>>  On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com> wrote:
>>>
>>>  https://tools.ietf.org/html/rfc4941 defines the concept of temporary
>>> IPv6 addresses. For, example, as enumerated on my local system:
>>>
>>>  inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
>>> inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf
>>> inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf
>>> *temporary *
>>>
>>>  As indicated in the RFC, the temporary addresses expire after hours or
>>> days, and therefore could be used to prevent long-term linkability of
>>> sessions. Expiration shouldn't be an issue for WebRTC, since we can simply
>>> do ICE restart if this occurs during a session.
>>>
>>>  Is this something we want to recommend in the transports doc?
>>>
>>>
>>>   Yes.
>>>
>>>  And it should be as frequent as every new 'call', if IPv6 privacy
>>> addresses are to provide the same privacy as a NAPT44, as a NAPT44 should
>>> be assigning random ports per reasons described in RFC6056.
>>>
>>>  The transport document should also recommend doing port randomization
>>> (RFC6056), as that was found pretty useful for DNS, but randomizing ports
>>> is still not popular with many OS's TCP stacks for whatever reason.
>>>
>>>  -d
>>>
>>>
>>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

--bcaec52d4acfc5375904f4fbb622
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">(and, to be clear, that by default, non-temporary addresse=
s SHOULD NOT be used by clients)</div><div class=3D"gmail_extra"><br><br><d=
iv class=3D"gmail_quote">On Wed, Mar 19, 2014 at 9:08 AM, Justin Uberti <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">=
juberti@google.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">ICE by default is going to =
gather host candidates from all interfaces, which means that the non-tempor=
ary addresses are going to be used, regardless of priority, for connectivit=
y checks at least.<div>

<br></div>
<div>As a result, if we don&#39;t do anything, it means 2x the v6 interface=
s, and 4x the v6-v6 ICE checks, in addition to any linkability concerns.=C2=
=A0</div><div><br></div><div>We could say that if a system has chosen to pr=
efer non-temporary-addresses, or privacy is not a concern, then non-tempora=
ry-addresses can be used.</div>


</div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><b=
r><br><div class=3D"gmail_quote">On Wed, Mar 19, 2014 at 8:21 AM, Harald Al=
vestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" targ=
et=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div>
    <div>On 03/19/2014 04:08 PM, Dan Wing wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <br>
      <div>
        <div>On Mar 19, 2014, at 12:02 AM, Harald Alvestrand &lt;<a href=3D=
"mailto:hta@google.com" target=3D"_blank">hta@google.com</a>&gt;
          wrote:</div>
        <br>
        <blockquote type=3D"cite">
         =20
          <div dir=3D"ltr">I&#39;d like to be silent on the issue, since wh=
ich
            IPv6 addresses to prefer is likely to be a matter of system
            policy. Trying to override system policy in an application
            specific profile usually leads to sadness.</div>
        </blockquote>
        <div><br>
        </div>
        <div>The application needs to indicate if it wants a temporary
          address. If the host&#39;s policy (or configuration or network)
          does not support temporary addresses, the application won&#39;t
          get a temporary address. =C2=A0I don&#39;t see why being silent h=
elps?</div>
      </div>
    </blockquote>
    <br></div>
    What API is it using?<br>
    <br>
    With a little Googling, the system policy I was thinking of was the
    policy in RFC 6724 (&quot;Default Address Selection for IPv6&quot;), in
    particular section 5 rule 7: &quot;Prefer=C2=A0 temporary addresses&quo=
t;.<br>
    <br>
    I&#39;m happy to say &quot;it is a good idea for systems to implement t=
he
    recommendations of RFC 6724&quot; (or some more 2119-like language). I
    wouldn&#39;t want to claim that if a system has chosen to prefer
    non-temporary addresses, it would have to change its non-conformance
    to RFC 6724 in order to be conformant with RTCWEB specifications.<br>
    <br>
    <blockquote type=3D"cite"><div>
      <div>
        <div><br>
        </div>
        <div>-d</div>
        <div><br>
        </div>
        <br>
        <blockquote type=3D"cite">
          <div dir=3D"ltr">
            <div>
              <br>
            </div>
          </div>
          <div class=3D"gmail_extra"><br>
            <br>
            <div class=3D"gmail_quote">On Wed, Mar 19, 2014 at 6:14 AM,
              Dan Wing <span dir=3D"ltr">&lt;<a href=3D"mailto:dwing@cisco.=
com" target=3D"_blank">dwing@cisco.com</a>&gt;</span>
              wrote:<br>
              <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
                <div style=3D"word-wrap:break-word">
                  <div>
                    <div><br>
                      <div>
                        <div>On Mar 18, 2014, at 6:00 PM, Justin Uberti
                          &lt;<a href=3D"mailto:juberti@google.com" target=
=3D"_blank">juberti@google.com</a>&gt;
                          wrote:</div>
                        <br>
                        <blockquote type=3D"cite">
                          <div dir=3D"ltr"><a href=3D"https://tools.ietf.or=
g/html/rfc4941" target=3D"_blank">https://tools.ietf.org/html/rfc4941</a>
                            defines the concept of temporary IPv6
                            addresses. For, example, as enumerated on my
                            local system:<br>
                            <div><br>
                            </div>
                            <div><span>inet
                                172.31.x.y netmask 0xfffffe00 broadcast
                                172.31.x.255</span><br>
                              <span>inet6
                                2620::1008:100b:e2f8:47ff:wwww</span><span>=
:xxxx
                                prefixlen 64 autoconf=C2=A0</span><br>
                              <span>inet6
                                2620::1008:100b:819e:1d3f:yyyy</span><span>=
:zzzz
                                prefixlen 64 autoconf <b>temporary=C2=A0</b=
></span><br>
                            </div>
                            <div><span><br>
                              </span></div>
                            <div>As indicated in the RFC, the temporary
                              addresses expire after hours or days, and
                              therefore could be used to prevent
                              long-term linkability of sessions.
                              Expiration shouldn&#39;t be an issue for
                              WebRTC, since we can simply do ICE restart
                              if this occurs during a session.</div>
                            <div><br>
                            </div>
                            <div>Is this something we want to recommend
                              in the transports doc?<br>
                            </div>
                          </div>
                        </blockquote>
                        <br>
                      </div>
                    </div>
                  </div>
                  <div>Yes.</div>
                  <div><br>
                  </div>
                  <div>And it should be as frequent as every new &#39;call&=
#39;,
                    if IPv6 privacy addresses are to provide the same
                    privacy as a NAPT44, as a NAPT44 should be assigning
                    random ports per reasons described in RFC6056.</div>
                  <div><br>
                  </div>
                  <div>The transport document should also recommend
                    doing port randomization (RFC6056), as that was
                    found pretty useful for DNS, but randomizing ports
                    is still not popular with many OS&#39;s TCP stacks for
                    whatever reason.</div>
                  <span><font color=3D"#888888">
                      <div><br>
                      </div>
                      <div>-d</div>
                      <div><br>
                      </div>
                      <br>
                    </font></span></div>
              </blockquote>
            </div>
            <br>
          </div>
        </blockquote>
      </div>
      <br>
      <br>
      <fieldset></fieldset>
      <br>
      </div><pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--bcaec52d4acfc5375904f4fbb622--


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From: Justin Uberti <juberti@google.com>
Date: Wed, 19 Mar 2014 17:21:05 -0700
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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On Mon, Mar 17, 2014 at 6:23 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-14 17:26, Justin Uberti wrote:
> >
> >
> >
> > On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund
> > <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>=
>
> > wrote:
> >
> >     Hi,
> >     (as individual)
> >
> >     Although this discussion really should be on MMUSIC, I am still
> >     following up here. Anything we come to conclude here really need to
> go
> >     to MMUSIC for confirmation.
> >
> >     Justin, I do agree that it appear unnecessarily of having to includ=
e
> an
> >     RTCP transport in an OFFER when the offerer know the answerer is
> BUNDLE
> >     capable. I am fine with that and think the rule needed here is
> something
> >     along the line of the below:
> >
> >     A BUNDLE supporting endpoint MUST implement a=3Drtcp-mux (that is
> already
> >     required). In any answer by a BUNDLE capable endpoint, it MUST NOT
> >     change the usage of a=3Drtcp-mux. If it is present in the offer, it
> must
> >     be confirmed to be used in the answer, and if not present in the
> offer,
> >     it MUST NOT be added in the answer. An BUNDLE capable endpoint is
> >     RECOMMENDED to use a=3Drtcp-mux whenever suitable. Cases when it mi=
ght
> not
> >     be suitable include, lack RTP payload types, or cases when RTCP
> traffic
> >     is needed to be on its own transport flow (5-tuple) to enable QoS o=
r
> >     other traffic handling functionalities.
> >
> >     The reason for writing the rule like the above, is that then an
> endpoint
> >     can choose when creating an offer to renegotiate the use of
> a=3Drtcp-mux.
> >     For example due to it running out of available PTs.
> >
> >     Feedback on this proposal?
> >
> >
> > I am not OK with this. It just gets too complicated to deal with all th=
e
> > cases, especially if with the provision that rtcp-mux can change during
> > the session.
>
> (still as individual)
>
> Lets be clear what we are discussing here. I don't think this is lot of
> cases. And if you don't want to implement this in your code generating
> offer, that is fine. What this describes is that your code handling
> reception of offers and generating answers must be able to handle the
> lack of a=3Drtcp-mux in any offer, rather than just the initial one.
>
> I would question if even that assumption holds considering what happens
> if people start implementing session resumption or session mobility
> between devices using rehydration style establishment procedures. What
> one sides consider an initial offer might not be the state the answering
> party is in. Thus, I am worried about cases that are first offer/answer
> exchange related.
>
> To my understanding you will have code to handle the following cases
> when offers contains:
>
> 1) No bundle, no a=3Drtcp-mux
> 2) No bundle, with a=3Drtcp-mux
> 3) Bundle, with a=3Drtcp-mux
>
> Considering that you support bundle, and you support usage or not of
> a=3Drtcp-mux. Then I don't see how the code handling the SDP logic for
> Bundle, without a=3Drtcp-mux is that significant.
>

It is significant because you always know you can rtcp-mux. And you always
know you can BUNDLE when rtcp-muxing. But I think trying to BUNDLE a
no-rtcp-mux m=3D line into a rtcp-mux m=3D line is asking for problems.

>
> First of all, to be robust, you have to be capable of handling
> significant reconfigurations of the Peer Connection session in an Offer.
> If someone moves the peer of one peer connection to another device
> anything can happen, and everything can change.
>
> Secondly, to be able to turn off a=3Drtcp-mux is one method which can fre=
e
> up PTs if one run out. The other would be first unbundle the media
> types, then to completely unbundle the media sources. The later would be
> far worse when it comes to requiring providing ICE candidates and
> dealing with new transports. I however think unbundling the media types
> is likely to resolve many of the PT shortage situations.
>
> Just to confirm, you are also against allowing any initial offer request
> with bundle but without a=3Drtcp-mux within that bundle-group?
>

I am against _any_ offer request with bundle and without a=3Drtcp-mux (as
indicated in the current bundle draft).

>
> This, I would question, as it prevents anyone from treating the RTCP
> packets differently from the RTP packets within one RTP session. I know
> that this has been done in some cases, including IMS.
>
> From a principal point of view I think this should be either be use of
> BUNDLE groups results in MUST use a=3Drtcp-mux with RTP or that any offer
> may change the usage of a=3Drtcp-mux. Anything else depending on initial
> offer appears very messy and prone to errors.
>
> I would appreciate if someone else would provide their input and views
> here!
>
> First on the question if they need to separate the RTP and RTCP at all
> within a BUNDLE group?
>
> Secondly if they see that splitting a bundle group into different groups
> if one runs into issue is a sufficient and more preferred method for
> dealing with any PT exhaustion issue?
>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Mar 17, 2014 at 6:23 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 2014-03-14 17:26, Justin =
Uberti wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Mar 14, 2014 at 1:57 AM, Magnus Westerlund<br>
</div>&gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.wes=
terlund@ericsson.com</a> &lt;mailto:<a href=3D"mailto:magnus.westerlund@eri=
csson.com">magnus.westerlund@ericsson.com</a>&gt;&gt;<br>
<div><div class=3D"h5">&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Hi,<br>
&gt; =C2=A0 =C2=A0 (as individual)<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Although this discussion really should be on MMUSIC, I a=
m still<br>
&gt; =C2=A0 =C2=A0 following up here. Anything we come to conclude here rea=
lly need to go<br>
&gt; =C2=A0 =C2=A0 to MMUSIC for confirmation.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Justin, I do agree that it appear unnecessarily of havin=
g to include an<br>
&gt; =C2=A0 =C2=A0 RTCP transport in an OFFER when the offerer know the ans=
werer is BUNDLE<br>
&gt; =C2=A0 =C2=A0 capable. I am fine with that and think the rule needed h=
ere is something<br>
&gt; =C2=A0 =C2=A0 along the line of the below:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 A BUNDLE supporting endpoint MUST implement a=3Drtcp-mux=
 (that is already<br>
&gt; =C2=A0 =C2=A0 required). In any answer by a BUNDLE capable endpoint, i=
t MUST NOT<br>
&gt; =C2=A0 =C2=A0 change the usage of a=3Drtcp-mux. If it is present in th=
e offer, it must<br>
&gt; =C2=A0 =C2=A0 be confirmed to be used in the answer, and if not presen=
t in the offer,<br>
&gt; =C2=A0 =C2=A0 it MUST NOT be added in the answer. An BUNDLE capable en=
dpoint is<br>
&gt; =C2=A0 =C2=A0 RECOMMENDED to use a=3Drtcp-mux whenever suitable. Cases=
 when it might not<br>
&gt; =C2=A0 =C2=A0 be suitable include, lack RTP payload types, or cases wh=
en RTCP traffic<br>
&gt; =C2=A0 =C2=A0 is needed to be on its own transport flow (5-tuple) to e=
nable QoS or<br>
&gt; =C2=A0 =C2=A0 other traffic handling functionalities.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 The reason for writing the rule like the above, is that =
then an endpoint<br>
&gt; =C2=A0 =C2=A0 can choose when creating an offer to renegotiate the use=
 of a=3Drtcp-mux.<br>
&gt; =C2=A0 =C2=A0 For example due to it running out of available PTs.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Feedback on this proposal?<br>
&gt;<br>
&gt;<br>
&gt; I am not OK with this. It just gets too complicated to deal with all t=
he<br>
&gt; cases, especially if with the provision that rtcp-mux can change durin=
g<br>
&gt; the session.<br>
<br>
</div></div>(still as individual)<br>
<br>
Lets be clear what we are discussing here. I don&#39;t think this is lot of=
<br>
cases. And if you don&#39;t want to implement this in your code generating<=
br>
offer, that is fine. What this describes is that your code handling<br>
reception of offers and generating answers must be able to handle the<br>
lack of a=3Drtcp-mux in any offer, rather than just the initial one.<br>
<br>
I would question if even that assumption holds considering what happens<br>
if people start implementing session resumption or session mobility<br>
between devices using rehydration style establishment procedures. What<br>
one sides consider an initial offer might not be the state the answering<br=
>
party is in. Thus, I am worried about cases that are first offer/answer<br>
exchange related.<br>
<br>
To my understanding you will have code to handle the following cases<br>
when offers contains:<br>
<br>
1) No bundle, no a=3Drtcp-mux<br>
2) No bundle, with a=3Drtcp-mux<br>
3) Bundle, with a=3Drtcp-mux<br>
<br>
Considering that you support bundle, and you support usage or not of<br>
a=3Drtcp-mux. Then I don&#39;t see how the code handling the SDP logic for<=
br>
Bundle, without a=3Drtcp-mux is that significant.<br></blockquote><div><br>=
</div><div>It is significant because you always know you can rtcp-mux. And =
you always know you can BUNDLE when rtcp-muxing. But I think trying to BUND=
LE a no-rtcp-mux m=3D line into a rtcp-mux m=3D line is asking for problems=
.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
First of all, to be robust, you have to be capable of handling<br>
significant reconfigurations of the Peer Connection session in an Offer.<br=
>
If someone moves the peer of one peer connection to another device<br>
anything can happen, and everything can change.<br>
<br>
Secondly, to be able to turn off a=3Drtcp-mux is one method which can free<=
br>
up PTs if one run out. The other would be first unbundle the media<br>
types, then to completely unbundle the media sources. The later would be<br=
>
far worse when it comes to requiring providing ICE candidates and<br>
dealing with new transports. I however think unbundling the media types<br>
is likely to resolve many of the PT shortage situations.<br>
<br>
Just to confirm, you are also against allowing any initial offer request<br=
>
with bundle but without a=3Drtcp-mux within that bundle-group?<br></blockqu=
ote><div><br></div><div>I am against _any_ offer request with bundle and wi=
thout a=3Drtcp-mux (as indicated in the current bundle draft).=C2=A0</div><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">


<br>
This, I would question, as it prevents anyone from treating the RTCP<br>
packets differently from the RTP packets within one RTP session. I know<br>
that this has been done in some cases, including IMS.<br>
<br>
>From a principal point of view I think this should be either be use of<br>
BUNDLE groups results in MUST use a=3Drtcp-mux with RTP or that any offer<b=
r>
may change the usage of a=3Drtcp-mux. Anything else depending on initial<br=
>
offer appears very messy and prone to errors.<br>
<br>
I would appreciate if someone else would provide their input and views<br>
here!<br>
<br>
First on the question if they need to separate the RTP and RTCP at all<br>
within a BUNDLE group?<br>
<br>
Secondly if they see that splitting a bundle group into different groups<br=
>
if one runs into issue is a sufficient and more preferred method for<br>
dealing with any PT exhaustion issue?<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7=
148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+4=
6 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br></div></div>

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Subject: Re: [rtcweb] CNAMEs and multiple peer connections
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Good point. So CNAME is perhaps the most important in terms of linkability,
but DTLS certificate and also IPv6 address may allow linkage by some
network elements.


On Mon, Mar 17, 2014 at 3:29 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-14 17:44, Justin Uberti wrote:
>
> >
> > At an implementation level, one could imagine at least 3 policies for
> > generating CNAMEs:
> > a) per-session (i.e. per-PeerConnection)
> > b) per-page (i.e. shared between all PCs on a page)
> > c) per-page, persistent (i.e. shared between all PCs on a page,
> > including across page loads)
> >
> > While we seem to agree that a) is the right solution for CNAMEs, it is
> > worth pointing out that we (Chrome) are currently doing c) for DTLS
> > certificates, to avoid performance problems with cert generation at pag=
e
> > load. Ergo, this linkability concern already exists, and I don't think
> > it is easy to solve it in the default case. There have been some
> > proposals to allow generation/storage of unique certs to prevent this
> > linkability, but this will require app input.
> >
> > Ergo, we might want to match the DTLS behavior (i.e. generate unique
> > CNAMEs only when the certs are unique), to ensure we treat linkability
> > consistently.
>
> Actually, the DTLS cert and the CNAME is actually not equivalent when it
> comes to visibility scope. The DTLS is show only to the DTLS peer, i.e.
> the address at the other end of a peer connection. The CNAME in cases of
> SFM or RTP mixer based using CSRC lists type of RTP middleboxes, can
> result in the CNAME being forward to all participants in the same
> multi-party conference.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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<div dir=3D"ltr">Good point. So CNAME is perhaps the most important in term=
s of linkability, but DTLS certificate and also IPv6 address may allow link=
age by some network elements.</div><div class=3D"gmail_extra"><br><br><div =
class=3D"gmail_quote">

On Mon, Mar 17, 2014 at 3:29 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.we=
sterlund@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">

<div class=3D"">On 2014-03-14 17:44, Justin Uberti wrote:<br>
<br>
&gt;<br>
&gt; At an implementation level, one could imagine at least 3 policies for<=
br>
&gt; generating CNAMEs:<br>
&gt; a) per-session (i.e. per-PeerConnection)<br>
&gt; b) per-page (i.e. shared between all PCs on a page)<br>
&gt; c) per-page, persistent (i.e. shared between all PCs on a page,<br>
&gt; including across page loads)<br>
&gt;<br>
&gt; While we seem to agree that a) is the right solution for CNAMEs, it is=
<br>
&gt; worth pointing out that we (Chrome) are currently doing c) for DTLS<br=
>
&gt; certificates, to avoid performance problems with cert generation at pa=
ge<br>
&gt; load. Ergo, this linkability concern already exists, and I don&#39;t t=
hink<br>
&gt; it is easy to solve it in the default case. There have been some<br>
&gt; proposals to allow generation/storage of unique certs to prevent this<=
br>
&gt; linkability, but this will require app input.<br>
&gt;<br>
&gt; Ergo, we might want to match the DTLS behavior (i.e. generate unique<b=
r>
&gt; CNAMEs only when the certs are unique), to ensure we treat linkability=
<br>
&gt; consistently.<br>
<br>
</div>Actually, the DTLS cert and the CNAME is actually not equivalent when=
 it<br>
comes to visibility scope. The DTLS is show only to the DTLS peer, i.e.<br>
the address at the other end of a peer connection. The CNAME in cases of<br=
>
SFM or RTP mixer based using CSRC lists type of RTP middleboxes, can<br>
result in the CNAME being forward to all participants in the same<br>
multi-party conference.<br>
<br>
Cheers<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Magnus Westerlund<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | Phone=
 =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7=
148287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 | Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+4=
6 73 0949079</a><br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br></div>

--e89a8f92190653db7204f4fec737--


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From: Justin Uberti <juberti@google.com>
Date: Wed, 19 Mar 2014 17:27:29 -0700
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Subject: Re: [rtcweb] On ptimes and maxptimes
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On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> (As individual)
>
> On 2014-03-05 17:26, Justin Uberti wrote:
> > From the session this morning, I recall this consensus:
> >
> >   * We will add recommendations on the frame sizes to use for PCMU/A to
> >     the rtcweb audio codec I-D (20, 30, 60 ms frames).
>
> The above applies to sending payloads
>

Right, thanks for pointing that out.

>
> Yes, but that was given that a receiver will be capable of receiving any
> valid number of frames, or samples within a reasonable packetization,
> and I think that should be 120 ms.
>
> >   * We will consider adding an a=maxptime attribute that represents the
> >     "minimum maxptime" of all the offered codecs. That is, if both Opus
> >     (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a
> >     maxptime=60 SHOULD be included.
> >       o Since maxptime spans all codecs, this means that some modes
> >         (e.g. Opus 120ms) will be unavailable, unless only Opus is
> offered.
>
> I think you base the assumption on maxptime based on the sending
> recommendations, not what is capable of receiving. If one is capable of
> receiving 200 ms of a-law and Opus, then you would need to say 120 ms as
> Opus will limit you.
>

I agree that Opus' maxptime is 120, but should that prevent sending PCMU
with 200 ms? That is, since sending Opus with 200 ms is not legal, is there
any point in forcing WebRTC impls to indicate this in SDP?


>
> >   * We will consider adding an a=ptime attribute, set to the receiver's
> >     preferred frame size to receive. Again, this value spans all codecs,
> >     so the specified value may not be viable for all codecs (e.g. 30 ms
> >     works for PCMU but not for Opus)
> >
> > After thinking about this some, I'm not sure the maxptime/ptime values
> > will lead to any different behavior than if they were not specified.
> > Since there is a complexity cost (especially if the values need to
> > change based on which codecs are offered), is there a clear upside here?
>
> The maxptime values may in some cases be limited downwards for
> interoperability, for example if one like to gateway AMR into a CS
> network then the gateway might indicate a maxptime that is less, more
> like 20 or 40 ms.
>

Sure - WebRTC impls need to respect any maxptime value that they receive,
but it's not clear to me that they need to include one in the SDP they
send.

>
> Please remember that these rules do apply to any codec, not only the MTI
> ones.
>

Yes.

>
>

--001a11334c38cc8e4004f4fed6d4
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
(As individual)<br>
<div class=3D""><br>
On 2014-03-05 17:26, Justin Uberti wrote:<br>
&gt; From the session this morning, I recall this consensus:<br>
&gt;<br>
</div>&gt; =C2=A0 * We will add recommendations on the frame sizes to use f=
or PCMU/A to<br>
<div class=3D"">&gt; =C2=A0 =C2=A0 the rtcweb audio codec I-D (20, 30, 60 m=
s frames).<br>
<br>
</div>The above applies to sending payloads<br></blockquote><div><br></div>=
<div>Right, thanks for pointing that out.=C2=A0</div><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">


<br>
Yes, but that was given that a receiver will be capable of receiving any<br=
>
valid number of frames, or samples within a reasonable packetization,<br>
and I think that should be 120 ms.<br>
<br>
&gt; =C2=A0 * We will consider adding an a=3Dmaxptime attribute that repres=
ents the<br>
<div class=3D"">&gt; =C2=A0 =C2=A0 &quot;minimum maxptime&quot; of all the =
offered codecs. That is, if both Opus<br>
&gt; =C2=A0 =C2=A0 (maxptime of 120ms) and PCMU (maxptime of 60) are offere=
d, a<br>
&gt; =C2=A0 =C2=A0 maxptime=3D60 SHOULD be included.<br>
</div>&gt; =C2=A0 =C2=A0 =C2=A0 o Since maxptime spans all codecs, this mea=
ns that some modes<br>
<div class=3D"">&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 (e.g. Opus 120ms) will be =
unavailable, unless only Opus is offered.<br>
<br>
</div>I think you base the assumption on maxptime based on the sending<br>
recommendations, not what is capable of receiving. If one is capable of<br>
receiving 200 ms of a-law and Opus, then you would need to say 120 ms as<br=
>
Opus will limit you.<br></blockquote><div><br></div><div>I agree that Opus&=
#39; maxptime is 120, but should that prevent sending PCMU with 200 ms? Tha=
t is, since sending Opus with 200 ms is not legal, is there any point in fo=
rcing WebRTC impls to indicate this in SDP?</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
&gt; =C2=A0 * We will consider adding an a=3Dptime attribute, set to the re=
ceiver&#39;s<br>
<div class=3D"">&gt; =C2=A0 =C2=A0 preferred frame size to receive. Again, =
this value spans all codecs,<br>
&gt; =C2=A0 =C2=A0 so the specified value may not be viable for all codecs =
(e.g. 30 ms<br>
&gt; =C2=A0 =C2=A0 works for PCMU but not for Opus)<br>
&gt;<br>
&gt; After thinking about this some, I&#39;m not sure the maxptime/ptime va=
lues<br>
&gt; will lead to any different behavior than if they were not specified.<b=
r>
&gt; Since there is a complexity cost (especially if the values need to<br>
&gt; change based on which codecs are offered), is there a clear upside her=
e?<br>
<br>
</div>The maxptime values may in some cases be limited downwards for<br>
interoperability, for example if one like to gateway AMR into a CS<br>
network then the gateway might indicate a maxptime that is less, more<br>
like 20 or 40 ms.<br></blockquote><div><br></div><div>Sure - WebRTC impls n=
eed to respect any maxptime value that they receive, but it&#39;s not clear=
 to me that they need to include one in the SDP they send.=C2=A0</div><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex">


<br>
Please remember that these rules do apply to any codec, not only the MTI<br=
>
ones.<br></blockquote><div><br></div><div>Yes.=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><br></blockquote></div></div></div>

--001a11334c38cc8e4004f4fed6d4--


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Should consent checks be optimized further?
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On 2014-03-18 23:59, Randell Jesup wrote:=0A=
> On 3/12/2014 4:56 PM, Justin Uberti wrote:=0A=
>> Agreed. Aren't there also patent declarations against this doc from=0A=
>> multiple holders?=0A=
>>=0A=
>> While SDP will likely be removed from the API in the future, the=0A=
>> replacement would be a app-specific message sent over websockets,=0A=
>> which seems like it would work just fine.=0A=
>=0A=
> Except that WebSockets doesn't work peer2peer, which means both delay=0A=
> and issues with signaling configurations without a server sitting=0A=
> between the users.=0A=
>=0A=
> However, an app-specific message sent over a DataChannel should work=0A=
> just fine.=0A=
>=0A=
> The downside would be that everything becomes app-specific, which would=
=0A=
> be a problem for any sort of federation or peering setup, and likely=0A=
> gatewaying to legacy.=0A=
=0A=
I'm of the opinion that we should push a lot of signaling like this =0A=
(pause/resume) to RTP/RTCP for those reasons, and in addition to enable =0A=
the UA to save power/transmission for the user.=0A=
=0A=
But perhaps Bernard and Justin are right, this (v1) is not the right time.=
=0A=
=0A=
>=0A=
>=0A=
=0A=


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Subject: Re: [rtcweb] BUNDLE with implicit rtcp-mux
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Please see inline.

This is as an individual!

On 2014-03-20 01:21, Justin Uberti wrote:

> On Mon, Mar 17, 2014 at 6:23 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 

> 
>     (still as individual)
> 
>     Lets be clear what we are discussing here. I don't think this is lot of
>     cases. And if you don't want to implement this in your code generating
>     offer, that is fine. What this describes is that your code handling
>     reception of offers and generating answers must be able to handle the
>     lack of a=rtcp-mux in any offer, rather than just the initial one.
> 
>     I would question if even that assumption holds considering what happens
>     if people start implementing session resumption or session mobility
>     between devices using rehydration style establishment procedures. What
>     one sides consider an initial offer might not be the state the answering
>     party is in. Thus, I am worried about cases that are first offer/answer
>     exchange related.
> 
>     To my understanding you will have code to handle the following cases
>     when offers contains:
> 
>     1) No bundle, no a=rtcp-mux
>     2) No bundle, with a=rtcp-mux
>     3) Bundle, with a=rtcp-mux
> 
>     Considering that you support bundle, and you support usage or not of
>     a=rtcp-mux. Then I don't see how the code handling the SDP logic for
>     Bundle, without a=rtcp-mux is that significant.
> 
> 
> It is significant because you always know you can rtcp-mux. And you
> always know you can BUNDLE when rtcp-muxing. But I think trying to
> BUNDLE a no-rtcp-mux m= line into a rtcp-mux m= line is asking for
> problems. 

To be clear, I am arguing for consistent behavior and usage across the
whole bundle group. Certainly you MUST NOT be allowed to have some m=
line for RTP to use a=rtcp-mux and some other m= line in the same bundle
group to not use it.

What I am considering is the possibility to have a bundle group with
multiple m= RTP lines that do not use a=rtcp-mux and instead have RTCP
over a separate port.

I started arguing that this was important for being able to recover PT
space. But there is actually another reason for this. In cases when you
have QoS, for example in cellular environment, mandating the use of
rtcp-mux forces one to send the RTCP over the expensive QoS bearer. The
amount of RTCP to support adaptation as well as codec control means that
it is still likely that you want to spend at least 5% of the Video
bandwidth on RTCP. If the cost associated with providing QoS for video
can be reduced by 3-5% we in the cellular business definitely like to be
able to utilize it.

I would also like to stress that we are discussing what BUNDLE enables.
RTCWEB is the first user of bundle, but likely not the only one. We have
to be careful to apply all WebRTC context specific limitations on a
general mechanism.

First of all I really like to see that BUNDLE does not restrict ones
choices if one like to multiplex RTP or RTCP when doing BUNDLE. As a
general IETF SDP Signaling mechanism we need this generality.

Secondly, I would strongly prefer that also RTCWEB's specifications do
support this mode of operation.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Subject: Re: [rtcweb] On ptimes and maxptimes
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On 2014-03-20 01:27, Justin Uberti wrote:
> 
> 
> 
> On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     Hi,
> 
>     (As individual)
> 
>     On 2014-03-05 17:26, Justin Uberti wrote:
>     > From the session this morning, I recall this consensus:
>     >
>     >   * We will add recommendations on the frame sizes to use for
>     PCMU/A to
>     >     the rtcweb audio codec I-D (20, 30, 60 ms frames).
> 
>     The above applies to sending payloads
> 
> 
> Right, thanks for pointing that out. 
> 
> 
>     Yes, but that was given that a receiver will be capable of receiving any
>     valid number of frames, or samples within a reasonable packetization,
>     and I think that should be 120 ms.
> 
>     >   * We will consider adding an a=maxptime attribute that
>     represents the
>     >     "minimum maxptime" of all the offered codecs. That is, if both
>     Opus
>     >     (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a
>     >     maxptime=60 SHOULD be included.
>     >       o Since maxptime spans all codecs, this means that some modes
>     >         (e.g. Opus 120ms) will be unavailable, unless only Opus is
>     offered.
> 
>     I think you base the assumption on maxptime based on the sending
>     recommendations, not what is capable of receiving. If one is capable of
>     receiving 200 ms of a-law and Opus, then you would need to say 120 ms as
>     Opus will limit you.
> 
> 
> I agree that Opus' maxptime is 120, but should that prevent sending PCMU
> with 200 ms? That is, since sending Opus with 200 ms is not legal, is
> there any point in forcing WebRTC impls to indicate this in SDP?

What I wished we fixed this issue with maxptime a long time ago. This is
a well known problem that it really has PT scope, but are signalled on
m= line scope.

>  
> 
> 
>     >   * We will consider adding an a=ptime attribute, set to the
>     receiver's
>     >     preferred frame size to receive. Again, this value spans all
>     codecs,
>     >     so the specified value may not be viable for all codecs (e.g.
>     30 ms
>     >     works for PCMU but not for Opus)
>     >
>     > After thinking about this some, I'm not sure the maxptime/ptime values
>     > will lead to any different behavior than if they were not specified.
>     > Since there is a complexity cost (especially if the values need to
>     > change based on which codecs are offered), is there a clear upside
>     here?
> 
>     The maxptime values may in some cases be limited downwards for
>     interoperability, for example if one like to gateway AMR into a CS
>     network then the gateway might indicate a maxptime that is less, more
>     like 20 or 40 ms.
> 
> 
> Sure - WebRTC impls need to respect any maxptime value that they
> receive, but it's not clear to me that they need to include one in the
> SDP they send. 

Ok, but then we are back to my previous argument for writing down what
receiver requirement in regards to each payload type one is required to
support in the audio draft for the ones specified there. But what about
the other codecs that aren't included in the audio draft?

How can a peer know if there are limitations? I wouldn't attempt to
solve the general problem of the scope issue with maxptime. Simply state
that you should include this. It is going to be limited but still
provide information in many cases that is quite useful for the sender.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Justin Uberti'" <juberti@google.com>, "'Bernard Aboba'" <bernard.aboba@gmail.com>
References: <1447FA0C20ED5147A1AA0EF02890A64B1CF8B463@ESESSMB209.ericsson.se> <CAOW+2dsb1GqQmOxf7V6C1Xd_LG12d+kanSm80=kSwmQY=B7GSg@mail.gmail.com> <CAOJ7v-3S1axGPVWB8TF_ALwZ6ExF-D7m3MGsfrkx6EsQNWNpxQ@mail.gmail.com>
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Date: Thu, 20 Mar 2014 13:41:47 +0200
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Subject: Re: [rtcweb] Should we reference the pause/resume I-D?
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Hi,

The pause and resume has the tmmbr=3D0 option based on RFC5104 (No IPR =
in the data base for RFC 5104; =
http://datatracker.ietf.org/ipr/search/?option=3Ddocument_search&id=3Ddra=
ft-ietf-avt-avpf-ccm)  . this works well for point to point case.

Roni

=20

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Justin Uberti
Sent: 12 March, 2014 10:56 PM
To: Bernard Aboba
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Should we reference the pause/resume I-D?

=20

Agreed. Aren't there also patent declarations against this doc from =
multiple holders?

=20

While SDP will likely be removed from the API in the future, the =
replacement would be a app-specific message sent over websockets, which =
seems like it would work just fine.

=20

On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba =
<bernard.aboba@gmail.com> wrote:

While I do like the pause/resume draft, having core RTCWEB WG documents =
(such as RTP Usage) depend on it seems like a bit of a stretch. After =
all, the document was only adopted last week, and it is a rare IETF WG =
document that can go from a -00 WG draft to publication as an RFC in =
under a year.=20

=20

On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=C3=A5kansson LK =
<stefan.lk.hakansson@ericsson.com> wrote:

Hi,

at the IETF last week there was consensus in the AVTEXT WG meeting to
adopt the pause/resume draft [1] as a WG draft.

In rtcweb/webrtc we're have the situation that we're discussing so
called "doo-hickeys" as an API surface where the web app (amongst other
things) can pause and resume the sending of a track. This can
be signaled with the direction attribute and a SDP O/A exchange (and the
app pausing/resuming sending of a track would presumably lead to a
"negotiationneeded" event being fired).

But I think we should in addition require the browser to signal it
according to one of the methods in [1] (e.g. TMMBN =3D 0), and also
understand that signaling (a browser receiving TMMBN =3D 0 must know =
that
the other end-point will pause sending).

My argument is that we know that many dislike SDP in rtcweb, and a
likely development is that it will be removed in a later version. My
speculation is that signaling as outlined in [1] will then be used for
pause/resume. If we support this from the beginning earlier
implementations could more easily interop with those later versions.


Stefan

[1]
https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.txt=


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

=20


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

=20


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style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The pause and resume has the tmmbr=3D0 option based on RFC5104 (No =
IPR in the data base for RFC 5104; =
http://datatracker.ietf.org/ipr/search/?option=3Ddocument_search&amp;id=3D=
draft-ietf-avt-avpf-ccm)=C2=A0 . this works well for point to point =
case.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
rtcweb [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Justin =
Uberti<br><b>Sent:</b> 12 March, 2014 10:56 PM<br><b>To:</b> Bernard =
Aboba<br><b>Cc:</b> rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] =
Should we reference the pause/resume =
I-D?<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>Agreed. =
Aren't there also patent declarations against this doc from multiple =
holders?<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>While SDP will likely be removed from the API in the =
future, the replacement would be a app-specific message sent over =
websockets, which seems like it would work just =
fine.<o:p></o:p></p></div></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba &lt;<a =
href=3D"mailto:bernard.aboba@gmail.com" =
target=3D"_blank">bernard.aboba@gmail.com</a>&gt; =
wrote:<o:p></o:p></p><div><p class=3DMsoNormal>While I do like the =
pause/resume draft, having core RTCWEB WG documents (such as RTP Usage) =
depend on it seems like a bit of a stretch. After all, the document was =
only adopted last week, and it is a rare IETF WG document that can go =
from a -00 WG draft to publication as an RFC in under a year. =
<o:p></o:p></p></div><div><div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Wed, Mar 12, 2014 at 11:01 AM, Stefan =
H=C3=A5kansson LK &lt;<a =
href=3D"mailto:stefan.lk.hakansson@ericsson.com" =
target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal>Hi,<br><br>at the IETF last =
week there was consensus in the AVTEXT WG meeting to<br>adopt the =
pause/resume draft [1] as a WG draft.<br><br>In rtcweb/webrtc we're have =
the situation that we're discussing so<br>called &quot;doo-hickeys&quot; =
as an API surface where the web app (amongst other<br>things) can pause =
and resume the sending of a track. This can<br>be signaled with the =
direction attribute and a SDP O/A exchange (and the<br>app =
pausing/resuming sending of a track would presumably lead to =
a<br>&quot;negotiationneeded&quot; event being fired).<br><br>But I =
think we should in addition require the browser to signal =
it<br>according to one of the methods in [1] (e.g. TMMBN =3D 0), and =
also<br>understand that signaling (a browser receiving TMMBN =3D 0 must =
know that<br>the other end-point will pause sending).<br><br>My argument =
is that we know that many dislike SDP in rtcweb, and a<br>likely =
development is that it will be removed in a later version. =
My<br>speculation is that signaling as outlined in [1] will then be used =
for<br>pause/resume. If we support this from the beginning =
earlier<br>implementations could more easily interop with those later =
versions.<br><br><br>Stefan<br><br>[1]<br><a =
href=3D"https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-paus=
e-05.txt" =
target=3D"_blank">https://tools.ietf.org/id/draft-westerlund-avtext-rtp-s=
tream-pause-05.txt</a><br><br>___________________________________________=
____<br>rtcweb mailing list<br><a href=3D"mailto:rtcweb@ietf.org" =
target=3D"_blank">rtcweb@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></=
o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div><p =
class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><br>______________________________________=
_________<br>rtcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></=
o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On Mar 19, 2014, at 8:21 AM, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> On 03/19/2014 04:08 PM, Dan Wing wrote:
>>=20
>> On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com> =
wrote:
>>=20
>>> I'd like to be silent on the issue, since which IPv6 addresses to =
prefer is likely to be a matter of system policy. Trying to override =
system policy in an application specific profile usually leads to =
sadness.
>>=20
>> The application needs to indicate if it wants a temporary address. If =
the host's policy (or configuration or network) does not support =
temporary addresses, the application won't get a temporary address.  I =
don't see why being silent helps?
>=20
> What API is it using?
>=20
> With a little Googling, the system policy I was thinking of was the =
policy in RFC 6724 ("Default Address Selection for IPv6"), in particular =
section 5 rule 7: "Prefer  temporary addresses".
>=20
> I'm happy to say "it is a good idea for systems to implement the =
recommendations of RFC 6724" (or some more 2119-like language). I =
wouldn't want to claim that if a system has chosen to prefer =
non-temporary addresses, it would have to change its non-conformance to =
RFC 6724 in order to be conformant with RTCWEB specifications.

So perhaps:
   "An RTCWEB implementation SHOULD prefer to use temporary addresses =
[RFC4941] where host and network policy permit [RFC6724]."
?

-d


>=20
>>=20
>> -d
>>=20
>>=20
>>>=20
>>>=20
>>>=20
>>> On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com> wrote:
>>>=20
>>> On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com> =
wrote:
>>>=20
>>>> https://tools.ietf.org/html/rfc4941 defines the concept of =
temporary IPv6 addresses. For, example, as enumerated on my local =
system:
>>>>=20
>>>> inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
>>>> inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf=20
>>>> inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf =
temporary=20
>>>>=20
>>>> As indicated in the RFC, the temporary addresses expire after hours =
or days, and therefore could be used to prevent long-term linkability of =
sessions. Expiration shouldn't be an issue for WebRTC, since we can =
simply do ICE restart if this occurs during a session.
>>>>=20
>>>> Is this something we want to recommend in the transports doc?
>>>=20
>>> Yes.
>>>=20
>>> And it should be as frequent as every new 'call', if IPv6 privacy =
addresses are to provide the same privacy as a NAPT44, as a NAPT44 =
should be assigning random ports per reasons described in RFC6056.
>>>=20
>>> The transport document should also recommend doing port =
randomization (RFC6056), as that was found pretty useful for DNS, but =
randomizing ports is still not popular with many OS's TCP stacks for =
whatever reason.
>>>=20
>>> -d
>>>=20
>>>=20
>>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--Apple-Mail=_62047197-D5B8-4167-8334-CAF775DE4C11
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><br><div><div>On Mar 19, 2014, at 8:21 AM, Harald =
Alvestrand &lt;<a =
href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div bgcolor=3D"#FFFFFF" text=3D"#000000" =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;"><div class=3D"moz-cite-prefix">On =
03/19/2014 04:08 PM, Dan Wing wrote:<br></div><blockquote =
cite=3D"mid:CD747481-EBDA-4FFC-A31D-618E6E217420@cisco.com" =
type=3D"cite"><br><div><div>On Mar 19, 2014, at 12:02 AM, Harald =
Alvestrand &lt;<a moz-do-not-send=3D"true" =
href=3D"mailto:hta@google.com">hta@google.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
dir=3D"ltr">I'd like to be silent on the issue, since which IPv6 =
addresses to prefer is likely to be a matter of system policy. Trying to =
override system policy in an application specific profile usually leads =
to sadness.</div></blockquote><div><br></div><div>The application needs =
to indicate if it wants a temporary address. If the host's policy (or =
configuration or network) does not support temporary addresses, the =
application won't get a temporary address. &nbsp;I don't see why being =
silent helps?</div></div></blockquote><br>What API is it =
using?<br><br>With a little Googling, the system policy I was thinking =
of was the policy in RFC 6724 ("Default Address Selection for IPv6"), in =
particular section 5 rule 7: "Prefer&nbsp; temporary =
addresses".<br><br>I'm happy to say "it is a good idea for systems to =
implement the recommendations of RFC 6724" (or some more 2119-like =
language). I wouldn't want to claim that if a system has chosen to =
prefer non-temporary addresses, it would have to change its =
non-conformance to RFC 6724 in order to be conformant with RTCWEB =
specifications.<br></div></blockquote><div><br></div><div>So =
perhaps:</div><div>&nbsp; &nbsp;"An RTCWEB implementation SHOULD prefer =
to use temporary addresses [RFC4941] where host and network policy =
permit =
[RFC6724]."</div><div>?</div><div><br></div><div>-d</div><div><br></div><b=
r><blockquote type=3D"cite"><div bgcolor=3D"#FFFFFF" text=3D"#000000" =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;"><br><blockquote =
cite=3D"mid:CD747481-EBDA-4FFC-A31D-618E6E217420@cisco.com" =
type=3D"cite"><div><div><br></div><div>-d</div><div><br></div><br><blockqu=
ote type=3D"cite"><div dir=3D"ltr"><br></div><div =
class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, Mar 19, =
2014 at 6:14 AM, Dan Wing<span =
class=3D"Apple-converted-space">&nbsp;</span><span dir=3D"ltr">&lt;<a =
moz-do-not-send=3D"true" href=3D"mailto:dwing@cisco.com" =
target=3D"_blank">dwing@cisco.com</a>&gt;</span><span =
class=3D"Apple-converted-space">&nbsp;</span>wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; =
border-left-width: 1px; border-left-color: rgb(204, 204, 204); =
border-left-style: solid; padding-left: 1ex;"><div style=3D"word-wrap: =
break-word;"><div><div class=3D"h5"><br><div><div>On Mar 18, 2014, at =
6:00 PM, Justin Uberti &lt;<a moz-do-not-send=3D"true" =
href=3D"mailto:juberti@google.com" =
target=3D"_blank">juberti@google.com</a>&gt; wrote:</div><br><blockquote =
type=3D"cite"><div dir=3D"ltr"><a moz-do-not-send=3D"true" =
href=3D"https://tools.ietf.org/html/rfc4941" =
target=3D"_blank">https://tools.ietf.org/html/rfc4941</a><span =
class=3D"Apple-converted-space">&nbsp;</span>defines the concept of =
temporary IPv6 addresses. For, example, as enumerated on my local =
system:<br><div><br></div><div><span>inet 172.31.x.y netmask 0xfffffe00 =
broadcast 172.31.x.255</span><br><span>inet6 =
2620::1008:100b:e2f8:47ff:wwww</span><span>:xxxx prefixlen 64 =
autoconf&nbsp;</span><br><span>inet6 =
2620::1008:100b:819e:1d3f:yyyy</span><span>:zzzz prefixlen 64 =
autoconf<span =
class=3D"Apple-converted-space">&nbsp;</span><b>temporary&nbsp;</b></span>=
<br></div><div><span><br></span></div><div>As indicated in the RFC, the =
temporary addresses expire after hours or days, and therefore could be =
used to prevent long-term linkability of sessions. Expiration shouldn't =
be an issue for WebRTC, since we can simply do ICE restart if this =
occurs during a session.</div><div><br></div><div>Is this something we =
want to recommend in the transports =
doc?<br></div></div></blockquote><br></div></div></div><div>Yes.</div><div=
><br></div><div>And it should be as frequent as every new 'call', if =
IPv6 privacy addresses are to provide the same privacy as a NAPT44, as a =
NAPT44 should be assigning random ports per reasons described in =
RFC6056.</div><div><br></div><div>The transport document should also =
recommend doing port randomization (RFC6056), as that was found pretty =
useful for DNS, but randomizing ports is still not popular with many =
OS's TCP stacks for whatever reason.</div><span class=3D"HOEnZb"><font =
color=3D"#888888"><div><br></div><div>-d</div><div><br></div><br></font></=
span></div></blockquote></div><br></div></blockquote></div><br><br><fields=
et class=3D"mimeAttachmentHeader"></fieldset><br><pre =
wrap=3D"">_______________________________________________
rtcweb mailing list
<a class=3D"moz-txt-link-abbreviated" =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class=3D"moz-txt-link-freetext" =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a>
=
</pre></blockquote><br>_______________________________________________<br>=
rtcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><br></div></blockquote></div><br></body></html=
>=

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From: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On Mar 19, 2014, at 8:39 AM, Simon Perreault =
<simon.perreault@viagenie.ca> wrote:

> Le 2014-03-19 03:02, Harald Alvestrand a =E9crit :
>> I'd like to be silent on the issue, since which IPv6 addresses to =
prefer
>> is likely to be a matter of system policy. Trying to override system
>> policy in an application specific profile usually leads to sadness.
>=20
> What would be useful to mention IMHO is that ICE restart should be =
used
> to gracefully handle the issue of IP address expiry, of which =
temporary
> IPv6 addresses are a frequently-encountered example.

When a temporary address expires, existing sockets bound that address =
should continue working with that expired address -- have you found =
otherwise on any OS, such that we would need ICE Mobility for temporary =
address expiration?

-d



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Subject: [rtcweb] Draft minutes for Tuesday RTCWEB session in London
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WG,

Here is the initial version of the minutes for Tuesdays (2014-03-04)
RTCWEB WG session in London. It is available here:

http://www.ietf.org/proceedings/89/minutes/minutes-89-rtcweb

Please send any corrections to the mailing list.

Wednesday's minutes are forthcoming.

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Thu Mar 20 08:52:02 2014
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Date: Thu, 20 Mar 2014 11:51:43 -0400
From: Simon Perreault <simon.perreault@viagenie.ca>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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Le 2014-03-20 10:54, Dan Wing a écrit :
> 
> On Mar 19, 2014, at 8:39 AM, Simon Perreault <simon.perreault@viagenie.ca> wrote:
> 
>> Le 2014-03-19 03:02, Harald Alvestrand a écrit :
>>> I'd like to be silent on the issue, since which IPv6 addresses to prefer
>>> is likely to be a matter of system policy. Trying to override system
>>> policy in an application specific profile usually leads to sadness.
>>
>> What would be useful to mention IMHO is that ICE restart should be used
>> to gracefully handle the issue of IP address expiry, of which temporary
>> IPv6 addresses are a frequently-encountered example.
> 
> When a temporary address expires, existing sockets bound that address should continue working with that expired address -- have you found otherwise on any OS, such that we would need ICE Mobility for temporary address expiration?

When the address is removed from the interface, it doesn't matter if the
socket is still "good" or not. The address is no longer on the interface
so you will not receive packets sent to that address.

I'm looking at the Linux and OpenBSD IPv6 stacks right now and neither
seems to care about existing sockets before expiring an address. Haven't
tried it though.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
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From: Justin Uberti <juberti@google.com>
Date: Thu, 20 Mar 2014 09:30:08 -0700
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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--089e0129421884a16104f50c499c
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On Thu, Mar 20, 2014 at 8:51 AM, Simon Perreault <
simon.perreault@viagenie.ca> wrote:

> Le 2014-03-20 10:54, Dan Wing a =C3=A9crit :
> >
> > On Mar 19, 2014, at 8:39 AM, Simon Perreault <
> simon.perreault@viagenie.ca> wrote:
> >
> >> Le 2014-03-19 03:02, Harald Alvestrand a =C3=A9crit :
> >>> I'd like to be silent on the issue, since which IPv6 addresses to
> prefer
> >>> is likely to be a matter of system policy. Trying to override system
> >>> policy in an application specific profile usually leads to sadness.
> >>
> >> What would be useful to mention IMHO is that ICE restart should be use=
d
> >> to gracefully handle the issue of IP address expiry, of which temporar=
y
> >> IPv6 addresses are a frequently-encountered example.
> >
> > When a temporary address expires, existing sockets bound that address
> should continue working with that expired address -- have you found
> otherwise on any OS, such that we would need ICE Mobility for temporary
> address expiration?
>
> When the address is removed from the interface, it doesn't matter if the
> socket is still "good" or not. The address is no longer on the interface
> so you will not receive packets sent to that address.
>
> I'm looking at the Linux and OpenBSD IPv6 stacks right now and neither
> seems to care about existing sockets before expiring an address. Haven't
> tried it though.
>
>
On Mac, the temporary addresses seem to change every day, but any existing
addresses seem to remain valid for a week.

--089e0129421884a16104f50c499c
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Mar 20, 2014 at 8:51 AM, Simon Perreault <span dir=3D"ltr">=
&lt;<a href=3D"mailto:simon.perreault@viagenie.ca" target=3D"_blank">simon.=
perreault@viagenie.ca</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Le 2014-03-20 10:54, Dan Wing a =C3=A9crit :=
<br>
<div class=3D"">&gt;<br>
&gt; On Mar 19, 2014, at 8:39 AM, Simon Perreault &lt;<a href=3D"mailto:sim=
on.perreault@viagenie.ca">simon.perreault@viagenie.ca</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; Le 2014-03-19 03:02, Harald Alvestrand a =C3=A9crit :<br>
&gt;&gt;&gt; I&#39;d like to be silent on the issue, since which IPv6 addre=
sses to prefer<br>
&gt;&gt;&gt; is likely to be a matter of system policy. Trying to override =
system<br>
&gt;&gt;&gt; policy in an application specific profile usually leads to sad=
ness.<br>
&gt;&gt;<br>
&gt;&gt; What would be useful to mention IMHO is that ICE restart should be=
 used<br>
&gt;&gt; to gracefully handle the issue of IP address expiry, of which temp=
orary<br>
&gt;&gt; IPv6 addresses are a frequently-encountered example.<br>
&gt;<br>
&gt; When a temporary address expires, existing sockets bound that address =
should continue working with that expired address -- have you found otherwi=
se on any OS, such that we would need ICE Mobility for temporary address ex=
piration?<br>


<br>
</div>When the address is removed from the interface, it doesn&#39;t matter=
 if the<br>
socket is still &quot;good&quot; or not. The address is no longer on the in=
terface<br>
so you will not receive packets sent to that address.<br>
<br>
I&#39;m looking at the Linux and OpenBSD IPv6 stacks right now and neither<=
br>
seems to care about existing sockets before expiring an address. Haven&#39;=
t<br>
tried it though.<br>
<div class=3D"im HOEnZb"><br></div></blockquote><div><br></div><div>On Mac,=
 the temporary addresses seem to change every day, but any existing address=
es seem to remain valid for a week.=C2=A0</div></div></div></div>

--089e0129421884a16104f50c499c--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 20 Mar 2014 09:34:35 -0700
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On Thu, Mar 20, 2014 at 7:52 AM, Dan Wing <dwing@cisco.com> wrote:

>
> On Mar 19, 2014, at 8:21 AM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
> On 03/19/2014 04:08 PM, Dan Wing wrote:
>
>
> On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com> wrote:
>
> I'd like to be silent on the issue, since which IPv6 addresses to prefer
> is likely to be a matter of system policy. Trying to override system policy
> in an application specific profile usually leads to sadness.
>
>
> The application needs to indicate if it wants a temporary address. If the
> host's policy (or configuration or network) does not support temporary
> addresses, the application won't get a temporary address.  I don't see why
> being silent helps?
>
>
> What API is it using?
>
> With a little Googling, the system policy I was thinking of was the policy
> in RFC 6724 ("Default Address Selection for IPv6"), in particular section 5
> rule 7: "Prefer  temporary addresses".
>
> I'm happy to say "it is a good idea for systems to implement the
> recommendations of RFC 6724" (or some more 2119-like language). I wouldn't
> want to claim that if a system has chosen to prefer non-temporary
> addresses, it would have to change its non-conformance to RFC 6724 in order
> to be conformant with RTCWEB specifications.
>
>
> So perhaps:
>    "An RTCWEB implementation SHOULD prefer to use temporary addresses
> [RFC4941] where host and network policy permit [RFC6724]."
> ?
>

I think it needs to be stronger than that - something like
"where host and network policy permit, RTCWEB implementations SHOULD gather
IPv6 temporary addresses and SHOULD NOT gather non-temporary addresses".

Preferring to use temporary addresses is probably not sufficient to prevent
linkage, since you will have connectivity checks from the non-temporary
addresses. (i.e. an eavesdropper listening over an extended period of time
could determine calls are from the same endpoint)

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Mar 20, 2014 at 7:52 AM, Dan Wing <span dir=3D"ltr">&lt;<a =
href=3D"mailto:dwing@cisco.com" target=3D"_blank">dwing@cisco.com</a>&gt;</=
span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word"><br><div=
><div class=3D""><div>On Mar 19, 2014, at 8:21 AM, Harald Alvestrand &lt;<a=
 href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.n=
o</a>&gt; wrote:</div>

<br><blockquote type=3D"cite"><div bgcolor=3D"#FFFFFF" text=3D"#000000" sty=
le=3D"font-family:Helvetica;font-size:12px;font-style:normal;font-variant:n=
ormal;font-weight:normal;letter-spacing:normal;line-height:normal;text-alig=
n:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing=
:0px">

<div>On 03/19/2014 04:08 PM, Dan Wing wrote:<br></div><blockquote type=3D"c=
ite"><br><div><div>On Mar 19, 2014, at 12:02 AM, Harald Alvestrand &lt;<a h=
ref=3D"mailto:hta@google.com" target=3D"_blank">hta@google.com</a>&gt; wrot=
e:</div>

<br><blockquote type=3D"cite"><div dir=3D"ltr">I&#39;d like to be silent on=
 the issue, since which IPv6 addresses to prefer is likely to be a matter o=
f system policy. Trying to override system policy in an application specifi=
c profile usually leads to sadness.</div>

</blockquote><div><br></div><div>The application needs to indicate if it wa=
nts a temporary address. If the host&#39;s policy (or configuration or netw=
ork) does not support temporary addresses, the application won&#39;t get a =
temporary address. =C2=A0I don&#39;t see why being silent helps?</div>

</div></blockquote><br>What API is it using?<br><br>With a little Googling,=
 the system policy I was thinking of was the policy in RFC 6724 (&quot;Defa=
ult Address Selection for IPv6&quot;), in particular section 5 rule 7: &quo=
t;Prefer=C2=A0 temporary addresses&quot;.<br>

<br>I&#39;m happy to say &quot;it is a good idea for systems to implement t=
he recommendations of RFC 6724&quot; (or some more 2119-like language). I w=
ouldn&#39;t want to claim that if a system has chosen to prefer non-tempora=
ry addresses, it would have to change its non-conformance to RFC 6724 in or=
der to be conformant with RTCWEB specifications.<br>

</div></blockquote><div><br></div></div><div>So perhaps:</div><div>=C2=A0 =
=C2=A0&quot;An RTCWEB implementation SHOULD prefer to use temporary address=
es [RFC4941] where host and network policy permit [RFC6724].&quot;</div><di=
v>?</div>

</div></div></blockquote><div><br></div><div>I think it needs to be stronge=
r than that - something like=C2=A0</div><div>&quot;where host and network p=
olicy permit, RTCWEB implementations SHOULD gather IPv6 temporary addresses=
 and SHOULD NOT gather non-temporary addresses&quot;.</div>

<div><br></div><div>Preferring to use temporary addresses is probably not s=
ufficient to prevent linkage, since you will have connectivity checks from =
the non-temporary addresses. (i.e. an eavesdropper listening over an extend=
ed period of time could determine calls are from the same endpoint)</div>

</div><br></div></div>

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From: Justin Uberti <juberti@google.com>
Date: Thu, 20 Mar 2014 09:39:48 -0700
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Subject: Re: [rtcweb] On ptimes and maxptimes
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On Thu, Mar 20, 2014 at 3:53 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2014-03-20 01:27, Justin Uberti wrote:
> >
> >
> >
> > On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund
> > <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> > wrote:
> >
> >     Hi,
> >
> >     (As individual)
> >
> >     On 2014-03-05 17:26, Justin Uberti wrote:
> >     > From the session this morning, I recall this consensus:
> >     >
> >     >   * We will add recommendations on the frame sizes to use for
> >     PCMU/A to
> >     >     the rtcweb audio codec I-D (20, 30, 60 ms frames).
> >
> >     The above applies to sending payloads
> >
> >
> > Right, thanks for pointing that out.
> >
> >
> >     Yes, but that was given that a receiver will be capable of receiving
> any
> >     valid number of frames, or samples within a reasonable packetization,
> >     and I think that should be 120 ms.
> >
> >     >   * We will consider adding an a=maxptime attribute that
> >     represents the
> >     >     "minimum maxptime" of all the offered codecs. That is, if both
> >     Opus
> >     >     (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a
> >     >     maxptime=60 SHOULD be included.
> >     >       o Since maxptime spans all codecs, this means that some modes
> >     >         (e.g. Opus 120ms) will be unavailable, unless only Opus is
> >     offered.
> >
> >     I think you base the assumption on maxptime based on the sending
> >     recommendations, not what is capable of receiving. If one is capable
> of
> >     receiving 200 ms of a-law and Opus, then you would need to say 120
> ms as
> >     Opus will limit you.
> >
> >
> > I agree that Opus' maxptime is 120, but should that prevent sending PCMU
> > with 200 ms? That is, since sending Opus with 200 ms is not legal, is
> > there any point in forcing WebRTC impls to indicate this in SDP?
>
> What I wished we fixed this issue with maxptime a long time ago. This is
> a well known problem that it really has PT scope, but are signalled on
> m= line scope.
>
> >
> >
> >
> >     >   * We will consider adding an a=ptime attribute, set to the
> >     receiver's
> >     >     preferred frame size to receive. Again, this value spans all
> >     codecs,
> >     >     so the specified value may not be viable for all codecs (e.g.
> >     30 ms
> >     >     works for PCMU but not for Opus)
> >     >
> >     > After thinking about this some, I'm not sure the maxptime/ptime
> values
> >     > will lead to any different behavior than if they were not
> specified.
> >     > Since there is a complexity cost (especially if the values need to
> >     > change based on which codecs are offered), is there a clear upside
> >     here?
> >
> >     The maxptime values may in some cases be limited downwards for
> >     interoperability, for example if one like to gateway AMR into a CS
> >     network then the gateway might indicate a maxptime that is less, more
> >     like 20 or 40 ms.
> >
> >
> > Sure - WebRTC impls need to respect any maxptime value that they
> > receive, but it's not clear to me that they need to include one in the
> > SDP they send.
>
> Ok, but then we are back to my previous argument for writing down what
> receiver requirement in regards to each payload type one is required to
> support in the audio draft for the ones specified there. But what about
> the other codecs that aren't included in the audio draft?
>
> How can a peer know if there are limitations? I wouldn't attempt to
> solve the general problem of the scope issue with maxptime. Simply state
> that you should include this. It is going to be limited but still
> provide information in many cases that is quite useful for the sender.
>
>
Just to be sure I understand, your position is that WebRTC impls SHOULD
include an a=maxptime attribute indicating the "minimum maximum" frame size
they can receive, across all codecs?

Do you think anything needs to be included regarding a=ptime?

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Mar 20, 2014 at 3:53 AM, Magnus Westerlund <span dir=3D"ltr=
">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 2014-03-20 01:27, Justin =
Uberti wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund<br>
</div>&gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.wes=
terlund@ericsson.com</a> &lt;mailto:<a href=3D"mailto:magnus.westerlund@eri=
csson.com">magnus.westerlund@ericsson.com</a>&gt;&gt;<br>
<div><div class=3D"h5">&gt; wrote:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Hi,<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 (As individual)<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 On 2014-03-05 17:26, Justin Uberti wrote:<br>
&gt; =C2=A0 =C2=A0 &gt; From the session this morning, I recall this consen=
sus:<br>
&gt; =C2=A0 =C2=A0 &gt;<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 * We will add recommendations on the frame s=
izes to use for<br>
&gt; =C2=A0 =C2=A0 PCMU/A to<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 the rtcweb audio codec I-D (20, 30, 6=
0 ms frames).<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 The above applies to sending payloads<br>
&gt;<br>
&gt;<br>
&gt; Right, thanks for pointing that out.<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 Yes, but that was given that a receiver will be capable =
of receiving any<br>
&gt; =C2=A0 =C2=A0 valid number of frames, or samples within a reasonable p=
acketization,<br>
&gt; =C2=A0 =C2=A0 and I think that should be 120 ms.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 * We will consider adding an a=3Dmaxptime at=
tribute that<br>
&gt; =C2=A0 =C2=A0 represents the<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 &quot;minimum maxptime&quot; of all t=
he offered codecs. That is, if both<br>
&gt; =C2=A0 =C2=A0 Opus<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 (maxptime of 120ms) and PCMU (maxptim=
e of 60) are offered, a<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 maxptime=3D60 SHOULD be included.<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 =C2=A0 o Since maxptime spans all cod=
ecs, this means that some modes<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 (e.g. Opus 120ms) will =
be unavailable, unless only Opus is<br>
&gt; =C2=A0 =C2=A0 offered.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 I think you base the assumption on maxptime based on the=
 sending<br>
&gt; =C2=A0 =C2=A0 recommendations, not what is capable of receiving. If on=
e is capable of<br>
&gt; =C2=A0 =C2=A0 receiving 200 ms of a-law and Opus, then you would need =
to say 120 ms as<br>
&gt; =C2=A0 =C2=A0 Opus will limit you.<br>
&gt;<br>
&gt;<br>
&gt; I agree that Opus&#39; maxptime is 120, but should that prevent sendin=
g PCMU<br>
&gt; with 200 ms? That is, since sending Opus with 200 ms is not legal, is<=
br>
&gt; there any point in forcing WebRTC impls to indicate this in SDP?<br>
<br>
</div></div>What I wished we fixed this issue with maxptime a long time ago=
. This is<br>
a well known problem that it really has PT scope, but are signalled on<br>
m=3D line scope.<br>
<div class=3D""><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 * We will consider adding an a=3Dptime attri=
bute, set to the<br>
&gt; =C2=A0 =C2=A0 receiver&#39;s<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 preferred frame size to receive. Agai=
n, this value spans all<br>
&gt; =C2=A0 =C2=A0 codecs,<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 so the specified value may not be via=
ble for all codecs (e.g.<br>
&gt; =C2=A0 =C2=A0 30 ms<br>
&gt; =C2=A0 =C2=A0 &gt; =C2=A0 =C2=A0 works for PCMU but not for Opus)<br>
&gt; =C2=A0 =C2=A0 &gt;<br>
&gt; =C2=A0 =C2=A0 &gt; After thinking about this some, I&#39;m not sure th=
e maxptime/ptime values<br>
&gt; =C2=A0 =C2=A0 &gt; will lead to any different behavior than if they we=
re not specified.<br>
&gt; =C2=A0 =C2=A0 &gt; Since there is a complexity cost (especially if the=
 values need to<br>
&gt; =C2=A0 =C2=A0 &gt; change based on which codecs are offered), is there=
 a clear upside<br>
&gt; =C2=A0 =C2=A0 here?<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 The maxptime values may in some cases be limited downwar=
ds for<br>
&gt; =C2=A0 =C2=A0 interoperability, for example if one like to gateway AMR=
 into a CS<br>
&gt; =C2=A0 =C2=A0 network then the gateway might indicate a maxptime that =
is less, more<br>
&gt; =C2=A0 =C2=A0 like 20 or 40 ms.<br>
&gt;<br>
&gt;<br>
&gt; Sure - WebRTC impls need to respect any maxptime value that they<br>
&gt; receive, but it&#39;s not clear to me that they need to include one in=
 the<br>
&gt; SDP they send.<br>
<br>
</div>Ok, but then we are back to my previous argument for writing down wha=
t<br>
receiver requirement in regards to each payload type one is required to<br>
support in the audio draft for the ones specified there. But what about<br>
the other codecs that aren&#39;t included in the audio draft?<br>
<br>
How can a peer know if there are limitations? I wouldn&#39;t attempt to<br>
solve the general problem of the scope issue with maxptime. Simply state<br=
>
that you should include this. It is going to be limited but still<br>
provide information in many cases that is quite useful for the sender.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br></div></div></blockquote><div><=
br></div><div>Just to be sure I understand, your position is that WebRTC im=
pls SHOULD include an a=3Dmaxptime attribute indicating the &quot;minimum m=
aximum&quot; frame size they can receive, across all codecs?</div>

<div><br></div><div>Do you think anything needs to be included regarding a=
=3Dptime?</div></div></div></div>

--20cf302d497a1b169504f50c6c87--


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Le 2014-03-20 12:30, Justin Uberti a Ã©crit :
> On Mac, the temporary addresses seem to change every day, but any
> existing addresses seem to remain valid for a week. 

Correct. Addresses go from "preferred and valid" (one day) to "valid"
(one week) to "invalid". The address is removed from the interface when
it becomes invalid. So in theory you would need to use WebRTC for a
whole week before an temporary address is yanked from under your feet.

Simon
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On Mar 20, 2014, at 9:34 AM, Justin Uberti <juberti@google.com> wrote:

>=20
>=20
>=20
> On Thu, Mar 20, 2014 at 7:52 AM, Dan Wing <dwing@cisco.com> wrote:
>=20
> On Mar 19, 2014, at 8:21 AM, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>=20
>> On 03/19/2014 04:08 PM, Dan Wing wrote:
>>>=20
>>> On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com> =
wrote:
>>>=20
>>>> I'd like to be silent on the issue, since which IPv6 addresses to =
prefer is likely to be a matter of system policy. Trying to override =
system policy in an application specific profile usually leads to =
sadness.
>>>=20
>>> The application needs to indicate if it wants a temporary address. =
If the host's policy (or configuration or network) does not support =
temporary addresses, the application won't get a temporary address.  I =
don't see why being silent helps?
>>=20
>> What API is it using?
>>=20
>> With a little Googling, the system policy I was thinking of was the =
policy in RFC 6724 ("Default Address Selection for IPv6"), in particular =
section 5 rule 7: "Prefer  temporary addresses".
>>=20
>> I'm happy to say "it is a good idea for systems to implement the =
recommendations of RFC 6724" (or some more 2119-like language). I =
wouldn't want to claim that if a system has chosen to prefer =
non-temporary addresses, it would have to change its non-conformance to =
RFC 6724 in order to be conformant with RTCWEB specifications.
>=20
> So perhaps:
>    "An RTCWEB implementation SHOULD prefer to use temporary addresses =
[RFC4941] where host and network policy permit [RFC6724]."
> ?
>=20
> I think it needs to be stronger than that - something like=20
> "where host and network policy permit, RTCWEB implementations SHOULD =
gather IPv6 temporary addresses and SHOULD NOT gather non-temporary =
addresses".
>=20
> Preferring to use temporary addresses is probably not sufficient to =
prevent linkage, since you will have connectivity checks from the =
non-temporary addresses. (i.e. an eavesdropper listening over an =
extended period of time could determine calls are from the same =
endpoint)

Agreed.  I like your suggested wording.

-d


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<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><br><div><div>On Mar 20, 2014, at 9:34 AM, Justin Uberti &lt;<a href="mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><meta http-equiv="Content-Type" content="text/html; charset=utf-8"><div dir="ltr"><br><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Mar 20, 2014 at 7:52 AM, Dan Wing <span dir="ltr">&lt;<a href="mailto:dwing@cisco.com" target="_blank">dwing@cisco.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word"><br><div><div class=""><div>On Mar 19, 2014, at 8:21 AM, Harald Alvestrand &lt;<a href="mailto:harald@alvestrand.no" target="_blank">harald@alvestrand.no</a>&gt; wrote:</div>

<br><blockquote type="cite"><div bgcolor="#FFFFFF" text="#000000" style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">

<div>On 03/19/2014 04:08 PM, Dan Wing wrote:<br></div><blockquote type="cite"><br><div><div>On Mar 19, 2014, at 12:02 AM, Harald Alvestrand &lt;<a href="mailto:hta@google.com" target="_blank">hta@google.com</a>&gt; wrote:</div>

<br><blockquote type="cite"><div dir="ltr">I'd like to be silent on the issue, since which IPv6 addresses to prefer is likely to be a matter of system policy. Trying to override system policy in an application specific profile usually leads to sadness.</div>

</blockquote><div><br></div><div>The application needs to indicate if it wants a temporary address. If the host's policy (or configuration or network) does not support temporary addresses, the application won't get a temporary address. &nbsp;I don't see why being silent helps?</div>

</div></blockquote><br>What API is it using?<br><br>With a little Googling, the system policy I was thinking of was the policy in RFC 6724 ("Default Address Selection for IPv6"), in particular section 5 rule 7: "Prefer&nbsp; temporary addresses".<br>

<br>I'm happy to say "it is a good idea for systems to implement the recommendations of RFC 6724" (or some more 2119-like language). I wouldn't want to claim that if a system has chosen to prefer non-temporary addresses, it would have to change its non-conformance to RFC 6724 in order to be conformant with RTCWEB specifications.<br>

</div></blockquote><div><br></div></div><div>So perhaps:</div><div>&nbsp; &nbsp;"An RTCWEB implementation SHOULD prefer to use temporary addresses [RFC4941] where host and network policy permit [RFC6724]."</div><div>?</div>

</div></div></blockquote><div><br></div><div>I think it needs to be stronger than that - something like&nbsp;</div><div>"where host and network policy permit, RTCWEB implementations SHOULD gather IPv6 temporary addresses and SHOULD NOT gather non-temporary addresses".</div>

<div><br></div><div>Preferring to use temporary addresses is probably not sufficient to prevent linkage, since you will have connectivity checks from the non-temporary addresses. (i.e. an eavesdropper listening over an extended period of time could determine calls are from the same endpoint)</div></div></div></div>
</blockquote></div><br><div>Agreed. &nbsp;I like your suggested wording.</div><div><br></div><div>-d</div><div><br></div></body></html>
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(Please forgive the crosspost, and reply only to the rai list):
Please see
<http://mailarchive.ietf.org/arch/msg/rai/llvQvf7FWMRM-3ZV0R1aDE5wpRg>


From nobody Thu Mar 20 13:09:19 2014
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The RTCWEB working group will hold an Interim meeting with The W3c 
WEBRTC and Media Capture groups May 19-21, 2014 in Washington DC. The 
location will be :

1101 New York Ave NW, Washington, DC 20005

Details on remote participation and registration will be sent to the 
RTCWEB list (https://www.ietf.org/mailman/listinfo/rtcweb) in the next 
few weeks.


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Roni Even <ron.even.tlv@gmail.com>, "'Justin Uberti'" <juberti@google.com>, "'Bernard Aboba'" <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Should we reference the pause/resume I-D?
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Subject: Re: [rtcweb] Should we reference the pause/resume I-D?
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Caveats to be aware of with the tmmbr=3D0 approach include:

  *   tmmbr=3D0 implies pause, but it is indistinguishable for tmmbr being =
set to 0 for other reasons
  *   knowledge of an appropriate value to set for resume is not always ava=
ilable at the RTCP layer (implementation dependent)
  *   theoretically not appropriate to jump directly back to tmmbr=3D<value=
 prior to pause> even when known

Despite these caveats, I agree with Roni that using tmmbr=3D0 for pause has=
 worked relatively well in enterprise SIP deployments.

Cheers,
Charles


From: Roni Even <ron.even.tlv@gmail.com<mailto:ron.even.tlv@gmail.com>>
Date: Thursday, March 20, 2014 at 4:41 AM
To: Justin Uberti <juberti@google.com<mailto:juberti@google.com>>, 'Bernard=
 Aboba' <bernard.aboba@gmail.com<mailto:bernard.aboba@gmail.com>>
Cc: "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [rtcweb] Should we reference the pause/resume I-D?

Hi,
The pause and resume has the tmmbr=3D0 option based on RFC5104 (No IPR in t=
he data base for RFC 5104; http://datatracker.ietf.org/ipr/search/?option=
=3Ddocument_search&id=3Ddraft-ietf-avt-avpf-ccm)  . this works well for poi=
nt to point case.
Roni

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Justin Uberti
Sent: 12 March, 2014 10:56 PM
To: Bernard Aboba
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Should we reference the pause/resume I-D?

Agreed. Aren't there also patent declarations against this doc from multipl=
e holders?

While SDP will likely be removed from the API in the future, the replacemen=
t would be a app-specific message sent over websockets, which seems like it=
 would work just fine.

On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba <bernard.aboba@gmail.com<ma=
ilto:bernard.aboba@gmail.com>> wrote:
While I do like the pause/resume draft, having core RTCWEB WG documents (su=
ch as RTP Usage) depend on it seems like a bit of a stretch. After all, the=
 document was only adopted last week, and it is a rare IETF WG document tha=
t can go from a -00 WG draft to publication as an RFC in under a year.

On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=E5kansson LK <stefan.lk.hakansso=
n@ericsson.com<mailto:stefan.lk.hakansson@ericsson.com>> wrote:
Hi,

at the IETF last week there was consensus in the AVTEXT WG meeting to
adopt the pause/resume draft [1] as a WG draft.

In rtcweb/webrtc we're have the situation that we're discussing so
called "doo-hickeys" as an API surface where the web app (amongst other
things) can pause and resume the sending of a track. This can
be signaled with the direction attribute and a SDP O/A exchange (and the
app pausing/resuming sending of a track would presumably lead to a
"negotiationneeded" event being fired).

But I think we should in addition require the browser to signal it
according to one of the methods in [1] (e.g. TMMBN =3D 0), and also
understand that signaling (a browser receiving TMMBN =3D 0 must know that
the other end-point will pause sending).

My argument is that we know that many dislike SDP in rtcweb, and a
likely development is that it will be removed in a later version. My
speculation is that signaling as outlined in [1] will then be used for
pause/resume. If we support this from the beginning earlier
implementations could more easily interop with those later versions.


Stefan

[1]
https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pause-05.txt

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Caveats to be aware of with the tmmbr=3D0 approach include:</div>
<ul>
<li>tmmbr=3D0 implies pause, but it is indistinguishable for tmmbr being se=
t to 0 for other reasons</li><li>knowledge of an appropriate value to set f=
or resume is not always available at the RTCP layer (implementation depende=
nt)</li><li>theoretically not appropriate to jump directly back to tmmbr=3D=
&lt;value prior to pause&gt; even when known</li></ul>
<div>Despite these caveats, I agree with Roni that using tmmbr=3D0 for paus=
e has worked relatively well in enterprise SIP deployments.</div>
<div><br>
</div>
<div>Cheers,</div>
<div>Charles</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
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 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Roni Even &lt;<a href=3D"mail=
to:ron.even.tlv@gmail.com">ron.even.tlv@gmail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, March 20, 2014 at 4=
:41 AM<br>
<span style=3D"font-weight:bold">To: </span>Justin Uberti &lt;<a href=3D"ma=
ilto:juberti@google.com">juberti@google.com</a>&gt;, 'Bernard Aboba' &lt;<a=
 href=3D"mailto:bernard.aboba@gmail.com">bernard.aboba@gmail.com</a>&gt;<br=
>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a href=3D"mailto:rtcweb@=
ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">=
rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [rtcweb] Should we ref=
erence the pause/resume I-D?<br>
</div>
<div><br>
</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-micro=
soft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" x=
mlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:/=
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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125);">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125);">The pause and resume has the tmmbr=
=3D0 option based on RFC5104 (No IPR in the data base for RFC 5104;
<a href=3D"http://datatracker.ietf.org/ipr/search/?option=3Ddocument_search=
&amp;id=3Ddraft-ietf-avt-avpf-ccm">
http://datatracker.ietf.org/ipr/search/?option=3Ddocument_search&amp;id=3Dd=
raft-ietf-avt-avpf-ccm</a>)&nbsp; . this works well for point to point case=
.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125);">Roni<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: Calibri=
, sans-serif; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size: 10pt; font-family: Taho=
ma, sans-serif;">From:</span></b><span style=3D"font-size: 10pt; font-famil=
y: Tahoma, sans-serif;"> rtcweb [<a href=3D"mailto:rtcweb-bounces@ietf.org"=
>mailto:rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Justin Uberti<br>
<b>Sent:</b> 12 March, 2014 10:56 PM<br>
<b>To:</b> Bernard Aboba<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] Should we reference the pause/resume I-D?<o:p>=
</o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Agreed. Aren't there also patent declarations agains=
t this doc from multiple holders?<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">While SDP will likely be removed from the API in the=
 future, the replacement would be a app-specific message sent over websocke=
ts, which seems like it would work just fine.<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Wed, Mar 12, 2014 at 12:50 PM, Bernard Aboba &lt;=
<a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@=
gmail.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">While I do like the pause/resume draft, having core =
RTCWEB WG documents (such as RTP Usage) depend on it seems like a bit of a =
stretch. After all, the document was only adopted last week, and it is a ra=
re IETF WG document that can go from
 a -00 WG draft to publication as an RFC in under a year. <o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Wed, Mar 12, 2014 at 11:01 AM, Stefan H=E5kansson=
 LK &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blan=
k">stefan.lk.hakansson@ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">Hi,<br>
<br>
at the IETF last week there was consensus in the AVTEXT WG meeting to<br>
adopt the pause/resume draft [1] as a WG draft.<br>
<br>
In rtcweb/webrtc we're have the situation that we're discussing so<br>
called &quot;doo-hickeys&quot; as an API surface where the web app (amongst=
 other<br>
things) can pause and resume the sending of a track. This can<br>
be signaled with the direction attribute and a SDP O/A exchange (and the<br=
>
app pausing/resuming sending of a track would presumably lead to a<br>
&quot;negotiationneeded&quot; event being fired).<br>
<br>
But I think we should in addition require the browser to signal it<br>
according to one of the methods in [1] (e.g. TMMBN =3D 0), and also<br>
understand that signaling (a browser receiving TMMBN =3D 0 must know that<b=
r>
the other end-point will pause sending).<br>
<br>
My argument is that we know that many dislike SDP in rtcweb, and a<br>
likely development is that it will be removed in a later version. My<br>
speculation is that signaling as outlined in [1] will then be used for<br>
pause/resume. If we support this from the beginning earlier<br>
implementations could more easily interop with those later versions.<br>
<br>
<br>
Stefan<br>
<br>
[1]<br>
<a href=3D"https://tools.ietf.org/id/draft-westerlund-avtext-rtp-stream-pau=
se-05.txt" target=3D"_blank">https://tools.ietf.org/id/draft-westerlund-avt=
ext-rtp-stream-pause-05.txt</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</body>
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Date: Thu, 20 Mar 2014 13:56:04 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
To: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On Thu, Mar 20, 2014 at 10:07 AM, Dan Wing <dwing@cisco.com> wrote:

>
> On Mar 20, 2014, at 9:34 AM, Justin Uberti <juberti@google.com> wrote:
>
>
>
>
>
> So perhaps:
>>    "An RTCWEB implementation SHOULD prefer to use temporary addresses
>> [RFC4941] where host and network policy permit [RFC6724]."
>> ?
>>
>
> I think it needs to be stronger than that - something like
> "where
> 
> host and network policy permit, RTCWEB implementations SHOULD gather IPv6
> temporary addresses and SHOULD NOT gather non-temporary addresses".
>
> Preferring to use temporary addresses is probably not sufficient to
> prevent linkage, since you will have connectivity checks from the
> non-temporary addresses. (i.e. an eavesdropper listening over an extended
> period of time could determine calls are from the same endpoint)
>
>
> Agreed.  I like your suggested wording.
>
> -d
>
>
>
So, I note that in this case where a non-temporary IPv6 address is present
and  no temporary IPv6 address is present, this appears to push IPv6 out of
the gathered list completely.  If I have that right, then my view as an
individual is that this is the wrong result.  It will either force the use
of IPv4 addresses which are just as linkable as IPv6 non-temporary
addresses or rely on NATs to get the non-linkability (and provide us all
the other subtle joys of NAT).

As a friendly amendment, may I suggest "Where both non-temporary and
temporary addresses are present and host and network policy permit, RTCWEB
implementations SHOULD gather IPv6 temporary addresses and SHOULD NOT
gather non-temporary addresses"?

I also confess to a suspicion that Harald's view is the most
sensible--having a separate policy for this application either won't happen
or doesn't make much sense.  But if we have one, I'd prefer one that
doesn't shove IPv6 out the door completely if the host doesn't use
temporary addresses.

regards,

Ted

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Mar 20, 2014 at 10:07 A=
M, Dan Wing <span dir=3D"ltr">&lt;<a href=3D"mailto:dwing@cisco.com" target=
=3D"_blank">dwing@cisco.com</a>&gt;</span> wrote:<br><div class=3D"gmail_qu=
ote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left:1px solid rgb(204,204,204);padding-left:1ex">
<div style=3D"word-wrap:break-word"><div><div class=3D"h5"><br><div><div>On=
 Mar 20, 2014, at 9:34 AM, Justin Uberti &lt;<a href=3D"mailto:juberti@goog=
le.com" target=3D"_blank">juberti@google.com</a>&gt; wrote:</div><br><block=
quote type=3D"cite">
<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote"><br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D=
"word-wrap:break-word">
<div><div>So perhaps:</div><div>=A0 =A0&quot;An RTCWEB implementation SHOUL=
D prefer to use temporary addresses [RFC4941] where host and network policy=
 permit [RFC6724].&quot;</div><div>?</div>

</div></div></blockquote><div><br></div><div>I think it needs to be stronge=
r than that - something like=A0</div><div>&quot;where <div class=3D"gmail_d=
efault" style=3D"font-family:georgia,serif;display:inline"></div>host and n=
etwork policy permit, RTCWEB implementations SHOULD gather IPv6 temporary a=
ddresses and SHOULD NOT gather non-temporary addresses&quot;.</div>


<div><br></div><div>Preferring to use temporary addresses is probably not s=
ufficient to prevent linkage, since you will have connectivity checks from =
the non-temporary addresses. (i.e. an eavesdropper listening over an extend=
ed period of time could determine calls are from the same endpoint)</div>
</div></div></div>
</blockquote></div><br></div></div><div>Agreed. =A0I like your suggested wo=
rding.</div><span class=3D""><font color=3D"#888888"><div><br></div><div>-d=
</div><div><br></div></font></span></div><br></blockquote><div><br><div cla=
ss=3D"gmail_default" style=3D"font-family:georgia,serif;display:inline">
So, I note that in this case where a non-temporary IPv6 address is present =
and=A0 no temporary IPv6 address is present, this appears to push IPv6 out =
of the gathered list completely.=A0 If I have that right, then my view as a=
n individual is that this is the wrong result.=A0 It will either force the =
use of IPv4 addresses which are just as linkable as IPv6 non-temporary addr=
esses or rely on NATs to get the non-linkability (and provide us all the ot=
her subtle joys of NAT). <br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;d=
isplay:inline">As a friendly amendment, may I suggest &quot;Where both non-=
temporary and temporary addresses are present and host
 and network policy permit, RTCWEB implementations SHOULD gather IPv6=20
temporary addresses and SHOULD NOT gather non-temporary addresses&quot;?<br=
><br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;=
display:inline">I also confess to a suspicion that Harald&#39;s view is the=
 most sensible--having a separate policy for this application either won&#3=
9;t happen or doesn&#39;t make much sense.=A0 But if we have one, I&#39;d p=
refer one that doesn&#39;t shove IPv6 out the door completely if the host d=
oesn&#39;t use temporary addresses.<br>
<br>regards,<br><br>Ted<br></div></div></div></div></div>

--089e010d97f8654c8304f50fff5c--


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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: [rtcweb] DRAFT Minutes for Day 2 IETF 89
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Attached are the draft minutes for the second day of the last meeting;
thanks to Dan and Matt for taking the notes.  Please send comments to the
list; we'll then aggregate the edits for day one and day two in order to
produce consolidated minutes.

Ted

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">Attached are the draft minutes for the second day of the last meetin=
g; thanks to Dan and Matt for taking the notes.=A0 Please send comments to =
the list; we&#39;ll then aggregate the edits for day one and day two in ord=
er to produce consolidated minutes.<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">=
Ted<br></div></div>

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From nobody Thu Mar 20 16:04:32 2014
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From: Justin Uberti <juberti@google.com>
Date: Thu, 20 Mar 2014 16:03:52 -0700
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To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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--bcaec5299f9ba0c75404f511c90c
Content-Type: text/plain; charset=UTF-8

Your take is what I had in mind. Basically a ruleset like this:

 gather_ipv4_addresses();
 if (has_ipv6) {
  if (has_temporary_addresses && temporaries_not_forbidden_by_policy) {
    gather_temporary_ipv6_addresses();
  } else {
    gather_non_temporary_ipv6_addresses();
 }
}


On Thu, Mar 20, 2014 at 1:56 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Thu, Mar 20, 2014 at 10:07 AM, Dan Wing <dwing@cisco.com> wrote:
>
>>
>> On Mar 20, 2014, at 9:34 AM, Justin Uberti <juberti@google.com> wrote:
>>
>>
>>
>>
>>
>> So perhaps:
>>>    "An RTCWEB implementation SHOULD prefer to use temporary addresses
>>> [RFC4941] where host and network policy permit [RFC6724]."
>>> ?
>>>
>>
>> I think it needs to be stronger than that - something like
>> "where
>> host and network policy permit, RTCWEB implementations SHOULD gather IPv6
>> temporary addresses and SHOULD NOT gather non-temporary addresses".
>>
>> Preferring to use temporary addresses is probably not sufficient to
>> prevent linkage, since you will have connectivity checks from the
>> non-temporary addresses. (i.e. an eavesdropper listening over an extended
>> period of time could determine calls are from the same endpoint)
>>
>>
>> Agreed.  I like your suggested wording.
>>
>> -d
>>
>>
>>
> So, I note that in this case where a non-temporary IPv6 address is present
> and  no temporary IPv6 address is present, this appears to push IPv6 out of
> the gathered list completely.  If I have that right, then my view as an
> individual is that this is the wrong result.  It will either force the use
> of IPv4 addresses which are just as linkable as IPv6 non-temporary
> addresses or rely on NATs to get the non-linkability (and provide us all
> the other subtle joys of NAT).
>
> As a friendly amendment, may I suggest "Where both non-temporary and
> temporary addresses are present and host and network policy permit, RTCWEB
> implementations SHOULD gather IPv6 temporary addresses and SHOULD NOT
> gather non-temporary addresses"?
>
> I also confess to a suspicion that Harald's view is the most
> sensible--having a separate policy for this application either won't happen
> or doesn't make much sense.  But if we have one, I'd prefer one that
> doesn't shove IPv6 out the door completely if the host doesn't use
> temporary addresses.
>
> regards,
>
> Ted
>

--bcaec5299f9ba0c75404f511c90c
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<div dir=3D"ltr">Your take is what I had in mind. Basically a ruleset like =
this:<div><br></div><div>=C2=A0gather_ipv4_addresses();<br></div><div>=C2=
=A0if (has_ipv6) {</div><div>=C2=A0 if (has_temporary_addresses &amp;&amp; =
temporaries_not_forbidden_by_policy) {</div>

<div>=C2=A0 =C2=A0 gather_temporary_ipv6_addresses();</div><div>=C2=A0 } el=
se {</div><div>=C2=A0 =C2=A0 gather_non_temporary_ipv6_addresses();</div><d=
iv>=C2=A0}</div><div>}=C2=A0</div></div><div class=3D"gmail_extra"><br><br>=
<div class=3D"gmail_quote">On Thu, Mar 20, 2014 at 1:56 PM, Ted Hardie <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">t=
ed.ietf@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div class=3D"">On Thu, Mar 20, 2014 at 10:07 AM, Dan Wing <span dir=3D"ltr=
">&lt;<a href=3D"mailto:dwing@cisco.com" target=3D"_blank">dwing@cisco.com<=
/a>&gt;</span> wrote:<br>

</div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D=
"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-le=
ft:1ex">
<div style=3D"word-wrap:break-word"><div><div><br><div><div class=3D""><div=
>On Mar 20, 2014, at 9:34 AM, Justin Uberti &lt;<a href=3D"mailto:juberti@g=
oogle.com" target=3D"_blank">juberti@google.com</a>&gt; wrote:</div><br></d=
iv>

<div class=3D""><blockquote type=3D"cite">
<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote"><br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D=
"word-wrap:break-word">


<div><div>So perhaps:</div><div>=C2=A0 =C2=A0&quot;An RTCWEB implementation=
 SHOULD prefer to use temporary addresses [RFC4941] where host and network =
policy permit [RFC6724].&quot;</div><div>?</div>

</div></div></blockquote><div><br></div><div>I think it needs to be stronge=
r than that - something like=C2=A0</div><div>&quot;where <div class=3D"gmai=
l_default" style=3D"font-family:georgia,serif;display:inline"></div>host an=
d network policy permit, RTCWEB implementations SHOULD gather IPv6 temporar=
y addresses and SHOULD NOT gather non-temporary addresses&quot;.</div>




<div><br></div><div>Preferring to use temporary addresses is probably not s=
ufficient to prevent linkage, since you will have connectivity checks from =
the non-temporary addresses. (i.e. an eavesdropper listening over an extend=
ed period of time could determine calls are from the same endpoint)</div>


</div></div></div>
</blockquote></div></div><br></div></div><div class=3D""><div>Agreed. =C2=
=A0I like your suggested wording.</div><span><font color=3D"#888888"><div><=
br></div><div>-d</div><div><br></div></font></span></div></div><br></blockq=
uote><div>

<br><div class=3D"gmail_default" style=3D"font-family:georgia,serif;display=
:inline">
So, I note that in this case where a non-temporary IPv6 address is present =
and=C2=A0 no temporary IPv6 address is present, this appears to push IPv6 o=
ut of the gathered list completely.=C2=A0 If I have that right, then my vie=
w as an individual is that this is the wrong result.=C2=A0 It will either f=
orce the use of IPv4 addresses which are just as linkable as IPv6 non-tempo=
rary addresses or rely on NATs to get the non-linkability (and provide us a=
ll the other subtle joys of NAT). <br>


<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;d=
isplay:inline">As a friendly amendment, may I suggest &quot;Where both non-=
temporary and temporary addresses are present and host
 and network policy permit, RTCWEB implementations SHOULD gather IPv6=20
temporary addresses and SHOULD NOT gather non-temporary addresses&quot;?<br=
><br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif;=
display:inline">I also confess to a suspicion that Harald&#39;s view is the=
 most sensible--having a separate policy for this application either won&#3=
9;t happen or doesn&#39;t make much sense.=C2=A0 But if we have one, I&#39;=
d prefer one that doesn&#39;t shove IPv6 out the door completely if the hos=
t doesn&#39;t use temporary addresses.<br>


<br>regards,<br><br>Ted<br></div></div></div></div></div>
</blockquote></div><br></div>

--bcaec5299f9ba0c75404f511c90c--


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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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--047d7b874abe5eed9804f511fa5a
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On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <juberti@google.com> wrote:

> Your take is what I had in mind. Basically a ruleset like this:
>
>  gather_ipv4_addresses();
>  if (has_ipv6) {
>   if (has_temporary_addresses && temporaries_not_forbidden_by_policy) {
>     gather_temporary_ipv6_addresses();
>   } else {
>     gather_non_temporary_ipv6_addresses();
>  }
> }
>
>
What should be done when temporary enabled only on some of the network
interfaces of the device, i.e. if, for instance, WiFI interface has only
non temp ipv6 address and LTE has both temp and permanent address present?
_____________
Roman Shpount

--047d7b874abe5eed9804f511fa5a
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 20, 2014 at 7:03 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D=
"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</s=
pan> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr">Your take is what I had in mind. Basicall=
y a ruleset like this:<div>
<br></div><div>=A0gather_ipv4_addresses();<br></div><div>=A0if (has_ipv6) {=
</div><div>=A0 if (has_temporary_addresses &amp;&amp; temporaries_not_forbi=
dden_by_policy) {</div>

<div>=A0 =A0 gather_temporary_ipv6_addresses();</div><div>=A0 } else {</div=
><div>=A0 =A0 gather_non_temporary_ipv6_addresses();</div><div>=A0}</div><d=
iv>}=A0</div></div><div class=3D"gmail_extra"><br></div></blockquote><div><=
br></div><div>
What should be done when temporary enabled only on some of the network inte=
rfaces of the device, i.e. if, for instance, WiFI interface has only non te=
mp ipv6 address and LTE has both temp and permanent address present?</div>
<div>_____________<br>Roman Shpount</div><div>=A0</div></div></div></div>

--047d7b874abe5eed9804f511fa5a--


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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On Mar 20, 2014, at 1:56 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Thu, Mar 20, 2014 at 10:07 AM, Dan Wing <dwing@cisco.com> wrote:
>=20
> On Mar 20, 2014, at 9:34 AM, Justin Uberti <juberti@google.com> wrote:
>=20
>>=20
>>=20
>>=20
>>=20
>> So perhaps:
>>    "An RTCWEB implementation SHOULD prefer to use temporary addresses =
[RFC4941] where host and network policy permit [RFC6724]."
>> ?
>>=20
>> I think it needs to be stronger than that - something like=20
>> "where host and network policy permit, RTCWEB implementations SHOULD =
gather IPv6 temporary addresses and SHOULD NOT gather non-temporary =
addresses".
>>=20
>> Preferring to use temporary addresses is probably not sufficient to =
prevent linkage, since you will have connectivity checks from the =
non-temporary addresses. (i.e. an eavesdropper listening over an =
extended period of time could determine calls are from the same =
endpoint)
>=20
> Agreed.  I like your suggested wording.
>=20
> -d
>=20
>=20
>=20
> So, I note that in this case where a non-temporary IPv6 address is =
present and  no temporary IPv6 address is present, this appears to push =
IPv6 out of the gathered list completely.  If I have that right, then my =
view as an individual is that this is the wrong result.  It will either =
force the use of IPv4 addresses which are just as linkable as IPv6 =
non-temporary addresses or rely on NATs to get the non-linkability (and =
provide us all the other subtle joys of NAT).=20
>=20
> As a friendly amendment, may I suggest "Where both non-temporary and =
temporary addresses are present and host and network policy permit, =
RTCWEB implementations SHOULD gather IPv6 temporary addresses and SHOULD =
NOT gather non-temporary addresses"?
>=20
> I also confess to a suspicion that Harald's view is the most =
sensible--having a separate policy for this application either won't =
happen or doesn't make much sense.=20

Ok, so preface your suggested sentence with,

  =93WebRTC applications are encouraged to follow IPv6 Socket API for =
Source Address Selection [RFC5014] with regards to IPv6 temporary =
addresses, specifically <insert your sentence>.=94

It=92s not that WebRTC is trying to do something unique here =97 we just =
want this done, and not just hope an implementor stumbles into the right =
IETF document explaining temporary IPv6 addresses.

-d



> But if we have one, I'd prefer one that doesn't shove IPv6 out the =
door completely if the host doesn't use temporary addresses.
>=20
> regards,
>=20
> Ted


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From: Justin Uberti <juberti@google.com>
Date: Thu, 20 Mar 2014 17:28:16 -0700
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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--bcaec51a71907179d804f512f79b
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On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <roman@telurix.com> wrote:

> On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <juberti@google.com> wrote:
>
>> Your take is what I had in mind. Basically a ruleset like this:
>>
>>  gather_ipv4_addresses();
>>  if (has_ipv6) {
>>   if (has_temporary_addresses && temporaries_not_forbidden_by_policy) {
>>     gather_temporary_ipv6_addresses();
>>   } else {
>>     gather_non_temporary_ipv6_addresses();
>>  }
>> }
>>
>>
>  What should be done when temporary enabled only on some of the network
> interfaces of the device, i.e. if, for instance, WiFI interface has only
> non temp ipv6 address and LTE has both temp and permanent address present?
>
>
Is this a real-world problem? As I understand it, temporary addresses are
assigned by the host, so you either support them or you don't.

--bcaec51a71907179d804f512f79b
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <span dir=3D"ltr">&l=
t;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com<=
/a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div class=3D"gmail_quote"><div class=3D"">On Thu, Mar 20, 2014 at 7:03 PM,=
 Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" =
target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr">Your take is what I had in mind. Basicall=
y a ruleset like this:<div>


<br></div><div>=C2=A0gather_ipv4_addresses();<br></div><div>=C2=A0if (has_i=
pv6) {</div><div>=C2=A0 if (has_temporary_addresses &amp;&amp; temporaries_=
not_forbidden_by_policy) {</div>

<div>=C2=A0 =C2=A0 gather_temporary_ipv6_addresses();</div><div>=C2=A0 } el=
se {</div><div>=C2=A0 =C2=A0 gather_non_temporary_ipv6_addresses();</div><d=
iv>=C2=A0}</div><div>}=C2=A0</div></div><div class=3D"gmail_extra"><br></di=
v></blockquote><div><br></div></div>

<div>
What should be done when temporary enabled only on some of the network inte=
rfaces of the device, i.e. if, for instance, WiFI interface has only non te=
mp ipv6 address and LTE has both temp and permanent address present?</div>


<div><span class=3D"HOEnZb"><font color=3D"#888888"><br></font></span></div=
></div></div></div></blockquote><div>=C2=A0</div><div>Is this a real-world =
problem? As I understand it, temporary addresses are assigned by the host, =
so you either support them or you don&#39;t.</div>

</div></div></div>

--bcaec51a71907179d804f512f79b--


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From: Roman Shpount <roman@telurix.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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--047d7b873a100d1ad304f514cccf
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On Thu, Mar 20, 2014 at 8:28 PM, Justin Uberti <juberti@google.com> wrote:

>
> On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <roman@telurix.com> wrote:
>
>> On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <juberti@google.com>wrote:
>>
>>> Your take is what I had in mind. Basically a ruleset like this:
>>>
>>>  gather_ipv4_addresses();
>>>  if (has_ipv6) {
>>>   if (has_temporary_addresses && temporaries_not_forbidden_by_policy) {
>>>     gather_temporary_ipv6_addresses();
>>>   } else {
>>>     gather_non_temporary_ipv6_addresses();
>>>  }
>>> }
>>>
>>>
>>  What should be done when temporary enabled only on some of the network
>> interfaces of the device, i.e. if, for instance, WiFI interface has only
>> non temp ipv6 address and LTE has both temp and permanent address present?
>>
>>
> Is this a real-world problem? As I understand it, temporary addresses are
> assigned by the host, so you either support them or you don't.
>

On Linux you can enable temporary addresses per interface, so it is
possible.

The whole problem (with using temp or permanent addresses) is a bit
imaginary since under most common client setups you only see temporary
addresses. Permanent IPv6 addresses show up only on servers or if
specifically configured on the host.
_____________
Roman Shpount

--047d7b873a100d1ad304f514cccf
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Mar 20, 2014 at 8:28 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D=
"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</s=
pan> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">
<div><div class=3D"h5">On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">rom=
an@telurix.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"=
gmail_quote">
<div>On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <span dir=3D"ltr">&lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr">Your take is what I had in mind. Basicall=
y a ruleset like this:<div>



<br></div><div>=A0gather_ipv4_addresses();<br></div><div>=A0if (has_ipv6) {=
</div><div>=A0 if (has_temporary_addresses &amp;&amp; temporaries_not_forbi=
dden_by_policy) {</div>

<div>=A0 =A0 gather_temporary_ipv6_addresses();</div><div>=A0 } else {</div=
><div>=A0 =A0 gather_non_temporary_ipv6_addresses();</div><div>=A0}</div><d=
iv>}=A0</div></div><div class=3D"gmail_extra"><br></div></blockquote><div><=
br></div></div>


<div>
What should be done when temporary enabled only on some of the network inte=
rfaces of the device, i.e. if, for instance, WiFI interface has only non te=
mp ipv6 address and LTE has both temp and permanent address present?</div>



<div><span><font color=3D"#888888"><br></font></span></div></div></div></di=
v></blockquote><div>=A0</div></div></div><div>Is this a real-world problem?=
 As I understand it, temporary addresses are assigned by the host, so you e=
ither support them or you don&#39;t.</div>


</div></div></div>
</blockquote></div><div><br></div><div>On Linux you can enable temporary ad=
dresses per interface, so it is possible.</div><div><br></div><div>The whol=
e problem (with using temp or permanent addresses) is a bit imaginary since=
 under most common client setups you only see temporary addresses. Permanen=
t IPv6 addresses show up only on servers or if specifically configured on t=
he host.</div>
<div>_____________<br>Roman Shpount</div><div><br></div></div></div>

--047d7b873a100d1ad304f514cccf--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 20 Mar 2014 21:12:59 -0700
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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OK. Perhaps the guidance should be that if an interface has a temporary and
a non-temporary address, only the temporary one should be gathered. This
avoids any problems if local policy chooses not to use temporary addresses.

FWIW, I have seen this setup where a client has both non-temporary and
temporary addresses in the wild, so this is a real-world problem.


On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <roman@telurix.com> wrote:

> On Thu, Mar 20, 2014 at 8:28 PM, Justin Uberti <juberti@google.com> wrote:
>
>>
>> On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <roman@telurix.com> wrote:
>>
>>> On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <juberti@google.com>wrote:
>>>
>>>> Your take is what I had in mind. Basically a ruleset like this:
>>>>
>>>>  gather_ipv4_addresses();
>>>>  if (has_ipv6) {
>>>>   if (has_temporary_addresses && temporaries_not_forbidden_by_policy) {
>>>>     gather_temporary_ipv6_addresses();
>>>>   } else {
>>>>     gather_non_temporary_ipv6_addresses();
>>>>  }
>>>> }
>>>>
>>>>
>>>  What should be done when temporary enabled only on some of the network
>>> interfaces of the device, i.e. if, for instance, WiFI interface has only
>>> non temp ipv6 address and LTE has both temp and permanent address present?
>>>
>>>
>> Is this a real-world problem? As I understand it, temporary addresses are
>> assigned by the host, so you either support them or you don't.
>>
>
> On Linux you can enable temporary addresses per interface, so it is
> possible.
>
> The whole problem (with using temp or permanent addresses) is a bit
> imaginary since under most common client setups you only see temporary
> addresses. Permanent IPv6 addresses show up only on servers or if
> specifically configured on the host.
> _____________
> Roman Shpount
>
>

--047d7b5d5b0a1e72c904f5161b48
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<div dir=3D"ltr">OK. Perhaps the guidance should be that if an interface ha=
s a temporary and a non-temporary address, only the temporary one should be=
 gathered. This avoids any problems if local policy chooses not to use temp=
orary addresses.<div>

<br></div><div>FWIW, I have seen this setup where a client has both non-tem=
porary and temporary addresses in the wild, so this is a real-world problem=
.</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">

On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"h5"><div cla=
ss=3D"gmail_quote">On Thu, Mar 20, 2014 at 8:28 PM, Justin Uberti <span dir=
=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">jubert=
i@google.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">


<div><div>On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <span dir=3D"ltr">=
&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.co=
m</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"=
gmail_quote">


<div>On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <span dir=3D"ltr">&lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr">Your take is what I had in mind. Basicall=
y a ruleset like this:<div>





<br></div><div>=C2=A0gather_ipv4_addresses();<br></div><div>=C2=A0if (has_i=
pv6) {</div><div>=C2=A0 if (has_temporary_addresses &amp;&amp; temporaries_=
not_forbidden_by_policy) {</div>

<div>=C2=A0 =C2=A0 gather_temporary_ipv6_addresses();</div><div>=C2=A0 } el=
se {</div><div>=C2=A0 =C2=A0 gather_non_temporary_ipv6_addresses();</div><d=
iv>=C2=A0}</div><div>}=C2=A0</div></div><div class=3D"gmail_extra"><br></di=
v></blockquote><div><br></div></div>




<div>
What should be done when temporary enabled only on some of the network inte=
rfaces of the device, i.e. if, for instance, WiFI interface has only non te=
mp ipv6 address and LTE has both temp and permanent address present?</div>





<div><span><font color=3D"#888888"><br></font></span></div></div></div></di=
v></blockquote><div>=C2=A0</div></div></div><div>Is this a real-world probl=
em? As I understand it, temporary addresses are assigned by the host, so yo=
u either support them or you don&#39;t.</div>




</div></div></div>
</blockquote></div><div><br></div></div></div><div>On Linux you can enable =
temporary addresses per interface, so it is possible.</div><div><br></div><=
div>The whole problem (with using temp or permanent addresses) is a bit ima=
ginary since under most common client setups you only see temporary address=
es. Permanent IPv6 addresses show up only on servers or if specifically con=
figured on the host.</div>


<div>_____________<span class=3D"HOEnZb"><font color=3D"#888888"><br>Roman =
Shpount</font></span></div><div><br></div></div></div>
</blockquote></div><br></div>

--047d7b5d5b0a1e72c904f5161b48--


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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On 21 Mar 2014, at 01:28, Justin Uberti <juberti@google.com> wrote:

>=20
>=20
>=20
> On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <roman@telurix.com> =
wrote:
> On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <juberti@google.com> =
wrote:
> Your take is what I had in mind. Basically a ruleset like this:
>=20
>  gather_ipv4_addresses();
>  if (has_ipv6) {
>   if (has_temporary_addresses && temporaries_not_forbidden_by_policy) =
{
>     gather_temporary_ipv6_addresses();
>   } else {
>     gather_non_temporary_ipv6_addresses();
>  }
> }=20
>=20
>=20
> What should be done when temporary enabled only on some of the network =
interfaces of the device, i.e. if, for instance, WiFI interface has only =
non temp ipv6 address and LTE has both temp and permanent address =
present?
>=20
> =20
> Is this a real-world problem? As I understand it, temporary addresses =
are assigned by the host, so you either support them or you don't.

There is a way to get them from a DHCP server.
http://tools.ietf.org/html/rfc3315#section-12

/O


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<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><br><div><div>On 21 Mar 2014, at 01:28, Justin Uberti &lt;<a href="mailto:juberti@google.com">juberti@google.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr"><br><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <span dir="ltr">&lt;<a href="mailto:roman@telurix.com" target="_blank">roman@telurix.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><div class="">On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <span dir="ltr">&lt;<a href="mailto:juberti@google.com" target="_blank">juberti@google.com</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Your take is what I had in mind. Basically a ruleset like this:<div>


<br></div><div>&nbsp;gather_ipv4_addresses();<br></div><div>&nbsp;if (has_ipv6) {</div><div>&nbsp; if (has_temporary_addresses &amp;&amp; temporaries_not_forbidden_by_policy) {</div>

<div>&nbsp; &nbsp; gather_temporary_ipv6_addresses();</div><div>&nbsp; } else {</div><div>&nbsp; &nbsp; gather_non_temporary_ipv6_addresses();</div><div>&nbsp;}</div><div>}&nbsp;</div></div><div class="gmail_extra"><br></div></blockquote><div><br></div></div>

<div>
What should be done when temporary enabled only on some of the network interfaces of the device, i.e. if, for instance, WiFI interface has only non temp ipv6 address and LTE has both temp and permanent address present?</div>


<div><span class="HOEnZb"><font color="#888888"><br></font></span></div></div></div></div></blockquote><div>&nbsp;</div><div>Is this a real-world problem? As I understand it, temporary addresses are assigned by the host, so you either support them or you don't.</div>

</div></div></div></blockquote><br></div><div>There is a way to get them from a DHCP server.</div><div><a href="http://tools.ietf.org/html/rfc3315#section-12">http://tools.ietf.org/html/rfc3315#section-12</a></div><div><br></div><div>/O</div><br></body></html>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On 03/20/2014 03:52 PM, Dan Wing wrote:
>
> On Mar 19, 2014, at 8:21 AM, Harald Alvestrand <harald@alvestrand.no
> <mailto:harald@alvestrand.no>> wrote:
>
>> On 03/19/2014 04:08 PM, Dan Wing wrote:
>>>
>>> On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com
>>> <mailto:hta@google.com>> wrote:
>>>
>>>> I'd like to be silent on the issue, since which IPv6 addresses to
>>>> prefer is likely to be a matter of system policy. Trying to
>>>> override system policy in an application specific profile usually
>>>> leads to sadness.
>>>
>>> The application needs to indicate if it wants a temporary address.
>>> If the host's policy (or configuration or network) does not support
>>> temporary addresses, the application won't get a temporary address.
>>>  I don't see why being silent helps?
>>
>> What API is it using?
>>
>> With a little Googling, the system policy I was thinking of was the
>> policy in RFC 6724 ("Default Address Selection for IPv6"), in
>> particular section 5 rule 7: "Prefer  temporary addresses".
>>
>> I'm happy to say "it is a good idea for systems to implement the
>> recommendations of RFC 6724" (or some more 2119-like language). I
>> wouldn't want to claim that if a system has chosen to prefer
>> non-temporary addresses, it would have to change its non-conformance
>> to RFC 6724 in order to be conformant with RTCWEB specifications.
>
> So perhaps:
>    "An RTCWEB implementation SHOULD prefer to use temporary addresses
> [RFC4941] where host and network policy permit [RFC6724]."
> ?
Or even be more explicit on what's being proposed:

"When a client gathers all IPv6 addresses on a host, and both temporary
addresses [RFC4941] and permanent addresses of the same scope are
present, the client SHOULD discard the permanent addresses before
forming pairs. This is consistent with the default policy described in
[RFC6724]."


Pulling back a little, I am starting to wonder if this is an ICE concern
rather than an RTCWEB concern. The explosion in candidate pairs as more
transport / network options come on line is a real operational problem,
and needs to be solved for all ICE usages.

I'll stick something in -transport, but RTCWEB is not the only thing
that will have this problem.


>
> -d
>
>
>>
>>>
>>> -d
>>>
>>>
>>>>
>>>>
>>>>
>>>> On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com
>>>> <mailto:dwing@cisco.com>> wrote:
>>>>
>>>>
>>>>     On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com
>>>>     <mailto:juberti@google.com>> wrote:
>>>>
>>>>>     https://tools.ietf.org/html/rfc4941 defines the concept of
>>>>>     temporary IPv6 addresses. For, example, as enumerated on my
>>>>>     local system:
>>>>>
>>>>>     inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
>>>>>     inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf 
>>>>>     inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64
>>>>>     autoconf *temporary *
>>>>>
>>>>>     As indicated in the RFC, the temporary addresses expire after
>>>>>     hours or days, and therefore could be used to prevent
>>>>>     long-term linkability of sessions. Expiration shouldn't be an
>>>>>     issue for WebRTC, since we can simply do ICE restart if this
>>>>>     occurs during a session.
>>>>>
>>>>>     Is this something we want to recommend in the transports doc?
>>>>
>>>>     Yes.
>>>>
>>>>     And it should be as frequent as every new 'call', if IPv6
>>>>     privacy addresses are to provide the same privacy as a NAPT44,
>>>>     as a NAPT44 should be assigning random ports per reasons
>>>>     described in RFC6056.
>>>>
>>>>     The transport document should also recommend doing port
>>>>     randomization (RFC6056), as that was found pretty useful for
>>>>     DNS, but randomizing ports is still not popular with many OS's
>>>>     TCP stacks for whatever reason.
>>>>
>>>>     -d
>>>>
>>>>
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> https://www.ietf.org/mailman/listinfo/rtcweb
>


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 03/20/2014 03:52 PM, Dan Wing wrote:<br>
    </div>
    <blockquote
      cite="mid:17885A74-50A3-49E3-8C54-E53C55019C73@cisco.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <br>
      <div>
        <div>On Mar 19, 2014, at 8:21 AM, Harald Alvestrand &lt;<a
            moz-do-not-send="true" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;
          wrote:</div>
        <br class="Apple-interchange-newline">
        <blockquote type="cite">
          <div bgcolor="#FFFFFF" text="#000000" style="font-family:
            Helvetica; font-size: 12px; font-style: normal;
            font-variant: normal; font-weight: normal; letter-spacing:
            normal; line-height: normal; orphans: auto; text-align:
            start; text-indent: 0px; text-transform: none; white-space:
            normal; widows: auto; word-spacing: 0px;
            -webkit-text-stroke-width: 0px;">
            <div class="moz-cite-prefix">On 03/19/2014 04:08 PM, Dan
              Wing wrote:<br>
            </div>
            <blockquote
              cite="mid:CD747481-EBDA-4FFC-A31D-618E6E217420@cisco.com"
              type="cite"><br>
              <div>
                <div>On Mar 19, 2014, at 12:02 AM, Harald Alvestrand
                  &lt;<a moz-do-not-send="true"
                    href="mailto:hta@google.com">hta@google.com</a>&gt;
                  wrote:</div>
                <br class="Apple-interchange-newline">
                <blockquote type="cite">
                  <div dir="ltr">I'd like to be silent on the issue,
                    since which IPv6 addresses to prefer is likely to be
                    a matter of system policy. Trying to override system
                    policy in an application specific profile usually
                    leads to sadness.</div>
                </blockquote>
                <div><br>
                </div>
                <div>The application needs to indicate if it wants a
                  temporary address. If the host's policy (or
                  configuration or network) does not support temporary
                  addresses, the application won't get a temporary
                  address. &nbsp;I don't see why being silent helps?</div>
              </div>
            </blockquote>
            <br>
            What API is it using?<br>
            <br>
            With a little Googling, the system policy I was thinking of
            was the policy in RFC 6724 ("Default Address Selection for
            IPv6"), in particular section 5 rule 7: "Prefer&nbsp; temporary
            addresses".<br>
            <br>
            I'm happy to say "it is a good idea for systems to implement
            the recommendations of RFC 6724" (or some more 2119-like
            language). I wouldn't want to claim that if a system has
            chosen to prefer non-temporary addresses, it would have to
            change its non-conformance to RFC 6724 in order to be
            conformant with RTCWEB specifications.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>So perhaps:</div>
        <div>&nbsp; &nbsp;"An RTCWEB implementation SHOULD prefer to use temporary
          addresses [RFC4941] where host and network policy permit
          [RFC6724]."</div>
        <div>?</div>
      </div>
    </blockquote>
    Or even be more explicit on what's being proposed:<br>
    <br>
    "When a client gathers all IPv6 addresses on a host, and both
    temporary addresses [RFC4941] and permanent addresses of the same
    scope are present, the client SHOULD discard the permanent addresses
    before forming pairs. This is consistent with the default policy
    described in [RFC6724]."<br>
    <br>
    <br>
    Pulling back a little, I am starting to wonder if this is an ICE
    concern rather than an RTCWEB concern. The explosion in candidate
    pairs as more transport / network options come on line is a real
    operational problem, and needs to be solved for all ICE usages.<br>
    <br>
    I'll stick something in -transport, but RTCWEB is not the only thing
    that will have this problem.<br>
    <br>
    <br>
    <blockquote
      cite="mid:17885A74-50A3-49E3-8C54-E53C55019C73@cisco.com"
      type="cite">
      <div>
        <div><br>
        </div>
        <div>-d</div>
        <div><br>
        </div>
        <br>
        <blockquote type="cite">
          <div bgcolor="#FFFFFF" text="#000000" style="font-family:
            Helvetica; font-size: 12px; font-style: normal;
            font-variant: normal; font-weight: normal; letter-spacing:
            normal; line-height: normal; orphans: auto; text-align:
            start; text-indent: 0px; text-transform: none; white-space:
            normal; widows: auto; word-spacing: 0px;
            -webkit-text-stroke-width: 0px;"><br>
            <blockquote
              cite="mid:CD747481-EBDA-4FFC-A31D-618E6E217420@cisco.com"
              type="cite">
              <div>
                <div><br>
                </div>
                <div>-d</div>
                <div><br>
                </div>
                <br>
                <blockquote type="cite">
                  <div dir="ltr"><br>
                  </div>
                  <div class="gmail_extra"><br>
                    <br>
                    <div class="gmail_quote">On Wed, Mar 19, 2014 at
                      6:14 AM, Dan Wing<span
                        class="Apple-converted-space">&nbsp;</span><span
                        dir="ltr">&lt;<a moz-do-not-send="true"
                          href="mailto:dwing@cisco.com" target="_blank">dwing@cisco.com</a>&gt;</span><span
                        class="Apple-converted-space">&nbsp;</span>wrote:<br>
                      <blockquote class="gmail_quote" style="margin: 0px
                        0px 0px 0.8ex; border-left-width: 1px;
                        border-left-color: rgb(204, 204, 204);
                        border-left-style: solid; padding-left: 1ex;">
                        <div style="word-wrap: break-word;">
                          <div>
                            <div class="h5"><br>
                              <div>
                                <div>On Mar 18, 2014, at 6:00 PM, Justin
                                  Uberti &lt;<a moz-do-not-send="true"
                                    href="mailto:juberti@google.com"
                                    target="_blank">juberti@google.com</a>&gt;
                                  wrote:</div>
                                <br>
                                <blockquote type="cite">
                                  <div dir="ltr"><a
                                      moz-do-not-send="true"
                                      href="https://tools.ietf.org/html/rfc4941"
                                      target="_blank">https://tools.ietf.org/html/rfc4941</a><span
                                      class="Apple-converted-space">&nbsp;</span>defines
                                    the concept of temporary IPv6
                                    addresses. For, example, as
                                    enumerated on my local system:<br>
                                    <div><br>
                                    </div>
                                    <div><span>inet 172.31.x.y netmask
                                        0xfffffe00 broadcast
                                        172.31.x.255</span><br>
                                      <span>inet6
                                        2620::1008:100b:e2f8:47ff:wwww</span><span>:xxxx
                                        prefixlen 64 autoconf&nbsp;</span><br>
                                      <span>inet6
                                        2620::1008:100b:819e:1d3f:yyyy</span><span>:zzzz
                                        prefixlen 64 autoconf<span
                                          class="Apple-converted-space">&nbsp;</span><b>temporary&nbsp;</b></span><br>
                                    </div>
                                    <div><span><br>
                                      </span></div>
                                    <div>As indicated in the RFC, the
                                      temporary addresses expire after
                                      hours or days, and therefore could
                                      be used to prevent long-term
                                      linkability of sessions.
                                      Expiration shouldn't be an issue
                                      for WebRTC, since we can simply do
                                      ICE restart if this occurs during
                                      a session.</div>
                                    <div><br>
                                    </div>
                                    <div>Is this something we want to
                                      recommend in the transports doc?<br>
                                    </div>
                                  </div>
                                </blockquote>
                                <br>
                              </div>
                            </div>
                          </div>
                          <div>Yes.</div>
                          <div><br>
                          </div>
                          <div>And it should be as frequent as every new
                            'call', if IPv6 privacy addresses are to
                            provide the same privacy as a NAPT44, as a
                            NAPT44 should be assigning random ports per
                            reasons described in RFC6056.</div>
                          <div><br>
                          </div>
                          <div>The transport document should also
                            recommend doing port randomization
                            (RFC6056), as that was found pretty useful
                            for DNS, but randomizing ports is still not
                            popular with many OS's TCP stacks for
                            whatever reason.</div>
                          <span class="HOEnZb"><font color="#888888">
                              <div><br>
                              </div>
                              <div>-d</div>
                              <div><br>
                              </div>
                              <br>
                            </font></span></div>
                      </blockquote>
                    </div>
                    <br>
                  </div>
                </blockquote>
              </div>
              <br>
              <br>
              <fieldset class="mimeAttachmentHeader"></fieldset>
              <br>
              <pre wrap="">_______________________________________________
rtcweb mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
            </blockquote>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </div>
        </blockquote>
      </div>
      <br>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
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Subject: Re: [rtcweb] On ptimes and maxptimes
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On 2014-03-20 17:39, Justin Uberti wrote:
> 
> 
> 
> On Thu, Mar 20, 2014 at 3:53 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     On 2014-03-20 01:27, Justin Uberti wrote:
>     >
>     >
>     >
>     > On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund
>     > <magnus.westerlund@ericsson.com
>     <mailto:magnus.westerlund@ericsson.com>
>     <mailto:magnus.westerlund@ericsson.com
>     <mailto:magnus.westerlund@ericsson.com>>>
>     > wrote:
>     >
>     >     Hi,
>     >
>     >     (As individual)
>     >
>     >     On 2014-03-05 17:26, Justin Uberti wrote:
>     >     > From the session this morning, I recall this consensus:
>     >     >
>     >     >   * We will add recommendations on the frame sizes to use for
>     >     PCMU/A to
>     >     >     the rtcweb audio codec I-D (20, 30, 60 ms frames).
>     >
>     >     The above applies to sending payloads
>     >
>     >
>     > Right, thanks for pointing that out.
>     >
>     >
>     >     Yes, but that was given that a receiver will be capable of
>     receiving any
>     >     valid number of frames, or samples within a reasonable
>     packetization,
>     >     and I think that should be 120 ms.
>     >
>     >     >   * We will consider adding an a=maxptime attribute that
>     >     represents the
>     >     >     "minimum maxptime" of all the offered codecs. That is,
>     if both
>     >     Opus
>     >     >     (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a
>     >     >     maxptime=60 SHOULD be included.
>     >     >       o Since maxptime spans all codecs, this means that
>     some modes
>     >     >         (e.g. Opus 120ms) will be unavailable, unless only
>     Opus is
>     >     offered.
>     >
>     >     I think you base the assumption on maxptime based on the sending
>     >     recommendations, not what is capable of receiving. If one is
>     capable of
>     >     receiving 200 ms of a-law and Opus, then you would need to say
>     120 ms as
>     >     Opus will limit you.
>     >
>     >
>     > I agree that Opus' maxptime is 120, but should that prevent
>     sending PCMU
>     > with 200 ms? That is, since sending Opus with 200 ms is not legal, is
>     > there any point in forcing WebRTC impls to indicate this in SDP?
> 
>     What I wished we fixed this issue with maxptime a long time ago. This is
>     a well known problem that it really has PT scope, but are signalled on
>     m= line scope.
> 
>     >
>     >
>     >
>     >     >   * We will consider adding an a=ptime attribute, set to the
>     >     receiver's
>     >     >     preferred frame size to receive. Again, this value spans all
>     >     codecs,
>     >     >     so the specified value may not be viable for all codecs
>     (e.g.
>     >     30 ms
>     >     >     works for PCMU but not for Opus)
>     >     >
>     >     > After thinking about this some, I'm not sure the
>     maxptime/ptime values
>     >     > will lead to any different behavior than if they were not
>     specified.
>     >     > Since there is a complexity cost (especially if the values
>     need to
>     >     > change based on which codecs are offered), is there a clear
>     upside
>     >     here?
>     >
>     >     The maxptime values may in some cases be limited downwards for
>     >     interoperability, for example if one like to gateway AMR into a CS
>     >     network then the gateway might indicate a maxptime that is
>     less, more
>     >     like 20 or 40 ms.
>     >
>     >
>     > Sure - WebRTC impls need to respect any maxptime value that they
>     > receive, but it's not clear to me that they need to include one in the
>     > SDP they send.
> 
>     Ok, but then we are back to my previous argument for writing down what
>     receiver requirement in regards to each payload type one is required to
>     support in the audio draft for the ones specified there. But what about
>     the other codecs that aren't included in the audio draft?
> 
>     How can a peer know if there are limitations? I wouldn't attempt to
>     solve the general problem of the scope issue with maxptime. Simply state
>     that you should include this. It is going to be limited but still
>     provide information in many cases that is quite useful for the sender.
> 
> 
> Just to be sure I understand, your position is that WebRTC impls SHOULD
> include an a=maxptime attribute indicating the "minimum maximum" frame
> size they can receive, across all codecs?

Yes, as long as we lack payload type specific, I think that is the best
we can do.
> 
> Do you think anything needs to be included regarding a=ptime?

I think it MAY be included to indicate an initial preference for
packetization interval. In cases that is not common across the set of
payload types for that m= section, then one should skip it.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Date: Fri, 21 Mar 2014 12:14:20 +0100
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On 03/19/2014 04:41 PM, Simon Perreault wrote:
> Le 2014-03-19 11:21, Harald Alvestrand a écrit :
>>> The application needs to indicate if it wants a temporary address. If
>>> the host's policy (or configuration or network) does not support
>>> temporary addresses, the application won't get a temporary address.  I
>>> don't see why being silent helps?
>> What API is it using?
>>
>> With a little Googling, the system policy I was thinking of was the
>> policy in RFC 6724 ("Default Address Selection for IPv6"), in particular
>> section 5 rule 7: "Prefer  temporary addresses".
>>
>> I'm happy to say "it is a good idea for systems to implement the
>> recommendations of RFC 6724" (or some more 2119-like language). I
>> wouldn't want to claim that if a system has chosen to prefer
>> non-temporary addresses, it would have to change its non-conformance to
>> RFC 6724 in order to be conformant with RTCWEB specifications.
> IMHO it would be perfectly sensible to recommend the use of
> IPV6_PREFER_SRC_TMP [RFC5014].
>
> Simon
Interesting: This rule was reversed from 3484 to 6724 - 3484 preferred 
public, 6724 prefers temporary. 5014 claims that the default is 
PREFER_SRC_PUBLIC, which is not 6724 conformant. And RFC 5245 (ICE) 
contains a pointer to RFC 3484, recommending its use in address selection.

With this amount of misleading stuff in RFCs, I think it's best to say 
something after all.


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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On Fri, Mar 21, 2014 at 1:15 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  On 03/20/2014 03:52 PM, Dan Wing wrote:
>
>
>  On Mar 19, 2014, at 8:21 AM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
>  On 03/19/2014 04:08 PM, Dan Wing wrote:
>
>
>  On Mar 19, 2014, at 12:02 AM, Harald Alvestrand <hta@google.com> wrote:
>
>  I'd like to be silent on the issue, since which IPv6 addresses to prefer
> is likely to be a matter of system policy. Trying to override system policy
> in an application specific profile usually leads to sadness.
>
>
>  The application needs to indicate if it wants a temporary address. If
> the host's policy (or configuration or network) does not support temporary
> addresses, the application won't get a temporary address.  I don't see why
> being silent helps?
>
>
> What API is it using?
>
> With a little Googling, the system policy I was thinking of was the policy
> in RFC 6724 ("Default Address Selection for IPv6"), in particular section 5
> rule 7: "Prefer  temporary addresses".
>
> I'm happy to say "it is a good idea for systems to implement the
> recommendations of RFC 6724" (or some more 2119-like language). I wouldn't
> want to claim that if a system has chosen to prefer non-temporary
> addresses, it would have to change its non-conformance to RFC 6724 in order
> to be conformant with RTCWEB specifications.
>
>
>  So perhaps:
>    "An RTCWEB implementation SHOULD prefer to use temporary addresses
> [RFC4941] where host and network policy permit [RFC6724]."
> ?
>
> Or even be more explicit on what's being proposed:
>
> "When a client gathers all IPv6 addresses on a host, and both temporary
> addresses [RFC4941] and permanent addresses of the same scope are present,
> the client SHOULD discard the permanent addresses before forming pairs.
> This is consistent with the default policy described in [RFC6724]."
>

The permanent addresses need to be discarded before sending them to the
remote side (which happens before pairing). I would simply say that the
permanent addresses should be ignored.

>
>
> Pulling back a little, I am starting to wonder if this is an ICE concern
> rather than an RTCWEB concern. The explosion in candidate pairs as more
> transport / network options come on line is a real operational problem, and
> needs to be solved for all ICE usages.
>
> I'll stick something in -transport, but RTCWEB is not the only thing that
> will have this problem.
>
>
>
>
>  -d
>
>
>
>
>  -d
>
>
>
>
>
> On Wed, Mar 19, 2014 at 6:14 AM, Dan Wing <dwing@cisco.com> wrote:
>
>>
>>  On Mar 18, 2014, at 6:00 PM, Justin Uberti <juberti@google.com> wrote:
>>
>>  https://tools.ietf.org/html/rfc4941 defines the concept of temporary
>> IPv6 addresses. For, example, as enumerated on my local system:
>>
>>  inet 172.31.x.y netmask 0xfffffe00 broadcast 172.31.x.255
>> inet6 2620::1008:100b:e2f8:47ff:wwww:xxxx prefixlen 64 autoconf
>> inet6 2620::1008:100b:819e:1d3f:yyyy:zzzz prefixlen 64 autoconf
>> *temporary *
>>
>>  As indicated in the RFC, the temporary addresses expire after hours or
>> days, and therefore could be used to prevent long-term linkability of
>> sessions. Expiration shouldn't be an issue for WebRTC, since we can simply
>> do ICE restart if this occurs during a session.
>>
>>  Is this something we want to recommend in the transports doc?
>>
>>
>>   Yes.
>>
>>  And it should be as frequent as every new 'call', if IPv6 privacy
>> addresses are to provide the same privacy as a NAPT44, as a NAPT44 should
>> be assigning random ports per reasons described in RFC6056.
>>
>>  The transport document should also recommend doing port randomization
>> (RFC6056), as that was found pretty useful for DNS, but randomizing ports
>> is still not popular with many OS's TCP stacks for whatever reason.
>>
>>  -d
>>
>>
>>
>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Mar 21, 2014 at 1:15 AM, Harald Alvestrand <span dir=3D"ltr=
">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alve=
strand.no</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"">
    <div>On 03/20/2014 03:52 PM, Dan Wing wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <br>
      <div>
        <div>On Mar 19, 2014, at 8:21 AM, Harald Alvestrand &lt;<a href=3D"=
mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;
          wrote:</div>
        <br>
        <blockquote type=3D"cite">
          <div bgcolor=3D"#FFFFFF" text=3D"#000000" style=3D"font-family:He=
lvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:no=
rmal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:=
0px;text-transform:none;white-space:normal;word-spacing:0px">


            <div>On 03/19/2014 04:08 PM, Dan
              Wing wrote:<br>
            </div>
            <blockquote type=3D"cite"><br>
              <div>
                <div>On Mar 19, 2014, at 12:02 AM, Harald Alvestrand
                  &lt;<a href=3D"mailto:hta@google.com" target=3D"_blank">h=
ta@google.com</a>&gt;
                  wrote:</div>
                <br>
                <blockquote type=3D"cite">
                  <div dir=3D"ltr">I&#39;d like to be silent on the issue,
                    since which IPv6 addresses to prefer is likely to be
                    a matter of system policy. Trying to override system
                    policy in an application specific profile usually
                    leads to sadness.</div>
                </blockquote>
                <div><br>
                </div>
                <div>The application needs to indicate if it wants a
                  temporary address. If the host&#39;s policy (or
                  configuration or network) does not support temporary
                  addresses, the application won&#39;t get a temporary
                  address. =C2=A0I don&#39;t see why being silent helps?</d=
iv>
              </div>
            </blockquote>
            <br>
            What API is it using?<br>
            <br>
            With a little Googling, the system policy I was thinking of
            was the policy in RFC 6724 (&quot;Default Address Selection for
            IPv6&quot;), in particular section 5 rule 7: &quot;Prefer=C2=A0=
 temporary
            addresses&quot;.<br>
            <br>
            I&#39;m happy to say &quot;it is a good idea for systems to imp=
lement
            the recommendations of RFC 6724&quot; (or some more 2119-like
            language). I wouldn&#39;t want to claim that if a system has
            chosen to prefer non-temporary addresses, it would have to
            change its non-conformance to RFC 6724 in order to be
            conformant with RTCWEB specifications.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>So perhaps:</div>
        <div>=C2=A0 =C2=A0&quot;An RTCWEB implementation SHOULD prefer to u=
se temporary
          addresses [RFC4941] where host and network policy permit
          [RFC6724].&quot;</div>
        <div>?</div>
      </div>
    </blockquote></div>
    Or even be more explicit on what&#39;s being proposed:<br>
    <br>
    &quot;When a client gathers all IPv6 addresses on a host, and both
    temporary addresses [RFC4941] and permanent addresses of the same
    scope are present, the client SHOULD discard the permanent addresses
    before forming pairs. This is consistent with the default policy
    described in [RFC6724].&quot;<br></div></blockquote><div><br></div><div=
>The permanent addresses need to be discarded before sending them to the re=
mote side (which happens before pairing). I would simply say that the perma=
nent addresses should be ignored.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <br>
    <br>
    Pulling back a little, I am starting to wonder if this is an ICE
    concern rather than an RTCWEB concern. The explosion in candidate
    pairs as more transport / network options come on line is a real
    operational problem, and needs to be solved for all ICE usages.<br>
    <br>
    I&#39;ll stick something in -transport, but RTCWEB is not the only thin=
g
    that will have this problem.<div class=3D""><br>
    <br>
    <br>
    <blockquote type=3D"cite">
      <div>
        <div><br>
        </div>
        <div>-d</div>
        <div><br>
        </div>
        <br>
        <blockquote type=3D"cite">
          <div bgcolor=3D"#FFFFFF" text=3D"#000000" style=3D"font-family:He=
lvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:no=
rmal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:=
0px;text-transform:none;white-space:normal;word-spacing:0px">

<br>
            <blockquote type=3D"cite">
              <div>
                <div><br>
                </div>
                <div>-d</div>
                <div><br>
                </div>
                <br>
                <blockquote type=3D"cite">
                  <div dir=3D"ltr"><br>
                  </div>
                  <div class=3D"gmail_extra"><br>
                    <br>
                    <div class=3D"gmail_quote">On Wed, Mar 19, 2014 at
                      6:14 AM, Dan Wing<span>=C2=A0</span><span dir=3D"ltr"=
>&lt;<a href=3D"mailto:dwing@cisco.com" target=3D"_blank">dwing@cisco.com</=
a>&gt;</span><span>=C2=A0</span>wrote:<br>
                      <blockquote class=3D"gmail_quote" style=3D"margin:0px=
 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bor=
der-left-style:solid;padding-left:1ex">
                        <div style=3D"word-wrap:break-word">
                          <div>
                            <div><br>
                              <div>
                                <div>On Mar 18, 2014, at 6:00 PM, Justin
                                  Uberti &lt;<a href=3D"mailto:juberti@goog=
le.com" target=3D"_blank">juberti@google.com</a>&gt;
                                  wrote:</div>
                                <br>
                                <blockquote type=3D"cite">
                                  <div dir=3D"ltr"><a href=3D"https://tools=
.ietf.org/html/rfc4941" target=3D"_blank">https://tools.ietf.org/html/rfc49=
41</a><span>=C2=A0</span>defines
                                    the concept of temporary IPv6
                                    addresses. For, example, as
                                    enumerated on my local system:<br>
                                    <div><br>
                                    </div>
                                    <div><span>inet 172.31.x.y netmask
                                        0xfffffe00 broadcast
                                        172.31.x.255</span><br>
                                      <span>inet6
                                        2620::1008:100b:e2f8:47ff:wwww</spa=
n><span>:xxxx
                                        prefixlen 64 autoconf=C2=A0</span><=
br>
                                      <span>inet6
                                        2620::1008:100b:819e:1d3f:yyyy</spa=
n><span>:zzzz
                                        prefixlen 64 autoconf<span>=C2=A0</=
span><b>temporary=C2=A0</b></span><br>
                                    </div>
                                    <div><span><br>
                                      </span></div>
                                    <div>As indicated in the RFC, the
                                      temporary addresses expire after
                                      hours or days, and therefore could
                                      be used to prevent long-term
                                      linkability of sessions.
                                      Expiration shouldn&#39;t be an issue
                                      for WebRTC, since we can simply do
                                      ICE restart if this occurs during
                                      a session.</div>
                                    <div><br>
                                    </div>
                                    <div>Is this something we want to
                                      recommend in the transports doc?<br>
                                    </div>
                                  </div>
                                </blockquote>
                                <br>
                              </div>
                            </div>
                          </div>
                          <div>Yes.</div>
                          <div><br>
                          </div>
                          <div>And it should be as frequent as every new
                            &#39;call&#39;, if IPv6 privacy addresses are t=
o
                            provide the same privacy as a NAPT44, as a
                            NAPT44 should be assigning random ports per
                            reasons described in RFC6056.</div>
                          <div><br>
                          </div>
                          <div>The transport document should also
                            recommend doing port randomization
                            (RFC6056), as that was found pretty useful
                            for DNS, but randomizing ports is still not
                            popular with many OS&#39;s TCP stacks for
                            whatever reason.</div>
                          <span><font color=3D"#888888">
                              <div><br>
                              </div>
                              <div>-d</div>
                              <div><br>
                              </div>
                              <br>
                            </font></span></div>
                      </blockquote>
                    </div>
                    <br>
                  </div>
                </blockquote>
              </div>
              <br>
              <br>
              <fieldset></fieldset>
              <br>
              <pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
            </blockquote>
            <br>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@iet=
f.org</a><br>
            <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </div>
        </blockquote>
      </div>
      <br>
    </blockquote>
    <br>
    <br>
    </div><span class=3D"HOEnZb"><font color=3D"#888888"><pre cols=3D"72">-=
-=20
Surveillance is pervasive. Go Dark.
</pre>
  </font></span></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a113656384861ad04f51f862c--


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From: Ted Hardie <ted.ietf@gmail.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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--089e0158aa1266176e04f520159b
Content-Type: text/plain; charset=ISO-8859-1

On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <roman@telurix.com> wrote:

> On Thu, Mar 20, 2014 at 8:28 PM, Justin Uberti <juberti@google.com> wrote:
>
>>
>> On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <roman@telurix.com> wrote:
>>
>>> On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <juberti@google.com>wrote:
>>>
>>>> Your take is what I had in mind. Basically a ruleset like this:
>>>>
>>>>  gather_ipv4_addresses();
>>>>  if (has_ipv6) {
>>>>   if (has_temporary_addresses && temporaries_not_forbidden_by_policy) {
>>>>     gather_temporary_ipv6_addresses();
>>>>   } else {
>>>>     gather_non_temporary_ipv6_addresses();
>>>>  }
>>>> }
>>>>
>>>>
>>>  What should be done when temporary enabled only on some of the network
>>> interfaces of the device, i.e. if, for instance, WiFI interface has only
>>> non temp ipv6 address and LTE has both temp and permanent address present?
>>>
>>>
>> Is this a real-world problem? As I understand it, temporary addresses are
>> assigned by the host, so you either support them or you don't.
>>
>
> On Linux you can enable temporary addresses per interface, so it is
> possible.
>
> The whole problem (with using temp or permanent addresses) is a bit
> imaginary since under most common client setups you only see temporary
> addresses. Permanent IPv6 addresses show up only on servers or if
> specifically configured on the host.
>

It's actually not imaginary in enterprise contexts, as there are shops
that disable temporary addresses to make tracking or other security
activities easier.  Not my favorite reasoning, personally, but there you go.

Ted





> _____________
> Roman Shpount
>
>

--089e0158aa1266176e04f520159b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Mar 20, 2014 at 7:39 PM=
, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" =
target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<br><div class=3D"=
gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div><div class=3D"h5"><div class=3D"gmail_quote">On Thu, Mar 20, 2014 at 8=
:28 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@googl=
e.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">

<div><div>On Thu, Mar 20, 2014 at 4:17 PM, Roman Shpount <span dir=3D"ltr">=
&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.co=
m</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"=
gmail_quote">

<div>On Thu, Mar 20, 2014 at 7:03 PM, Justin Uberti <span dir=3D"ltr">&lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr">Your take is what I had in mind. Basicall=
y a ruleset like this:<div>




<br></div><div>=A0gather_ipv4_addresses();<br></div><div>=A0if (has_ipv6) {=
</div><div>=A0 if (has_temporary_addresses &amp;&amp; temporaries_not_forbi=
dden_by_policy) {</div>

<div>=A0 =A0 gather_temporary_ipv6_addresses();</div><div>=A0 } else {</div=
><div>=A0 =A0 gather_non_temporary_ipv6_addresses();</div><div>=A0}</div><d=
iv>}=A0</div></div><div class=3D"gmail_extra"><br></div></blockquote><div><=
br></div></div>



<div>
What should be done when temporary enabled only on some of the network inte=
rfaces of the device, i.e. if, for instance, WiFI interface has only non te=
mp ipv6 address and LTE has both temp and permanent address present?</div>




<div><span><font color=3D"#888888"><br></font></span></div></div></div></di=
v></blockquote><div>=A0</div></div></div><div>Is this a real-world problem?=
 As I understand it, temporary addresses are assigned by the host, so you e=
ither support them or you don&#39;t.</div>



</div></div></div>
</blockquote></div><div><br></div></div></div><div>On Linux you can enable =
temporary addresses per interface, so it is possible.</div><div><br></div><=
div>The whole problem (with using temp or permanent addresses) is a bit ima=
ginary since under most common client setups you only see temporary address=
es. Permanent IPv6 addresses show up only on servers or if specifically con=
figured on the host.</div>

<div></div></div></div></blockquote><div><br><div class=3D"gmail_default" s=
tyle=3D"font-family:georgia,serif">It&#39;s actually not imaginary in enter=
prise contexts, as there are shops that disable temporary addresses to make=
 tracking or other security activities easier.=A0 Not my favorite reasoning=
, personally, but there you go.<br>
<br>Ted<br></div><br><br><br>=A0</div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div d=
ir=3D"ltr"><div class=3D"gmail_extra"><div>_____________<span class=3D"HOEn=
Zb"><font color=3D"#888888"><br>
Roman Shpount</font></span></div><div><br></div></div></div>
</blockquote></div><br></div></div>

--089e0158aa1266176e04f520159b--


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On Fri, Mar 21, 2014 at 12:07 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <roman@telurix.com> wrote:
>
>> The whole problem (with using temp or permanent addresses) is a bit
>> imaginary since under most common client setups you only see temporary
>> addresses. Permanent IPv6 addresses show up only on servers or if
>> specifically configured on the host.
>>
>
> It's actually not imaginary in enterprise contexts, as there are shops
> that disable temporary addresses to make tracking or other security
> activities easier.  Not my favorite reasoning, personally, but there you go.
>

What I meant is situation where you have both temp and permanent addresses
on the same interface is quite unusual. In most cases it is one or the
other. Getting both addresses configured usually requires some very
deliberate administrative action, and unless I am mistaken, is not common.
_____________
Roman Shpount

--f46d043892139242de04f520f539
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Fri, Mar 21, 2014 at 12:07 PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=
=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;=
</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div><div clas=
s=3D"h5">
On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;<=
/span> wrote:<br></div></div><div class=3D"gmail_quote"><div><div class=3D"=
h5">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div><div><div=
 class=3D"gmail_quote">
The whole problem (with using temp or permanent addresses) is a bit imagina=
ry since under most common client setups you only see temporary addresses. =
Permanent IPv6 addresses show up only on servers or if specifically configu=
red on the host.<br>
</div></div></div>

<div></div></div></div></blockquote></div></div><div><br><div style=3D"font=
-family:georgia,serif">It&#39;s actually not imaginary in enterprise contex=
ts, as there are shops that disable temporary addresses to make tracking or=
 other security activities easier.=A0 Not my favorite reasoning, personally=
, but there you go.<br>
</div></div></div></div></div></blockquote><div><br></div>What I meant is s=
ituation where you have both temp and permanent addresses on the same inter=
face is quite unusual. In most cases it is one or the other. Getting both a=
ddresses configured usually requires some very deliberate administrative ac=
tion, and unless I am mistaken, is not common.<div>
_____________<br>Roman Shpount</div><div>=A0</div></div></div></div>

--f46d043892139242de04f520f539--


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From: Dmitry Anipko <Dmitry.Anipko@microsoft.com>
To: Roman Shpount <roman@telurix.com>, Ted Hardie <ted.ietf@gmail.com>
Thread-Topic: [rtcweb] Transports: RFC 4941 support?
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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>>What I meant is situation where you have both temp and permanent addresse=
s on the same interface


This is a default for interfaces of Windows computers, using SLAAC.


________________________________
From: rtcweb <rtcweb-bounces@ietf.org> on behalf of Roman Shpount <roman@te=
lurix.com>
Sent: Friday, March 21, 2014 10:10 AM
To: Ted Hardie
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Transports: RFC 4941 support?


On Fri, Mar 21, 2014 at 12:07 PM, Ted Hardie <ted.ietf@gmail.com<mailto:ted=
.ietf@gmail.com>> wrote:
On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <roman@telurix.com<mailto:ro=
man@telurix.com>> wrote:
The whole problem (with using temp or permanent addresses) is a bit imagina=
ry since under most common client setups you only see temporary addresses. =
Permanent IPv6 addresses show up only on servers or if specifically configu=
red on the host.

It's actually not imaginary in enterprise contexts, as there are shops that=
 disable temporary addresses to make tracking or other security activities =
easier.  Not my favorite reasoning, personally, but there you go.

What I meant is situation where you have both temp and permanent addresses =
on the same interface is quite unusual. In most cases it is one or the othe=
r. Getting both addresses configured usually requires some very deliberate =
administrative action, and unless I am mistaken, is not common.
_____________
Roman Shpount


--_000_adf8e32e91564028bc97623b7dd6e938SN2PR03MB077namprd03pro_
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<div style=3D"font-size:12pt;color:#000000;background-color:#FFFFFF;font-fa=
mily:Calibri,Arial,Helvetica,sans-serif;">
<p>&gt;&gt;<span style=3D"color: #282828; font-family: calibri, arial, helv=
etica, sans-serif; font-size: 16px; background-color: #ffffff;">What I mean=
t is situation where you have both temp and permanent addresses on the same=
 interface&nbsp;</span></p>
<p><br>
</p>
<p>This is a default for interfaces of Windows computers, using SLAAC.<br>
</p>
<p><br>
</p>
<div style=3D"color: #282828;">
<hr tabindex=3D"-1" style=3D"display: inline-block; width: 98%;">
<div id=3D"divRplyFwdMsg" dir=3D"ltr"><font face=3D"Calibri, sans-serif" co=
lor=3D"#000000" style=3D"font-size: 11pt;"><b>From:</b> rtcweb &lt;rtcweb-b=
ounces@ietf.org&gt; on behalf of Roman Shpount &lt;roman@telurix.com&gt;<br=
>
<b>Sent:</b> Friday, March 21, 2014 10:10 AM<br>
<b>To:</b> Ted Hardie<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Transports: RFC 4941 support?</font>
<div>&nbsp;</div>
</div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Fri, Mar 21, 2014 at 12:07 PM, Ted Hardie <sp=
an dir=3D"ltr">
&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.=
com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; borde=
r-left-width: 1px; border-left-color: #cccccc; border-left-style: solid; pa=
dding-left: 1ex;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div>
<div class=3D"h5">On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <span dir=
=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@t=
elurix.com</a>&gt;</span> wrote:<br>
</div>
</div>
<div class=3D"gmail_quote">
<div>
<div class=3D"h5">
<blockquote class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; borde=
r-left-width: 1px; border-left-color: #cccccc; border-left-style: solid; pa=
dding-left: 1ex;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div>
<div>
<div class=3D"gmail_quote">The whole problem (with using temp or permanent =
addresses) is a bit imaginary since under most common client setups you onl=
y see temporary addresses. Permanent IPv6 addresses show up only on servers=
 or if specifically configured on
 the host.<br>
</div>
</div>
</div>
<div></div>
</div>
</div>
</blockquote>
</div>
</div>
<div><br>
<div style=3D"font-family: georgia, serif;">It's actually not imaginary in =
enterprise contexts, as there are shops that disable temporary addresses to=
 make tracking or other security activities easier.&nbsp; Not my favorite r=
easoning, personally, but there you go.<br>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
What I meant is situation where you have both temp and permanent addresses =
on the same interface is quite unusual. In most cases it is one or the othe=
r. Getting both addresses configured usually requires some very deliberate =
administrative action, and unless
 I am mistaken, is not common.
<div>_____________<br>
Roman Shpount</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From: Martin Thomson <martin.thomson@gmail.com>
To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On 21 March 2014 09:07, Ted Hardie <ted.ietf@gmail.com> wrote:
> It's actually not imaginary in enterprise contexts, as there are shops that
> disable temporary addresses to make tracking or other security activities
> easier.  Not my favorite reasoning, personally, but there you go.

Yeah, that's what super-cookies are for.


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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On 03/21/2014 06:10 PM, Roman Shpount wrote:
>
> On Fri, Mar 21, 2014 at 12:07 PM, Ted Hardie <ted.ietf@gmail.com
> <mailto:ted.ietf@gmail.com>> wrote:
>
>     On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <roman@telurix.com
>     <mailto:roman@telurix.com>> wrote:
>
>         The whole problem (with using temp or permanent addresses) is
>         a bit imaginary since under most common client setups you only
>         see temporary addresses. Permanent IPv6 addresses show up only
>         on servers or if specifically configured on the host.
>
>
>     It's actually not imaginary in enterprise contexts, as there are
>     shops that disable temporary addresses to make tracking or other
>     security activities easier.  Not my favorite reasoning,
>     personally, but there you go.
>
>
> What I meant is situation where you have both temp and permanent
> addresses on the same interface is quite unusual. In most cases it is
> one or the other. Getting both addresses configured usually requires
> some very deliberate administrative action, and unless I am mistaken,
> is not common.

I think you are mistaken..... at least, your claim does not fit with my
data.

This laptop, on a default-configured IPv6 home network:

    inet6 2001:470:de0a:27:9964:c162:8161:9a2e/64 scope global temporary
dynamic
       valid_lft 86356sec preferred_lft 14356sec
    inet6 2001:470:de0a:27:863a:4bff:fe0b:28b0/64 scope global dynamic
       valid_lft 86356sec preferred_lft 14356sec

Some very deliberate administrative action may have been taken (this is
a Google corporate device), but I suspect that it's a default Ubuntu
configuration.

> _____________
> Roman Shpount
>  
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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<html>
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    <div class="moz-cite-prefix">On 03/21/2014 06:10 PM, Roman Shpount
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAD5OKxtwH-rrkN5BA7zt9kFLC6+sTZAnGjBh+JvFc7FmYMoKVg@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra"><br>
          <div class="gmail_quote">On Fri, Mar 21, 2014 at 12:07 PM, Ted
            Hardie <span dir="ltr">&lt;<a moz-do-not-send="true"
                href="mailto:ted.ietf@gmail.com" target="_blank">ted.ietf@gmail.com</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
              <div dir="ltr">
                <div class="gmail_extra">
                  <div>
                    <div class="h5">
                      On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <span
                        dir="ltr">&lt;<a moz-do-not-send="true"
                          href="mailto:roman@telurix.com"
                          target="_blank">roman@telurix.com</a>&gt;</span>
                      wrote:<br>
                    </div>
                  </div>
                  <div class="gmail_quote">
                    <div>
                      <div class="h5">
                        <blockquote class="gmail_quote"
                          style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                          <div dir="ltr">
                            <div class="gmail_extra">
                              <div>
                                <div>
                                  <div class="gmail_quote">
                                    The whole problem (with using temp
                                    or permanent addresses) is a bit
                                    imaginary since under most common
                                    client setups you only see temporary
                                    addresses. Permanent IPv6 addresses
                                    show up only on servers or if
                                    specifically configured on the host.<br>
                                  </div>
                                </div>
                              </div>
                            </div>
                          </div>
                        </blockquote>
                      </div>
                    </div>
                    <div><br>
                      <div style="font-family:georgia,serif">It's
                        actually not imaginary in enterprise contexts,
                        as there are shops that disable temporary
                        addresses to make tracking or other security
                        activities easier.&nbsp; Not my favorite reasoning,
                        personally, but there you go.<br>
                      </div>
                    </div>
                  </div>
                </div>
              </div>
            </blockquote>
            <div><br>
            </div>
            What I meant is situation where you have both temp and
            permanent addresses on the same interface is quite unusual.
            In most cases it is one or the other. Getting both addresses
            configured usually requires some very deliberate
            administrative action, and unless I am mistaken, is not
            common.</div>
        </div>
      </div>
    </blockquote>
    <br>
    I think you are mistaken..... at least, your claim does not fit with
    my data.<br>
    <br>
    This laptop, on a default-configured IPv6 home network:<br>
    <br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:9964:c162:8161:9a2e/64 scope global
    temporary dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86356sec preferred_lft 14356sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:863a:4bff:fe0b:28b0/64 scope global
    dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86356sec preferred_lft 14356sec<br>
    <br>
    Some very deliberate administrative action may have been taken (this
    is a Google corporate device), but I suspect that it's a default
    Ubuntu configuration.<br>
    <br>
    <blockquote
cite="mid:CAD5OKxtwH-rrkN5BA7zt9kFLC6+sTZAnGjBh+JvFc7FmYMoKVg@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra">
          <div class="gmail_quote">
            <div>
              _____________<br>
              Roman Shpount</div>
            <div>&nbsp;</div>
          </div>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

--------------080606060707070906040509--


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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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On 03/22/2014 09:23 PM, Harald Alvestrand wrote:
> On 03/21/2014 06:10 PM, Roman Shpount wrote:
>>
>> On Fri, Mar 21, 2014 at 12:07 PM, Ted Hardie <ted.ietf@gmail.com 
>> <mailto:ted.ietf@gmail.com>> wrote:
>>
>>     On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <roman@telurix.com
>>     <mailto:roman@telurix.com>> wrote:
>>
>>         The whole problem (with using temp or permanent addresses) is
>>         a bit imaginary since under most common client setups you
>>         only see temporary addresses. Permanent IPv6 addresses show
>>         up only on servers or if specifically configured on the host.
>>
>>
>>     It's actually not imaginary in enterprise contexts, as there are
>>     shops that disable temporary addresses to make tracking or other
>>     security activities easier.  Not my favorite reasoning,
>>     personally, but there you go.
>>
>>
>> What I meant is situation where you have both temp and permanent 
>> addresses on the same interface is quite unusual. In most cases it is 
>> one or the other. Getting both addresses configured usually requires 
>> some very deliberate administrative action, and unless I am mistaken, 
>> is not common.
>
> I think you are mistaken..... at least, your claim does not fit with 
> my data.
>
> This laptop, on a default-configured IPv6 home network:
>
>     inet6 2001:470:de0a:27:9964:c162:8161:9a2e/64 scope global 
> temporary dynamic
>        valid_lft 86356sec preferred_lft 14356sec
>     inet6 2001:470:de0a:27:863a:4bff:fe0b:28b0/64 scope global dynamic
>        valid_lft 86356sec preferred_lft 14356sec
>
> Some very deliberate administrative action may have been taken (this 
> is a Google corporate device), but I suspect that it's a default 
> Ubuntu configuration.

And just to get people used to looking at stuff for themselves, here's 
my desktop machine, which is DEFINITELY a default Ubuntu setup:

hta@liset:~$ ip addr
1: lo: <LOOPBACK,UP,LOWER_UP> mtu 16436 qdisc noqueue state UNKNOWN
     link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
     inet 127.0.0.1/8 scope host lo
     inet6 ::1/128 scope host
        valid_lft forever preferred_lft forever
2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast 
state UP qlen 1000
     link/ether 6c:3b:e5:0b:6a:b4 brd ff:ff:ff:ff:ff:ff
     inet 192.168.1.17/24 brd 192.168.1.255 scope global eth0
     inet6 2001:470:de0a:27:d014:dc1:dd24:4393/64 scope global temporary 
dynamic
        valid_lft 86379sec preferred_lft 14379sec
     inet6 2001:470:de0a:27:31d8:d7fc:4af4:c17a/64 scope global 
temporary deprecated dynamic
        valid_lft 86379sec preferred_lft 0sec
     inet6 2001:470:de0a:27:8c26:e950:a09d:814c/64 scope global 
temporary deprecated dynamic
        valid_lft 86379sec preferred_lft 0sec
     inet6 2001:470:de0a:27:d883:fc90:cdcd:6b8f/64 scope global 
temporary deprecated dynamic
        valid_lft 86379sec preferred_lft 0sec
     inet6 2001:470:de0a:27:40c9:8eca:a92b:d0ac/64 scope global 
temporary deprecated dynamic
        valid_lft 86379sec preferred_lft 0sec
     inet6 2001:470:de0a:27:55e8:a88d:df95:df5d/64 scope global 
temporary deprecated dynamic
        valid_lft 86379sec preferred_lft 0sec
     inet6 2001:470:de0a:27:accc:2652:cc97:5b17/64 scope global 
temporary deprecated dynamic
        valid_lft 76189sec preferred_lft 0sec
     inet6 2001:470:de0a:27:6e3b:e5ff:fe0b:6ab4/64 scope global dynamic
        valid_lft 86379sec preferred_lft 14379sec
     inet6 fe80::6e3b:e5ff:fe0b:6ab4/64 scope link
        valid_lft forever preferred_lft forever

Ignoring the deprecated ones, one temporary and one non-temporary.


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    <div class="moz-cite-prefix">On 03/22/2014 09:23 PM, Harald
      Alvestrand wrote:<br>
    </div>
    <blockquote cite="mid:532DF135.1090509@alvestrand.no" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 03/21/2014 06:10 PM, Roman Shpount
        wrote:<br>
      </div>
      <blockquote
cite="mid:CAD5OKxtwH-rrkN5BA7zt9kFLC6+sTZAnGjBh+JvFc7FmYMoKVg@mail.gmail.com"
        type="cite">
        <div dir="ltr">
          <div class="gmail_extra"><br>
            <div class="gmail_quote">On Fri, Mar 21, 2014 at 12:07 PM,
              Ted Hardie <span dir="ltr">&lt;<a moz-do-not-send="true"
                  href="mailto:ted.ietf@gmail.com" target="_blank">ted.ietf@gmail.com</a>&gt;</span>
              wrote:<br>
              <blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                <div dir="ltr">
                  <div class="gmail_extra">
                    <div>
                      <div class="h5"> On Thu, Mar 20, 2014 at 7:39 PM,
                        Roman Shpount <span dir="ltr">&lt;<a
                            moz-do-not-send="true"
                            href="mailto:roman@telurix.com"
                            target="_blank">roman@telurix.com</a>&gt;</span>
                        wrote:<br>
                      </div>
                    </div>
                    <div class="gmail_quote">
                      <div>
                        <div class="h5">
                          <blockquote class="gmail_quote"
                            style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                            <div dir="ltr">
                              <div class="gmail_extra">
                                <div>
                                  <div>
                                    <div class="gmail_quote"> The whole
                                      problem (with using temp or
                                      permanent addresses) is a bit
                                      imaginary since under most common
                                      client setups you only see
                                      temporary addresses. Permanent
                                      IPv6 addresses show up only on
                                      servers or if specifically
                                      configured on the host.<br>
                                    </div>
                                  </div>
                                </div>
                              </div>
                            </div>
                          </blockquote>
                        </div>
                      </div>
                      <div><br>
                        <div style="font-family:georgia,serif">It's
                          actually not imaginary in enterprise contexts,
                          as there are shops that disable temporary
                          addresses to make tracking or other security
                          activities easier.&nbsp; Not my favorite reasoning,
                          personally, but there you go.<br>
                        </div>
                      </div>
                    </div>
                  </div>
                </div>
              </blockquote>
              <div><br>
              </div>
              What I meant is situation where you have both temp and
              permanent addresses on the same interface is quite
              unusual. In most cases it is one or the other. Getting
              both addresses configured usually requires some very
              deliberate administrative action, and unless I am
              mistaken, is not common.</div>
          </div>
        </div>
      </blockquote>
      <br>
      I think you are mistaken..... at least, your claim does not fit
      with my data.<br>
      <br>
      This laptop, on a default-configured IPv6 home network:<br>
      <br>
      &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:9964:c162:8161:9a2e/64 scope global
      temporary dynamic <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86356sec preferred_lft 14356sec<br>
      &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:863a:4bff:fe0b:28b0/64 scope global
      dynamic <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86356sec preferred_lft 14356sec<br>
      <br>
      Some very deliberate administrative action may have been taken
      (this is a Google corporate device), but I suspect that it's a
      default Ubuntu configuration.<br>
    </blockquote>
    <br>
    And just to get people used to looking at stuff for themselves,
    here's my desktop machine, which is DEFINITELY a default Ubuntu
    setup:<br>
    <br>
    hta@liset:~$ ip addr<br>
    1: lo: &lt;LOOPBACK,UP,LOWER_UP&gt; mtu 16436 qdisc noqueue state
    UNKNOWN <br>
    &nbsp;&nbsp;&nbsp; link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00<br>
    &nbsp;&nbsp;&nbsp; inet 127.0.0.1/8 scope host lo<br>
    &nbsp;&nbsp;&nbsp; inet6 ::1/128 scope host <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft forever preferred_lft forever<br>
    2: eth0: &lt;BROADCAST,MULTICAST,UP,LOWER_UP&gt; mtu 1500 qdisc
    pfifo_fast state UP qlen 1000<br>
    &nbsp;&nbsp;&nbsp; link/ether 6c:3b:e5:0b:6a:b4 brd ff:ff:ff:ff:ff:ff<br>
    &nbsp;&nbsp;&nbsp; inet 192.168.1.17/24 brd 192.168.1.255 scope global eth0<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:d014:dc1:dd24:4393/64 scope global
    temporary dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86379sec preferred_lft 14379sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:31d8:d7fc:4af4:c17a/64 scope global
    temporary deprecated dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86379sec preferred_lft 0sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:8c26:e950:a09d:814c/64 scope global
    temporary deprecated dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86379sec preferred_lft 0sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:d883:fc90:cdcd:6b8f/64 scope global
    temporary deprecated dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86379sec preferred_lft 0sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:40c9:8eca:a92b:d0ac/64 scope global
    temporary deprecated dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86379sec preferred_lft 0sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:55e8:a88d:df95:df5d/64 scope global
    temporary deprecated dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86379sec preferred_lft 0sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:accc:2652:cc97:5b17/64 scope global
    temporary deprecated dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 76189sec preferred_lft 0sec<br>
    &nbsp;&nbsp;&nbsp; inet6 2001:470:de0a:27:6e3b:e5ff:fe0b:6ab4/64 scope global
    dynamic <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft 86379sec preferred_lft 14379sec<br>
    &nbsp;&nbsp;&nbsp; inet6 fe80::6e3b:e5ff:fe0b:6ab4/64 scope link <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valid_lft forever preferred_lft forever<br>
    <br>
    Ignoring the deprecated ones, one temporary and one non-temporary.<br>
    <br>
  </body>
</html>

--------------070101020007090806080504--


From nobody Sat Mar 22 18:39:40 2014
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] On ptimes and maxptimes
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On Mar 21, 2014, at 2:19 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> On 2014-03-20 17:39, Justin Uberti wrote:
>>=20
>>=20
>>=20
>> On Thu, Mar 20, 2014 at 3:53 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com =
<mailto:magnus.westerlund@ericsson.com>>
>> wrote:
>>=20
>>    On 2014-03-20 01:27, Justin Uberti wrote:
>>>=20
>>>=20
>>>=20
>>> On Mon, Mar 10, 2014 at 8:57 AM, Magnus Westerlund
>>> <magnus.westerlund@ericsson.com
>>    <mailto:magnus.westerlund@ericsson.com>
>>    <mailto:magnus.westerlund@ericsson.com
>>    <mailto:magnus.westerlund@ericsson.com>>>
>>> wrote:
>>>=20
>>>    Hi,
>>>=20
>>>    (As individual)
>>>=20
>>>    On 2014-03-05 17:26, Justin Uberti wrote:
>>>> =46rom the session this morning, I recall this consensus:
>>>>=20
>>>>  * We will add recommendations on the frame sizes to use for
>>>    PCMU/A to
>>>>    the rtcweb audio codec I-D (20, 30, 60 ms frames).
>>>=20
>>>    The above applies to sending payloads
>>>=20
>>>=20
>>> Right, thanks for pointing that out.
>>>=20
>>>=20
>>>    Yes, but that was given that a receiver will be capable of
>>    receiving any
>>>    valid number of frames, or samples within a reasonable
>>    packetization,
>>>    and I think that should be 120 ms.
>>>=20
>>>>  * We will consider adding an a=3Dmaxptime attribute that
>>>    represents the
>>>>    "minimum maxptime" of all the offered codecs. That is,
>>    if both
>>>    Opus
>>>>    (maxptime of 120ms) and PCMU (maxptime of 60) are offered, a
>>>>    maxptime=3D60 SHOULD be included.
>>>>      o Since maxptime spans all codecs, this means that
>>    some modes
>>>>        (e.g. Opus 120ms) will be unavailable, unless only
>>    Opus is
>>>    offered.
>>>=20
>>>    I think you base the assumption on maxptime based on the sending
>>>    recommendations, not what is capable of receiving. If one is
>>    capable of
>>>    receiving 200 ms of a-law and Opus, then you would need to say
>>    120 ms as
>>>    Opus will limit you.
>>>=20
>>>=20
>>> I agree that Opus' maxptime is 120, but should that prevent
>>    sending PCMU
>>> with 200 ms? That is, since sending Opus with 200 ms is not legal, =
is
>>> there any point in forcing WebRTC impls to indicate this in SDP?
>>=20
>>    What I wished we fixed this issue with maxptime a long time ago. =
This is
>>    a well known problem that it really has PT scope, but are =
signalled on
>>    m=3D line scope.
>>=20
>>>=20
>>>=20
>>>=20
>>>>  * We will consider adding an a=3Dptime attribute, set to the
>>>    receiver's
>>>>    preferred frame size to receive. Again, this value spans all
>>>    codecs,
>>>>    so the specified value may not be viable for all codecs
>>    (e.g.
>>>    30 ms
>>>>    works for PCMU but not for Opus)
>>>>=20
>>>> After thinking about this some, I'm not sure the
>>    maxptime/ptime values
>>>> will lead to any different behavior than if they were not
>>    specified.
>>>> Since there is a complexity cost (especially if the values
>>    need to
>>>> change based on which codecs are offered), is there a clear
>>    upside
>>>    here?
>>>=20
>>>    The maxptime values may in some cases be limited downwards for
>>>    interoperability, for example if one like to gateway AMR into a =
CS
>>>    network then the gateway might indicate a maxptime that is
>>    less, more
>>>    like 20 or 40 ms.
>>>=20
>>>=20
>>> Sure - WebRTC impls need to respect any maxptime value that they
>>> receive, but it's not clear to me that they need to include one in =
the
>>> SDP they send.
>>=20
>>    Ok, but then we are back to my previous argument for writing down =
what
>>    receiver requirement in regards to each payload type one is =
required to
>>    support in the audio draft for the ones specified there. But what =
about
>>    the other codecs that aren't included in the audio draft?
>>=20
>>    How can a peer know if there are limitations? I wouldn't attempt =
to
>>    solve the general problem of the scope issue with maxptime. Simply =
state
>>    that you should include this. It is going to be limited but still
>>    provide information in many cases that is quite useful for the =
sender.
>>=20
>>=20
>> Just to be sure I understand, your position is that WebRTC impls =
SHOULD
>> include an a=3Dmaxptime attribute indicating the "minimum maximum" =
frame
>> size they can receive, across all codecs?
>=20
> Yes, as long as we lack payload type specific, I think that is the =
best
> we can do.
>>=20
>> Do you think anything needs to be included regarding a=3Dptime?
>=20
> I think it MAY be included to indicate an initial preference for
> packetization interval. In cases that is not common across the set of
> payload types for that m=3D section, then one should skip it.
>=20
> Cheers
>=20
> Magnus Westerlund


It seems like from interoperability point of view, we also need,  if a =
browser receives a maxptime or prime, it MUST be able to understand =
that.=20


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Date: Sat, 22 Mar 2014 21:46:34 -0400
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From: Roman Shpount <roman@telurix.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Transports: RFC 4941 support?
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--90e6ba309486f5ec9e04f53c4916
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I was incorrect. It is normal to get both temp and permanent IPv6 addresses
on Windows and Ubuntu.

_____________
Roman Shpount


On Sat, Mar 22, 2014 at 5:07 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  On 03/22/2014 09:23 PM, Harald Alvestrand wrote:
>
> On 03/21/2014 06:10 PM, Roman Shpount wrote:
>
>
> On Fri, Mar 21, 2014 at 12:07 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
>
>>   On Thu, Mar 20, 2014 at 7:39 PM, Roman Shpount <roman@telurix.com>wrote:
>>
>>>    The whole problem (with using temp or permanent addresses) is a bit
>>> imaginary since under most common client setups you only see temporary
>>> addresses. Permanent IPv6 addresses show up only on servers or if
>>> specifically configured on the host.
>>>
>>
>> It's actually not imaginary in enterprise contexts, as there are shops
>> that disable temporary addresses to make tracking or other security
>> activities easier.  Not my favorite reasoning, personally, but there you go.
>>
>
>  What I meant is situation where you have both temp and permanent
> addresses on the same interface is quite unusual. In most cases it is one
> or the other. Getting both addresses configured usually requires some very
> deliberate administrative action, and unless I am mistaken, is not common.
>
>
> I think you are mistaken..... at least, your claim does not fit with my
> data.
>
> This laptop, on a default-configured IPv6 home network:
>
>     inet6 2001:470:de0a:27:9964:c162:8161:9a2e/64 scope global temporary
> dynamic
>        valid_lft 86356sec preferred_lft 14356sec
>     inet6 2001:470:de0a:27:863a:4bff:fe0b:28b0/64 scope global dynamic
>        valid_lft 86356sec preferred_lft 14356sec
>
> Some very deliberate administrative action may have been taken (this is a
> Google corporate device), but I suspect that it's a default Ubuntu
> configuration.
>
>
> And just to get people used to looking at stuff for themselves, here's my
> desktop machine, which is DEFINITELY a default Ubuntu setup:
>
> hta@liset:~$ ip addr
> 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 16436 qdisc noqueue state UNKNOWN
>     link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
>     inet 127.0.0.1/8 scope host lo
>     inet6 ::1/128 scope host
>        valid_lft forever preferred_lft forever
> 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state
> UP qlen 1000
>     link/ether 6c:3b:e5:0b:6a:b4 brd ff:ff:ff:ff:ff:ff
>     inet 192.168.1.17/24 brd 192.168.1.255 scope global eth0
>     inet6 2001:470:de0a:27:d014:dc1:dd24:4393/64 scope global temporary
> dynamic
>        valid_lft 86379sec preferred_lft 14379sec
>     inet6 2001:470:de0a:27:31d8:d7fc:4af4:c17a/64 scope global temporary
> deprecated dynamic
>        valid_lft 86379sec preferred_lft 0sec
>     inet6 2001:470:de0a:27:8c26:e950:a09d:814c/64 scope global temporary
> deprecated dynamic
>        valid_lft 86379sec preferred_lft 0sec
>     inet6 2001:470:de0a:27:d883:fc90:cdcd:6b8f/64 scope global temporary
> deprecated dynamic
>        valid_lft 86379sec preferred_lft 0sec
>     inet6 2001:470:de0a:27:40c9:8eca:a92b:d0ac/64 scope global temporary
> deprecated dynamic
>        valid_lft 86379sec preferred_lft 0sec
>     inet6 2001:470:de0a:27:55e8:a88d:df95:df5d/64 scope global temporary
> deprecated dynamic
>        valid_lft 86379sec preferred_lft 0sec
>     inet6 2001:470:de0a:27:accc:2652:cc97:5b17/64 scope global temporary
> deprecated dynamic
>        valid_lft 76189sec preferred_lft 0sec
>     inet6 2001:470:de0a:27:6e3b:e5ff:fe0b:6ab4/64 scope global dynamic
>        valid_lft 86379sec preferred_lft 14379sec
>     inet6 fe80::6e3b:e5ff:fe0b:6ab4/64 scope link
>        valid_lft forever preferred_lft forever
>
> Ignoring the deprecated ones, one temporary and one non-temporary.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--90e6ba309486f5ec9e04f53c4916
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<div dir=3D"ltr">I was incorrect. It is normal to get both temp and permane=
nt IPv6 addresses on Windows and Ubuntu.</div><div class=3D"gmail_extra"><b=
r clear=3D"all"><div>_____________<br>Roman Shpount</div>
<br><br><div class=3D"gmail_quote">On Sat, Mar 22, 2014 at 5:07 PM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D"">
    <div>On 03/22/2014 09:23 PM, Harald
      Alvestrand wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <div>On 03/21/2014 06:10 PM, Roman Shpount
        wrote:<br>
      </div>
      <blockquote type=3D"cite">
        <div dir=3D"ltr">
          <div class=3D"gmail_extra"><br>
            <div class=3D"gmail_quote">On Fri, Mar 21, 2014 at 12:07 PM,
              Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@g=
mail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span>
              wrote:<br>
              <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
                <div dir=3D"ltr">
                  <div class=3D"gmail_extra">
                    <div>
                      <div> On Thu, Mar 20, 2014 at 7:39 PM,
                        Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mail=
to:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span>
                        wrote:<br>
                      </div>
                    </div>
                    <div class=3D"gmail_quote">
                      <div>
                        <div>
                          <blockquote class=3D"gmail_quote" style=3D"margin=
:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204)=
;border-left-style:solid;padding-left:1ex">
                            <div dir=3D"ltr">
                              <div class=3D"gmail_extra">
                                <div>
                                  <div>
                                    <div class=3D"gmail_quote"> The whole
                                      problem (with using temp or
                                      permanent addresses) is a bit
                                      imaginary since under most common
                                      client setups you only see
                                      temporary addresses. Permanent
                                      IPv6 addresses show up only on
                                      servers or if specifically
                                      configured on the host.<br>
                                    </div>
                                  </div>
                                </div>
                              </div>
                            </div>
                          </blockquote>
                        </div>
                      </div>
                      <div><br>
                        <div style=3D"font-family:georgia,serif">It&#39;s
                          actually not imaginary in enterprise contexts,
                          as there are shops that disable temporary
                          addresses to make tracking or other security
                          activities easier.=A0 Not my favorite reasoning,
                          personally, but there you go.<br>
                        </div>
                      </div>
                    </div>
                  </div>
                </div>
              </blockquote>
              <div><br>
              </div>
              What I meant is situation where you have both temp and
              permanent addresses on the same interface is quite
              unusual. In most cases it is one or the other. Getting
              both addresses configured usually requires some very
              deliberate administrative action, and unless I am
              mistaken, is not common.</div>
          </div>
        </div>
      </blockquote>
      <br>
      I think you are mistaken..... at least, your claim does not fit
      with my data.<br>
      <br>
      This laptop, on a default-configured IPv6 home network:<br>
      <br>
      =A0=A0=A0 inet6 2001:470:de0a:27:9964:c162:8161:9a2e/64 scope global
      temporary dynamic <br>
      =A0=A0=A0=A0=A0=A0 valid_lft 86356sec preferred_lft 14356sec<br>
      =A0=A0=A0 inet6 2001:470:de0a:27:863a:4bff:fe0b:28b0/64 scope global
      dynamic <br>
      =A0=A0=A0=A0=A0=A0 valid_lft 86356sec preferred_lft 14356sec<br>
      <br>
      Some very deliberate administrative action may have been taken
      (this is a Google corporate device), but I suspect that it&#39;s a
      default Ubuntu configuration.<br>
    </blockquote>
    <br></div>
    And just to get people used to looking at stuff for themselves,
    here&#39;s my desktop machine, which is DEFINITELY a default Ubuntu
    setup:<br>
    <br>
    hta@liset:~$ ip addr<br>
    1: lo: &lt;LOOPBACK,UP,LOWER_UP&gt; mtu 16436 qdisc noqueue state
    UNKNOWN <br>
    =A0=A0=A0 link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00<br>
    =A0=A0=A0 inet <a href=3D"http://127.0.0.1/8" target=3D"_blank">127.0.0=
.1/8</a> scope host lo<br>
    =A0=A0=A0 inet6 ::1/128 scope host <br>
    =A0=A0=A0=A0=A0=A0 valid_lft forever preferred_lft forever<br>
    2: eth0: &lt;BROADCAST,MULTICAST,UP,LOWER_UP&gt; mtu 1500 qdisc
    pfifo_fast state UP qlen 1000<br>
    =A0=A0=A0 link/ether 6c:3b:e5:0b:6a:b4 brd ff:ff:ff:ff:ff:ff<br>
    =A0=A0=A0 inet <a href=3D"http://192.168.1.17/24" target=3D"_blank">192=
.168.1.17/24</a> brd 192.168.1.255 scope global eth0<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:d014:dc1:dd24:4393/64 scope global
    temporary dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 86379sec preferred_lft 14379sec<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:31d8:d7fc:4af4:c17a/64 scope global
    temporary deprecated dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 86379sec preferred_lft 0sec<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:8c26:e950:a09d:814c/64 scope global
    temporary deprecated dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 86379sec preferred_lft 0sec<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:d883:fc90:cdcd:6b8f/64 scope global
    temporary deprecated dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 86379sec preferred_lft 0sec<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:40c9:8eca:a92b:d0ac/64 scope global
    temporary deprecated dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 86379sec preferred_lft 0sec<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:55e8:a88d:df95:df5d/64 scope global
    temporary deprecated dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 86379sec preferred_lft 0sec<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:accc:2652:cc97:5b17/64 scope global
    temporary deprecated dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 76189sec preferred_lft 0sec<br>
    =A0=A0=A0 inet6 2001:470:de0a:27:6e3b:e5ff:fe0b:6ab4/64 scope global
    dynamic <br>
    =A0=A0=A0=A0=A0=A0 valid_lft 86379sec preferred_lft 14379sec<br>
    =A0=A0=A0 inet6 fe80::6e3b:e5ff:fe0b:6ab4/64 scope link <br>
    =A0=A0=A0=A0=A0=A0 valid_lft forever preferred_lft forever<br>
    <br>
    Ignoring the deprecated ones, one temporary and one non-temporary.<br>
    <br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--90e6ba309486f5ec9e04f53c4916--


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Subject: Re: [rtcweb] On ptimes and maxptimes
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On 3/22/2014 2:39 PM, Cullen Jennings wrote:
> It seems like from interoperability point of view, we also need, if a 
> browser receives a maxptime or prime, it MUST be able to understand that.

maxptime - yes, MUST
ptime - since it's merely a preference, it's not a MUST understand.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


From nobody Sun Mar 23 19:13:51 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : STUN Usage for Consent Freshness
        Authors         : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
	Filename        : draft-ietf-rtcweb-stun-consent-freshness-01.txt
	Pages           : 7
	Date            : 2014-03-23

Abstract:
   Verification of peer consent before sending traffic is necessary in
   WebRTC deployments to ensure that a malicious JavaScript cannot use
   the browser as a platform for launching attacks.  A related problem
   is session liveness.  WebRTC application may want to detect
   connection failure and take appropriate action.

   This document describes how a WebRTC browser can verify peer consent
   to continue sending traffic and detect connection failure.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshness-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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From nobody Tue Mar 25 08:18:11 2014
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Consent for fate-sharing connections  (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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Subject: [rtcweb] Consent for fate-sharing connections (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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Hi all,

Certain media streams(For example audio, video) may require multiple
components (RTP, RTCP) each of which has to work for the media stream as a
whole to work. If consent fails for one of the component then the
application(JS) may decide to cease transmission of that media stream.

Is there a need to describe some text (RECOMMENDATION) on how fate-sharing
connections need to be handled when consent fails for one of the
components in Consent draft ?
Or should this be left out of this draft ?

Regards,
Ram


From nobody Tue Mar 25 09:25:32 2014
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Subject: Re: [rtcweb] Consent for fate-sharing connections (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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--089e0149495e88e05504f570cb12
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On Tue, Mar 25, 2014 at 8:17 AM, Ram Mohan R (rmohanr) <rmohanr@cisco.com>wrote:

> Hi all,
>
> Certain media streams(For example audio, video) may require multiple
> components (RTP, RTCP) each of which has to work for the media stream as a
> whole to work. If consent fails for one of the component then the
> application(JS) may decide to cease transmission of that media stream.
>
> Is there a need to describe some text (RECOMMENDATION) on how fate-sharing
> connections need to be handled when consent fails for one of the
> components in Consent draft ?
> Or should this be left out of this draft ?
>
>
So, it's my understanding that the application may always decide to cease
transmission of a media stream even  when it has consent to continue, so
I'm not sure what language you are looking for here.  Can you be a bit more
direct on which section would need this recommendation and what it might
say?

Ted





> Regards,
> Ram
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e0149495e88e05504f570cb12
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Tue, Mar 25, 2014 at 8:17 AM=
, Ram Mohan R (rmohanr) <span dir=3D"ltr">&lt;<a href=3D"mailto:rmohanr@cis=
co.com" target=3D"_blank">rmohanr@cisco.com</a>&gt;</span> wrote:<br><div c=
lass=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi all,<br>
<br>
Certain media streams(For example audio, video) may require multiple<br>
components (RTP, RTCP) each of which has to work for the media stream as a<=
br>
whole to work. If consent fails for one of the component then the<br>
application(JS) may decide to cease transmission of that media stream.<br>
<br>
Is there a need to describe some text (RECOMMENDATION) on how fate-sharing<=
br>
connections need to be handled when consent fails for one of the<br>
components in Consent draft ?<br>
Or should this be left out of this draft ?<br>
<br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-family=
:georgia,serif">So, it&#39;s my understanding that the application may alwa=
ys decide to cease transmission of a media stream even=A0 when it has conse=
nt to continue, so I&#39;m not sure what language you are looking for here.=
=A0 Can you be a bit more direct on which section would need this recommend=
ation and what it might say?<br>
<br>Ted</div><br><br><br>=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Regards,<br>
Ram<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--089e0149495e88e05504f570cb12--


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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Consent for fate-sharing connections (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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Where RTP and RTCP are multiplexed on the same port, they also share the sam=
e fate with respect to ICE consent; similarly, when audio/video/data channel=
 are multiplexed, they also share consent fate. Where audio and video use di=
fferent ports, the fates will not be shared, and it is possible that consent=
 might be given to one and not the other, but it is probably best left up to=
 the application to decide how to react. The one case which might be deservi=
ng of a recommendation is where RTP and RTCP aren't multiplexed and consent i=
s given to one of RTP/RTCP but not the other.


> So, it's my understanding that the application may always decide to cease t=
ransmission of a media stream even  when it has consent to continue, so I'm n=
ot sure what language you are looking for here.  Can you be a bit more direc=
t on which section would need this recommendation and what it might say?
>=20
> Ted

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>Where RTP and RTCP are multiplexed on t=
he same port, they also share the same fate with respect to ICE consent; sim=
ilarly, when audio/video/data channel are multiplexed, they also share conse=
nt fate. Where audio and video use different ports, the fates will not be sh=
ared, and it is possible that consent might be given to one and not the othe=
r, but it is probably best left up to the application to decide how to react=
. The one case which might be deserving of a recommendation is where RTP and=
 RTCP aren't multiplexed and consent is given to one of RTP/RTCP but not the=
 other.</div><div><br></div><div><br></div><blockquote type=3D"cite"><div di=
r=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div class=3D=
"gmail_default" style=3D"font-family:georgia,serif">So, it's my understandin=
g that the application may always decide to cease transmission of a media st=
ream even&nbsp; when it has consent to continue, so I'm not sure what langua=
ge you are looking for here.&nbsp; Can you be a bit more direct on which sec=
tion would need this recommendation and what it might say?<br>
<br>Ted</div></div></div></div></blockquote></body></html>=

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From nobody Tue Mar 25 18:21:00 2014
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Ted Hardie <ted.ietf@gmail.com>
Thread-Topic: [rtcweb] Consent for fate-sharing connections (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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Subject: Re: [rtcweb] Consent for fate-sharing connections (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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From nobody Wed Mar 26 10:21:33 2014
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Subject: [rtcweb] Firewall/http-proxy traversal
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During the meeting in London I think Andy was scheduled to discuss
firewall and http proxy traversal
(http://tools.ietf.org/agenda/89/slides/slides-89-rtcweb-8.pdf) but
unfortunately we ran out of time.

Is there any plan to discuss this before our next meeting?

Thank you,
-Victor


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Subject: Re: [rtcweb] Consent for fate-sharing connections (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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On 2014-03-26 02:20, Ram Mohan R (rmohanr) wrote:

> Currently there is no section that talks about this. One of the comments
> given by Magnus some time back on this draft asks about scope of consent.
> We (the authors of draft-ietf-stun-consent-freshness) were discussing
> among ourselves on whether there is a need to have any text for
> fate-sharing connections. We donÂ¹t see a real need to have any
> recommendation in this draft but wanted to bring this topic to mailer to
> see if any one else have any different opinion.
> 
> As you said it is completely up to the application what to do if consent
> fails. There may be many possibilities- For example for a single media
> stream when RTP and RTCP are sent over different addresses (no mux used),
> then if consent fails for one of the components, an application may choose
> to cease that stream. There may be also cases where consent may fail for
> audio but pass for video. Again it is up to the application (JavaScript)
> to take a decision on what to do.

I do think RTP MUST be fate shared with its RTCP. Because revoking
consent for RTCP should not be a way of turning off congestion control.
Thus, no RTCP should mean no RTP either as you get no feedback on how it
behaves.
> 
> Since it is application that always decide, I am ok to just leave it and
> not have any text in this document
> 

>From a security point of view, what is important is that one can't treat
consent on one transport flow (5-tuple) to mean consent to send on any
other. Because then an attacker could create a session where he receives
the low-bitrate audio and consent to the media there and direct the
video to a target for the attack, where no consent is received.

But, I am personally mostly fine with saying nothing that a failure to
retain consent on one RTP session or data channel's transport flows, may
or may not mean that the transmission should be killed on another RTP
session or data channel, where consent has not yet failed.

One question is if one doesn't fate share all transport flows within the
peer connection if one should trigger a consent check when the failure
happens on some other transport flow? That way one can minimize the
delay of discovering revoked consent or path failure on all transport
flows.

Cheers

Magnus Westerlund

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Subject: Re: [rtcweb] On ptimes and maxptimes
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On 2014-03-23 07:12, Randell Jesup wrote:
> On 3/22/2014 2:39 PM, Cullen Jennings wrote:
>> It seems like from interoperability point of view, we also need, if a
>> browser receives a maxptime or prime, it MUST be able to understand that.
> 
> maxptime - yes, MUST
> ptime - since it's merely a preference, it's not a MUST understand.
> 

Yes, but I think it SHOULD be understood and followed.

Cheers

Magnus Westerlund

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Subject: Re: [rtcweb] Consent for fate-sharing connections (Do we need text in draft-ietf-rtcweb-stun-consent-freshness ?)
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On 27 March 2014 07:14, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> I do think RTP MUST be fate shared with its RTCP. Because revoking
> consent for RTCP should not be a way of turning off congestion control.
> Thus, no RTCP should mean no RTP either as you get no feedback on how it
> behaves.

I would hope that a congestion control algorithm for RTP would a) use
RTCP as its feedback mechanism, and b) react fairly harshly if RTCP
suddenly disappeared.  Such that loss of consent on RTCP would
essentially cut bandwidth to zero, or close to.

I remember having this issue with Lync: we were dropping RTCP and the
video quality was absolutely appalling.  Turns out, Lync was doing
both a and b.


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On 03/27/2014 06:15 PM, Martin Thomson wrote:
> On 27 March 2014 07:14, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> I do think RTP MUST be fate shared with its RTCP. Because revoking
>> consent for RTCP should not be a way of turning off congestion control.
>> Thus, no RTCP should mean no RTP either as you get no feedback on how it
>> behaves.
> I would hope that a congestion control algorithm for RTP would a) use
> RTCP as its feedback mechanism, and b) react fairly harshly if RTCP
> suddenly disappeared.  Such that loss of consent on RTCP would
> essentially cut bandwidth to zero, or close to.
>
> I remember having this issue with Lync: we were dropping RTCP and the
> video quality was absolutely appalling.  Turns out, Lync was doing
> both a and b.

That's one of the properties of the "circuit breaker" algorithm that 
Colin Perkins has been editing: Lose RTCP for 3 RTCP reporting 
intervals, cut transmission.

draft-ietf-avtcore-rtp-circuit-breakers-03.txt

And our very own rtp-usage requires that we support circuit-breaker:
draft-ietf-rtcweb-rtp-usage-11.txt

7.1.  Boundary Conditions and Circuit Breakers

    In the absence of a concrete congestion control algorithm, all WebRTC
    implementations MUST implement the RTP circuit breaker algorithm that
    is in described [I-D.ietf-avtcore-rtp-circuit-breakers].  The RTP
    circuit breaker is designed to enable applications to recognise and
    react to situations of extreme network congestion.  However, since
    the RTP circuit breaker might not be triggered until congestion
    becomes extreme, it cannot be considered a substitute for congestion
    control, and applications MUST also implement congestion control to
    allow them to adapt to changes in network capacity.  Any future RTP
    congestion control algorithms are expected to operate within the
    envelope allowed by the circuit breaker.



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On 2014-03-27 20:01, Harald Alvestrand wrote:
> On 03/27/2014 06:15 PM, Martin Thomson wrote:
>> On 27 March 2014 07:14, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:
>>> I do think RTP MUST be fate shared with its RTCP. Because revoking
>>> consent for RTCP should not be a way of turning off congestion control.
>>> Thus, no RTCP should mean no RTP either as you get no feedback on how it
>>> behaves.
>> I would hope that a congestion control algorithm for RTP would a) use
>> RTCP as its feedback mechanism, and b) react fairly harshly if RTCP
>> suddenly disappeared.  Such that loss of consent on RTCP would
>> essentially cut bandwidth to zero, or close to.
>>
>> I remember having this issue with Lync: we were dropping RTCP and the
>> video quality was absolutely appalling.  Turns out, Lync was doing
>> both a and b.
> 
> That's one of the properties of the "circuit breaker" algorithm that
> Colin Perkins has been editing: Lose RTCP for 3 RTCP reporting
> intervals, cut transmission.
> 
> draft-ietf-avtcore-rtp-circuit-breakers-03.txt
> 
> And our very own rtp-usage requires that we support circuit-breaker:
> draft-ietf-rtcweb-rtp-usage-11.txt
> 
> 7.1.  Boundary Conditions and Circuit Breakers
> 
>    In the absence of a concrete congestion control algorithm, all WebRTC
>    implementations MUST implement the RTP circuit breaker algorithm that
>    is in described [I-D.ietf-avtcore-rtp-circuit-breakers].  The RTP
>    circuit breaker is designed to enable applications to recognise and
>    react to situations of extreme network congestion.  However, since
>    the RTP circuit breaker might not be triggered until congestion
>    becomes extreme, it cannot be considered a substitute for congestion
>    control, and applications MUST also implement congestion control to
>    allow them to adapt to changes in network capacity.  Any future RTP
>    congestion control algorithms are expected to operate within the
>    envelope allowed by the circuit breaker.
> 

Martin and Harald,
(as individual)

Yes, you are both correct that we have mechanisms that also is intended
to react to this happening.

I didn't meant to imply that these would not be present. Still, I think
for the consent mechanism it needs to at least perform consent and kill
on what ICE refers to as "Media Stream", i.e. where RTP and RTCP
transport flows when not performing RTP/RTCP multiplexing would be
different components, that failure to acquire consent on all components
revoke the rights on all those components. Because, if not then you can
end up in strange conditions where you still send some traffic. If we
turn the issue around and consent is revoked for RTP but not RTCP. Then
the RTCP stream continues to send. An application may considered this a
failed case as this higher layer is likely to considered RTP and RTCP as
a unit and not care about the distinction.


Also, I realize that circuit breakers have an issue when it comes to
protecting the network if an attacker can set the RTCP bandwidths
sufficiently low on their own. As all the breakers are calculated based
on reporting intervals, if one sets the RTCP RR bandwidth to 1 bps, then
the reporting interval will be very long, even a two SSRC RTP session
with minimal RTCP packets will get an Td= 2730,67 s, i.e. ~45 minutes.

I need to think a bit more, but I think we might require that one
provision RTCP with sufficient RTCP bandwidth that the minimal fixed
interval of 5 seconds is going to be the largest term when using the
circuit breaker. I will bring this up in AVTCORE.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Sat Mar 29 08:24:33 2014
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There is a thread going on public-webrtc: WebRTC and backpressure (how 
to stop reading?)

That has gotten to the point where there is an rtcweb issue:

On 3/29/14 1:41 AM, Randell Jesup wrote:

>> No. You could continue to map  the channel messages byte-by-byte
>> on SCTP messages. But you would introduce some control message
>> (also in SCTP DATA chunks) which you would send reliably and
>> put in whatever information is needed. For me this looks pretty simple...
>
> Agreed; we have considerable latitude to define additional control
> messages in DATA chunks, if we decide this is useful.
> If we're careful, it may even be backwards compatible (ignored if not
> supported) without have to negotiate or probe for support.  But we'd
> need to know exactly *why* we'd consider changing things at this late
> date, and for what practical gain, given it would perhaps negatively
> impact the duck-typing between WebSockets and DataChannels and cause
> more delay and slow adoption.

PPID 50, defined in draft-ietf-rtcweb-data-protocol, certainly has the 
potential to be extended with additional control messages. (It has a one 
byte message type field with only two values defined.) That seems the 
obvious place where such messages could be defined.

Even if the control messages aren't to be defined now, it would be good 
to set the stage. For one thing, those control messages ought to be 
usable even for data channels that are established by means other than 
DCEP. Right now PPID 50 is only for DCEP. Also, as Randell implies 
above, it might be a good idea to define the extensibility mechanism - 
e.g., that messages bearing unknown message types be ignored, or that 
they get a specific response indicating that they are not supported.

	Thanks,
	Paul


From nobody Sat Mar 29 10:49:57 2014
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On 29 Mar 2014, at 16:24, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> There is a thread going on public-webrtc: WebRTC and backpressure (how =
to stop reading?)
>=20
> That has gotten to the point where there is an rtcweb issue:
>=20
> On 3/29/14 1:41 AM, Randell Jesup wrote:
>=20
>>> No. You could continue to map  the channel messages byte-by-byte
>>> on SCTP messages. But you would introduce some control message
>>> (also in SCTP DATA chunks) which you would send reliably and
>>> put in whatever information is needed. For me this looks pretty =
simple...
>>=20
>> Agreed; we have considerable latitude to define additional control
>> messages in DATA chunks, if we decide this is useful.
>> If we're careful, it may even be backwards compatible (ignored if not
>> supported) without have to negotiate or probe for support.  But we'd
>> need to know exactly *why* we'd consider changing things at this late
>> date, and for what practical gain, given it would perhaps negatively
>> impact the duck-typing between WebSockets and DataChannels and cause
>> more delay and slow adoption.
>=20
> PPID 50, defined in draft-ietf-rtcweb-data-protocol, certainly has the =
potential to be extended with additional control messages. (It has a one =
byte message type field with only two values defined.) That seems the =
obvious place where such messages could be defined.
DCEP is extensible in the way that you can add new message types. But it =
is a protocol for in-band negotiation
of a data channel. We should keep this focus.

The discussion at W3C focussed on messages not related to the =
establishment of a data channel, so I would
suggest to use a different PPID.
>=20
> Even if the control messages aren't to be defined now, it would be =
good to set the stage. For one thing, those control messages ought to be =
usable even for data channels that are established by means other than =
DCEP. Right now PPID 50 is only for DCEP. Also, as Randell implies =
above, it might be a good idea to define the extensibility mechanism - =
e.g., that messages bearing unknown message types be ignored, or that =
they get a specific response indicating that they are not supported.
I think at the last IETF we agreed to reset the channel. We can change =
that to ignore unknown PPIDs...

Best regards
Michael
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Sat Mar 29 11:47:47 2014
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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: Victor Pascual Avila <victor.pascual.avila@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Firewall/http-proxy traversal
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Hi Victor,

We can of course discuss on the mailing list(s) whenever you want.

If you want to discuss how to deal with proxies then the PNTAW list was set=
up to do this and probably any comments you have on draft-hutton-nat-firewa=
ll-considerations should also be on the PNTAW list.

But as Harald pointed out during the discussion on -transports- we do need =
to decide what to say if anything there as this current has a reference to =
-firewall- and discussion on this draft should I assume be on the RTCWEB li=
st.

The discussion on the PNTAW list stopped which I think was mostly because w=
e couldn't work out how to proceed but I for one would be happy if the disc=
ussion was restarted.

Regards
Andy





> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Victor
> Pascual Avila
> Sent: 26 March 2014 17:21
> To: rtcweb@ietf.org
> Subject: [rtcweb] Firewall/http-proxy traversal
>=20
> During the meeting in London I think Andy was scheduled to discuss
> firewall and http proxy traversal
> (http://tools.ietf.org/agenda/89/slides/slides-89-rtcweb-8.pdf) but
> unfortunately we ran out of time.
>=20
> Is there any plan to discuss this before our next meeting?
>=20
> Thank you,
> -Victor
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Reviving an old thread, since I'm getting the changes in:

On 02/25/2014 11:46 AM, Magnus Westerlund wrote:
> On 2014-02-19 19:47, Harald Alvestrand wrote:
>> On 02/19/2014 11:08 AM, Magnus Westerlund wrote:
>>> Hi,
>>>
>>> I have reviewed the Transports for RTCWEB
>>> (draft-ietf-rtcweb-transports-02)
>> Thanks!
>>>  and have the following comments.
>>>
>>> 1. Section 1.
>>>
>>>  This document focuses on the data
>>>    transport protocos that are used by conforming implementations.
>>>
>>> Missing "l" in protocos.
>> Fixed.
>>> 2. Section 3.1:
>>>    For both protocols, IPv4 and IPv6 support is assumed; applications
>>>    MUST be able to utilize both IPv4 and IPv6 where available.
>>>
>>> What "applications" are intended in this sentence. I get a feeling that
>>> "applications" here could be replaced by "WebRTC implementations"
>> Javascript applications. I think I should add the word "application" to
>> -overview's vocabulary; it is actually used in that meaning in that
>> document (also in the forms "Web application" and "browser-based
>> application".
> Sorry, then I find the above requirement a bit strangely targeted. I
> think we can require that IPv4 and IPv6 support for the applicable
> protocol pieces in what we define. But, on the JS application level?
> Also what explicit addressing information is visible on that level.

If two JS applications want to communicate, and the two systems are
connected with IPv4 but not connected with IPv6, communication should
happen, otherwise the implementation is non-conformant.

If the two systems are connected with IPv6 and not IPv4, communication
should happen, otherwise the implementation is non-conformant.

Communication, or the lack of it, is clearly a property that's
observable at the JS application level.

(the detail that the IPv* addresses are visible in the candidates and in
the SDP is an implementation detail that doesn't belong here.)

>
> It might be that this requirement needs to be split or at least apply to
> multiple levels.

Or maybe rewritten to a language that we both understand. Leaving it
roughly as-is for the moment.

>
>>>
>>> 3. Section 3.1:
>>>    For UDP, this specification assumes the ability to set the DSCP code
>>>    point of the sockets opened on a per-packet basis, in order to
>>>    achieve the prioritizations described in
>>>    [I-D.dhesikan-tsvwg-rtcweb-qos] when multiple media types are
>>>    multiplexed.  It does not assume that the DSCP codepoints will be
>>>    honored, and does assume that they may be zeroed or changed, since
>>>    this is a local configuration issue.
>>>
>>> I wonder if this should be moved into 3.2 section instead. At least the
>>> part in 3.1 should focus on the DSCP setting part on the transport flow.
>>> The implications of it working or not should be moved to 3.2. A forward
>>> pointer to that second can be used instead.
>> I think it fits better in 3.1 - this is about the assumptions that the
>> specifcation makes about the parts of the system that it is not a
>> specification for.
> I am fine with the first sentence staying, it is the second part that
> needs more discussion and I think it should be moved and expanded on.
> Thus a forward pointer may be appropriate here.

Added forward pointer, since I now have somewhere for it to point.
>>> 4. Order of Section 3.2 and 3.3.
>>>
>>> I think the order should be swapped between 3.2. and 3.3. That way the
>>> Multiplexing section can provide definitions of all the stream concepts
>>> that exists in WebRTC and how they can be combined. More about this later.
>>>
>>> 5. Content of Section 3.2
>>>
>>> This section needs to be expanded to start with a general discussion of
>>> how QoS can be applied to get a benefit. But, I think one need to be
>>> clear that it can't be assumed to be available in implementations or
>>> WebRTC based applications.
>> I want to push back on this. It seems inappropriate that the RTCWEB QoS
>> specification start off with a tutorial on what QoS is and how one can
>> use it to gain a benefit. That material should be available elsewhere.
>> (Recommendations?)
> I would like to agree with you, however we are putting together the
> pieces in a way that stands some of the previous assumptions on their
> heads. Thus, we don't have easy access to pointers.
>
> I can see this discussion going into
> draft-ietf-avtcore-multiplex-guidelines when it concerns RTP. However,
> you also have the data channel to consider. Thus, there exist no
> references that I know that discusses this particular combination and
> its implications.
>
>>> It needs to also discuss flow-based QoS mechanism. What are the
>>> implications if that is what is available and what choices does one have
>>> to address this by configuring the multiplexing different.
>> I do not want to discuss material for which there are no references.
>> What particular flow-based mechanisms do you have in mind, and what are
>> the relevant references?
> RSVP we can start with, but I think equally applicable is 3GPP QoS. But
> finding the right reference here might be a bit problematic.
>
> I think one can cover the general property of flow based QoS in a single
> sentence. However, that usage has implications for the peer connection
> and its configuration. We have text in the rtp-usage that makes it clear
> about the requirement to be able to send multiple RTP sessions on
> different flows. But the combination with the data channel is not
> covered. Thus, I think that belongs in the transports document. The
> question is how much lead in material you need for that discussion?

I'll add some general words about 5-tuples and 6-tuples to the QoS section.
Let's see if that resolves the remaining issues below; I don't want to
use too much time on this before launching a revised document to invite
others' comments.



>
>>>  I think this
>>> is the logical place to take the main discussion of that, as it can
>>> address both RTP streams and the data channel together. Please consider
>>> pointers to appropriate discussion in Section 12.1.3 in
>>> draft-ietf-rtcweb-rtp-usage.
>> I do not think this document should repeat any material from rtp-usage,
>> but rtp-usage does not consider any aspect of QoS marking for multiple
>> streams using the same 5-tuple, nor does it say anything about
>> flow-based or DPI-based mechanisms apart from noting their existence.
> No, I am not talking about repeating. I am discussing the requirements
> on being able to separate the data channel on transport level (UDP) from
> the RTP sessions over their RTP, as well as being able to put them on
> common UDP flows.
>
>> Agree that the multiplexing aspect should be discussed somewhere, but
>> I'm not sure where.
>>> I do note that writing this section appropriately is difficult until the
>>> content of draft-dhesikan-tsvwg-rtcweb-qos has firmed up.
>> True.
>>> 6. Content of Section 3.3:
>>>
>>> I think this section should focus on making clear what different types
>>> of streams that exist in the WebRTC context, and what choices of
>>> multiplexing that are possible. Basically ensure that the full set of
>>> combinations are made clear.
>> Isn't this material that really should be in
>> draft-ietf-avtcore-rtp-multi-stream or
> This is not appropriate place, this only discusses multiple SSRCs in the
> same RTP session.
>
>> draft-ietf-avtcore-multiplex-guidelines?
> This one definitely needs such discussion yes.
>
> Also draft-ietf-avtcore-multi-media-rtp-session has some parts of this,
> due to the potential for different needs based on media type.
>
>> Again, this seems the wrong draft for having a discussion of those issues.
> However, adding the data channel into this, is something WebRTC
> specific, thus you are not getting away from dealing with it.
>
>>> A. All RTP packet streams and data channel over a single UDP flow.
>>>
>>> B. All RTP packet streams over one UDP flow and the data channel over
>>> another UDP flow.
>>>
>>> C. The RTP packet streams over different UDP flows based on content type
>>> and separate data channel.
>>>
>>> D. Each RTP packet stream over its own UDP flow (RTP session) and the
>>> data channel over its own UDP flow.
>>>
>>> The above with the additional dimension, that each RTP sessions, RTCP
>>> can be in its own UDP flow.
>>>
>>> There is also a question if the data channel can be aggregated with only
>>> one RTP session, where there are multiple ones configured.
>>>
>>> The QoS related discussion of this can be moved to the QoS section to
>>> keep that focused.
>>>
>>> However, the discussion of the pro and cons with having one or more flow
>>> is good and could even be expanded to list reasons such as legacy
>>> interop over a gateway.
>>>
>>> 7. Section 3.3:
>>>    RTCWEB implementations MUST support the ability to send and receive
>>>    multiple SSRCs on the same transport, and MUST support the ability to
>>>    send and receive multiple SSRCs on multiple simultaneous transports,
>>>    including the ability to send and receive audio and video on the same
>>>    transport.
>>>
>>> I think this sentence is restating what is already required by Section
>>> 4.1 in draft-ietf-rtcweb-rtp-usage: If you want to enforce this, I
>>> suggest to do that by reference to that text rather than using RFC 2119
>>> normative statements.
>> Yes, rtp-usage contains these requirements. I can delete it from here.
>> Thanks!
>>
>> However, this removes the handle on which I hung the discussion of why
>> one would want to multiplex or not, which I inserted because you
>> requested it. This discussion also seems to be present in rtp-usage, so
>> it seems appropriate to delete that too.
> As I argue above, there is a need to take a full WebRTC level discussion
> of the multiplexing issues. I think the appropriate here is to try to
> ensure the AVTCORE documents contains the relevant pieces so you know
> what to reference and then try to put togheter a high level RTP + Data
> channel multiplexing considerations in this document.
>
>>>
>>> 8. Section 3.4:
>>>    ICE [RFC5245] MUST be supported.  The implementation MUST be a full
>>>    ICE implementation, not ICE-Lite.
>>>
>>>    In order to deal with situations where both parties are behind NATs
>>>    which perform endpoint-dependent mapping (as defined in [RFC5128]
>>>    section 2.4), TURN [RFC5766] MUST be supported.
>>>
>>> These paragraph implies STUN and TURN server configuration. Should that
>>> configuration question be discussed with the appropriate pointers to the
>>> API?
>> This document has no pointers to the API at the moment, and I do not
>> want to add them - it's a layering violation of sorts. We can add
>> pointers to JSEP, which is kind of close to the API.
>>
>> But the main point seems to be that STUN and TURN servers MUST be
>> configurable, both from the browser setup and from the (JS) application.
>> I'm adding a sentence saying that.
> That is probably sufficient.
>
>>> 9. Section 3.4:
>>>    In order to deal with firewalls that block all UDP traffic, TURN
>>>    using TCP between the client and the server MUST be supported, and
>>>    TURN using TLS between the client and the server MUST be supported.
>>>    See [RFC5766] section 2.1 for details.
>>>
>>> I think the following part: "TURN using TLS between" should be changed
>>> over "TURN using TLS over TCP between" to make it clear that this is
>>> using TLS/TCP rather than TLS/FOO.
>> Done.
>>> 10. Section 3.4:
>>>    TURN TCP candidates [RFC6062] SHOULD be supported; this allows
>>>    applications to achieve peer-to-peer communication when both parties
>>>    are behind UDP-blocking firewalls using a single TURN server.  (In
>>>    this case, one can also achieve communication using two TURN servers
>>>    that use TCP between the server and the client, and UDP between the
>>>    TURN servers.)
>>>
>>> My understanding of RFC6062 is that it allows one to establish a vanilla
>>> TCP connection on the far side of the TURN server as well as connecting
>>> that one to an almost vanilla TCP connection (just a initial TURN
>>> ConnectionBind request) that A establish to the TURN server, i.e.
>>> WebRTC-A -(Turn/TCP)-> TURN_Server -(TCP)-> WebRTC-B.
>>>
>>> The end-result is that a relayed vanilla TCP connection is established
>>> between the WebRTC endpoints (A and B). As the WebRTC transport so far
>>> defined only works with datagrams, i.e. DTLS or RTP/RTCP packets they
>>> can't be sent natively over a TCP byte stream. Thus, if this mode of
>>> operation is to be supported a framing is required on this leg.
>>>
>>> A possibility here could be RFC 4571 if one want to go this way.
>> I added a 4571 reference, and a note saying this is up for discussion.
>> If we frame RTP over TCP, we also have to decide what we do about the SCTP.
> Yes, I get the impression that the people arguing for the WebRTC over
> IP/TCP has not thought through the implications and I definitely like a
> discussion of this.
>
> Also, in which context are you wondering over SCTP in the above?

This got cleared up in London - SCTP would go over 4571 too.

>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>


-- 
Surveillance is pervasive. Go Dark.


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Please see inline.


On 2014-03-30 22:46, Harald Alvestrand wrote:
> Reviving an old thread, since I'm getting the changes in:
> 
> On 02/25/2014 11:46 AM, Magnus Westerlund wrote:
>> On 2014-02-19 19:47, Harald Alvestrand wrote:
>>> On 02/19/2014 11:08 AM, Magnus Westerlund wrote:

>>>> 2. Section 3.1:
>>>>    For both protocols, IPv4 and IPv6 support is assumed; applications
>>>>    MUST be able to utilize both IPv4 and IPv6 where available.
>>>>
>>>> What "applications" are intended in this sentence. I get a feeling that
>>>> "applications" here could be replaced by "WebRTC implementations"
>>> Javascript applications. I think I should add the word "application" to
>>> -overview's vocabulary; it is actually used in that meaning in that
>>> document (also in the forms "Web application" and "browser-based
>>> application".
>> Sorry, then I find the above requirement a bit strangely targeted. I
>> think we can require that IPv4 and IPv6 support for the applicable
>> protocol pieces in what we define. But, on the JS application level?
>> Also what explicit addressing information is visible on that level.
> 
> If two JS applications want to communicate, and the two systems are
> connected with IPv4 but not connected with IPv6, communication should
> happen, otherwise the implementation is non-conformant.
> 
> If the two systems are connected with IPv6 and not IPv4, communication
> should happen, otherwise the implementation is non-conformant.
> 
> Communication, or the lack of it, is clearly a property that's
> observable at the JS application level.
> 
> (the detail that the IPv* addresses are visible in the candidates and in
> the SDP is an implementation detail that doesn't belong here.)

I am not disagreeing with what you write above. However, the
implementation requirement on supporting IPv6 and IPv4 is on the WebRTC
endpoint, i.e. the media framework, the ICE implementation etc. A JS
application should be IP version agnostic as long as it doesn't much SDP
details.

That is why I raised this issue about "application" rather than the
WebRTC endpoint being the one that should support both IPv4 if endpoint
is provisioned with them.

> 
>>
>> It might be that this requirement needs to be split or at least apply to
>> multiple levels.
> 
> Or maybe rewritten to a language that we both understand. Leaving it
> roughly as-is for the moment.

Yes, rewrite is likely good.

> 
>>
>>>>
>>>> 3. Section 3.1:
>>>>    For UDP, this specification assumes the ability to set the DSCP code
>>>>    point of the sockets opened on a per-packet basis, in order to
>>>>    achieve the prioritizations described in
>>>>    [I-D.dhesikan-tsvwg-rtcweb-qos] when multiple media types are
>>>>    multiplexed.  It does not assume that the DSCP codepoints will be
>>>>    honored, and does assume that they may be zeroed or changed, since
>>>>    this is a local configuration issue.
>>>>
>>>> I wonder if this should be moved into 3.2 section instead. At least the
>>>> part in 3.1 should focus on the DSCP setting part on the transport flow.
>>>> The implications of it working or not should be moved to 3.2. A forward
>>>> pointer to that second can be used instead.
>>> I think it fits better in 3.1 - this is about the assumptions that the
>>> specifcation makes about the parts of the system that it is not a
>>> specification for.
>> I am fine with the first sentence staying, it is the second part that
>> needs more discussion and I think it should be moved and expanded on.
>> Thus a forward pointer may be appropriate here.
> 
> Added forward pointer, since I now have somewhere for it to point.

Good


>>>> 5. Content of Section 3.2
>>>>
>>>> This section needs to be expanded to start with a general discussion of
>>>> how QoS can be applied to get a benefit. But, I think one need to be
>>>> clear that it can't be assumed to be available in implementations or
>>>> WebRTC based applications.
>>> I want to push back on this. It seems inappropriate that the RTCWEB QoS
>>> specification start off with a tutorial on what QoS is and how one can
>>> use it to gain a benefit. That material should be available elsewhere.
>>> (Recommendations?)
>> I would like to agree with you, however we are putting together the
>> pieces in a way that stands some of the previous assumptions on their
>> heads. Thus, we don't have easy access to pointers.
>>
>> I can see this discussion going into
>> draft-ietf-avtcore-multiplex-guidelines when it concerns RTP. However,
>> you also have the data channel to consider. Thus, there exist no
>> references that I know that discusses this particular combination and
>> its implications.
>>
>>>> It needs to also discuss flow-based QoS mechanism. What are the
>>>> implications if that is what is available and what choices does one have
>>>> to address this by configuring the multiplexing different.
>>> I do not want to discuss material for which there are no references.
>>> What particular flow-based mechanisms do you have in mind, and what are
>>> the relevant references?
>> RSVP we can start with, but I think equally applicable is 3GPP QoS. But
>> finding the right reference here might be a bit problematic.
>>
>> I think one can cover the general property of flow based QoS in a single
>> sentence. However, that usage has implications for the peer connection
>> and its configuration. We have text in the rtp-usage that makes it clear
>> about the requirement to be able to send multiple RTP sessions on
>> different flows. But the combination with the data channel is not
>> covered. Thus, I think that belongs in the transports document. The
>> question is how much lead in material you need for that discussion?
> 
> I'll add some general words about 5-tuples and 6-tuples to the QoS section.
> Let's see if that resolves the remaining issues below; I don't want to
> use too much time on this before launching a revised document to invite
> others' comments.

Okay, I am fine with doing this incremental so that we can figure out
the appropriate boundaries and bring other docs up to required level also.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Mon Mar 31 03:35:48 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Transports for RTCWEB
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-03.txt
	Pages           : 11
	Date            : 2014-03-31

Abstract:
   This document describes the data transport protocols used by RTCWEB,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-transports-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-03


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Mon Mar 31 06:40:32 2014
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Subject: [rtcweb] Transport -03, bundling question (Re: Comments on draft-ietf-rtcweb-transports-02)
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Version -03 is now published. I hope you like it!

New outstanding item:

I added requirements for implementations to be able to generate both
fully bundled (one 5-tuple for everything) and fully unbundled (one
5-tuple for each flow) configurations, and for implementations to be
able to tolerate being hit with any combination of bundling schemes.

Is there a need to specify at MUST, SHOULD or MAY levels other combinations?

Harald


From nobody Mon Mar 31 06:55:08 2014
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From: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
To: ext Harald Alvestrand <harald@alvestrand.no>, Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Harald Alvestrand <hta@google.com>
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Hi Harald,

The decision in London to make ICE-TCP a "MUST" has not been implemented in=
 the -03 draft
(I have noticed you have reflected the related TURN-TCP discussion already)=
.

What are your plans for addressing the ICE-TCP decision?

Kind regards,
Uwe=20


> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext Harald
> Alvestrand
> Sent: Monday, March 31, 2014 3:40 PM
> To: Magnus Westerlund; rtcweb@ietf.org; Harald Alvestrand
> Subject: [rtcweb] Transport -03, bundling question (Re: Comments on
> draft-ietf-rtcweb-transports-02)
>=20
> Version -03 is now published. I hope you like it!
>=20
> New outstanding item:
>=20
> I added requirements for implementations to be able to generate both
> fully bundled (one 5-tuple for everything) and fully unbundled (one
> 5-tuple for each flow) configurations, and for implementations to be
> able to tolerate being hit with any combination of bundling schemes.
>=20
> Is there a need to specify at MUST, SHOULD or MAY levels other
> combinations?
>=20
> Harald
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 2014-03-31 15:40, Harald Alvestrand wrote:
> Version -03 is now published. I hope you like it!
> 
> New outstanding item:
> 
> I added requirements for implementations to be able to generate both
> fully bundled (one 5-tuple for everything) and fully unbundled (one
> 5-tuple for each flow) configurations, and for implementations to be
> able to tolerate being hit with any combination of bundling schemes.
> 
> Is there a need to specify at MUST, SHOULD or MAY levels other combinations?

RTP Usage has in Section 4.4 requirements concerning implementations
support for bundling of RTP that is:

- Required to support one RTP session per media type
- Required to support one RTP session for all media types

We don't have a requirement regarding one MST per RTP session in the RTP
usage document currently.

To this open issue I would like to highlight that we might need text
describing which combinations of RTP and data channel we expect to be
supported.

>From my individual perspective, I think being able to have data channel
on an individual transport-layer flow (5-tuple) is important to be able
to put QoS treatments on RTP packets within an RTP session separate from
the data channel. Having everything bundled, i.e one RTP session for all
media types combined with the data channel I think have strong support.
This leaves what possibilities one have to combine the data channel with
one out of several RTP sessions?

Cheers

Magnus Westerlund
(As individual)

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Subject: Re: [rtcweb] Transport -03, bundling question (Re: Comments on draft-ietf-rtcweb-transports-02)
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On 03/31/2014 03:54 PM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
> Hi Harald,
>
> The decision in London to make ICE-TCP a "MUST" has not been implemented in the -03 draft
> (I have noticed you have reflected the related TURN-TCP discussion already).
>
> What are your plans for addressing the ICE-TCP decision?

Thanks!

Checking the minutes, I find:

Chairs called a hum between the alternatives:

1) TCP ICE Candidates are MUST implement
2) TCP ICE Candidates are SHOULD/MAY implement
3) TCP ICE Candidates will not be discussed in document. 

The hum indicated very strong support for 1). 


My memory was faulty; I had thought that the same conclusion had applied
for TCP ICE candidates as for TURN ICE candidates. Version -04 will have
this fixed.

>
> Kind regards,
> Uwe 
>
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext Harald
>> Alvestrand
>> Sent: Monday, March 31, 2014 3:40 PM
>> To: Magnus Westerlund; rtcweb@ietf.org; Harald Alvestrand
>> Subject: [rtcweb] Transport -03, bundling question (Re: Comments on
>> draft-ietf-rtcweb-transports-02)
>>
>> Version -03 is now published. I hope you like it!
>>
>> New outstanding item:
>>
>> I added requirements for implementations to be able to generate both
>> fully bundled (one 5-tuple for everything) and fully unbundled (one
>> 5-tuple for each flow) configurations, and for implementations to be
>> able to tolerate being hit with any combination of bundling schemes.
>>
>> Is there a need to specify at MUST, SHOULD or MAY levels other
>> combinations?
>>
>> Harald
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 03/31/2014 03:54 PM, Rauschenbach,
      Uwe (NSN - DE/Munich) wrote:<br>
    </div>
    <blockquote
cite="mid:56C2F665D49E0341B9DF5938005ACDF82B7921@DEMUMBX005.nsn-intra.net"
      type="cite">
      <pre wrap="">Hi Harald,

The decision in London to make ICE-TCP a "MUST" has not been implemented in the -03 draft
(I have noticed you have reflected the related TURN-TCP discussion already).

What are your plans for addressing the ICE-TCP decision?</pre>
    </blockquote>
    <br>
    Thanks!<br>
    <br>
    Checking the minutes, I find:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <pre style="color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; white-space: pre-wrap;">Chairs called a hum between the alternatives:

1) TCP ICE Candidates are MUST implement
2) TCP ICE Candidates are SHOULD/MAY implement
3) TCP ICE Candidates will not be discussed in document. 

The hum indicated very strong support for 1). 
</pre>
    <br class="Apple-interchange-newline">
    My memory was faulty; I had thought that the same conclusion had
    applied for TCP ICE candidates as for TURN ICE candidates. Version
    -04 will have this fixed.<br>
    <br>
    <blockquote
cite="mid:56C2F665D49E0341B9DF5938005ACDF82B7921@DEMUMBX005.nsn-intra.net"
      type="cite">
      <pre wrap="">

Kind regards,
Uwe 


</pre>
      <blockquote type="cite">
        <pre wrap="">-----Original Message-----
From: rtcweb [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] On Behalf Of ext Harald
Alvestrand
Sent: Monday, March 31, 2014 3:40 PM
To: Magnus Westerlund; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; Harald Alvestrand
Subject: [rtcweb] Transport -03, bundling question (Re: Comments on
draft-ietf-rtcweb-transports-02)

Version -03 is now published. I hope you like it!

New outstanding item:

I added requirements for implementations to be able to generate both
fully bundled (one 5-tuple for everything) and fully unbundled (one
5-tuple for each flow) configurations, and for implementations to be
able to tolerate being hit with any combination of bundling schemes.

Is there a need to specify at MUST, SHOULD or MAY levels other
combinations?

Harald

_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
      </blockquote>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

--------------040702040407090902040606--


From nobody Mon Mar 31 10:09:15 2014
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] Some language on "prioritization"
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I see that I have promised to sugggest language for -transport- about
prioritization.
Here's an attempt. References and so on to be filled in later.

Comments welcome - this is a first stab!

--------------------------------------------
The RTCWEB prioritization model is that the application tells the RTCWEB
implementation about the priority of media and data flows through an API.

The priority associated with a media or data flow is classified as
"normal", "below normal", "high" or "very high". There are only four
priority levels at the API.

The RTCWEB implementation is responsible for mapping these to protocol
mechanics at the protocol interfaces, and to behave appropriately when
choosing what packets to send when.

[draft-dhesikan] specifies the mapping of the four levels of priority,
combined with the media type, to DSCP markings. This marking SHOULD be
done; the implementation MAY turn off use of DSCP markings if it detects
symptoms of unexpected behaviour like priority inversion or blocking of
packets with certain DSCP markings. The detection of these conditions is
implementation dependent. (Question: Does there need to be an API knob
to turn off DSCP markings?)

When an RTCWEB implementation has packets to send on multiple streams
that are congestion-controlled under the same congestion controller, the
RTCWEB implementation SHOULD serve the streams in a weighted round-robin
fashion, with each level of priority being given twice the transmission
capacity of the level below; thus, when congestion occurs, a "very high"
priority flow will have the ability to send 8 times as much data as a
"below normal" flow if both have data to send. This prioritization is
independent of the media type, but will lead to packet loss due to full
send buffers occuring first on the highest volume flows at any given
priority level. The details of which packet to send first are
implementation defined.

-- TODO: Specify a relative priority scheme that makes sense with SCTP,
with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
specifies a priority policy, but it's about discarding packets, not
deciding which packets to send first, and it also makes it impossible to
specify time-bounded retransmission. --

-- 
Surveillance is pervasive. Go Dark.


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Subject: Re: [rtcweb] Some language on "prioritization"
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On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> wrote:
> -- TODO: Specify a relative priority scheme that makes sense with SCTP,
> with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
> specifies a priority policy, but it's about discarding packets, not
> deciding which packets to send first, and it also makes it impossible to
> specify time-bounded retransmission. --


Why would SCTP need special treatment?  I can understand if there are
particular time-sensitive control messages that need to be given
higher priority, but this is all time-sensitive.


From nobody Mon Mar 31 10:43:13 2014
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Subject: Re: [rtcweb] Some language on "prioritization"
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On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com> wrote:

> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> wrote:
>> -- TODO: Specify a relative priority scheme that makes sense with SCTP,
>> with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
>> specifies a priority policy, but it's about discarding packets, not
>> deciding which packets to send first, and it also makes it impossible to
>> specify time-bounded retransmission. --
> 
> 
> Why would SCTP need special treatment?  I can understand if there are
> particular time-sensitive control messages that need to be given
> higher priority, but this is all time-sensitive.
A single transport connection (in this case an SCTP association) can
only use a single DSCP. So it would be OK to use the same priority for
all data channels, but things get complicated when when some data channels
would have different priorities requiring different DSCP markings.

Best regards
Michael
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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Subject: Re: [rtcweb] Some language on "prioritization"
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On 03/31/2014 07:42 PM, Michael Tuexen wrote:
> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com> wrote:
>
>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> wrote:
>>> -- TODO: Specify a relative priority scheme that makes sense with SCTP,
>>> with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
>>> specifies a priority policy, but it's about discarding packets, not
>>> deciding which packets to send first, and it also makes it impossible to
>>> specify time-bounded retransmission. --
>>
>> Why would SCTP need special treatment?  I can understand if there are
>> particular time-sensitive control messages that need to be given
>> higher priority, but this is all time-sensitive.
> A single transport connection (in this case an SCTP association) can
> only use a single DSCP. So it would be OK to use the same priority for
> all data channels, but things get complicated when when some data channels
> would have different priorities requiring different DSCP markings.

The SCTP protocol machine will, at some given moment in time, have
packets in its send buffers that belong to multiple data channels. Only
one of them can go on the wire first.

Which packet does it choose to send first, and why?

That's the question I'm looking for an answer to.

If the answer has an RFC number, chapter and section, I'd be very happy.
If it doesn't - perhaps we have to invent one.
Or say that it's "implementation dependent".


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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Some language on "prioritization"
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Some language on "prioritization"
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And if SCTP and RTP have packets to send, which goes first, why, and under =
what circumstances?

Matthew Kaufman

(Sent from my iPhone)

> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand" <harald@alvestrand.no> =
wrote:
>=20
>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com> wro=
te:
>>>=20
>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> wrote=
:
>>>> -- TODO: Specify a relative priority scheme that makes sense with SCTP=
,
>>>> with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
>>>> specifies a priority policy, but it's about discarding packets, not
>>>> deciding which packets to send first, and it also makes it impossible =
to
>>>> specify time-bounded retransmission. --
>>>=20
>>> Why would SCTP need special treatment?  I can understand if there are
>>> particular time-sensitive control messages that need to be given
>>> higher priority, but this is all time-sensitive.
>> A single transport connection (in this case an SCTP association) can
>> only use a single DSCP. So it would be OK to use the same priority for
>> all data channels, but things get complicated when when some data channe=
ls
>> would have different priorities requiring different DSCP markings.
>=20
> The SCTP protocol machine will, at some given moment in time, have
> packets in its send buffers that belong to multiple data channels. Only
> one of them can go on the wire first.
>=20
> Which packet does it choose to send first, and why?
>=20
> That's the question I'm looking for an answer to.
>=20
> If the answer has an RFC number, chapter and section, I'd be very happy.
> If it doesn't - perhaps we have to invent one.
> Or say that it's "implementation dependent".
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

