
From tim@phonefromhere.com  Sun Apr  1 10:35:43 2012
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From: Tim Panton <tim@phonefromhere.com>
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Subject: [rtcweb] The importance of selecting a realistic mandatory video codec
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In the ITEF 83 meeting I put forward the importance of a common video =
codec, "even if it is a grotty one".
What I tried to convey and may not have got over is that it needs to be =
a codec that all the browser
vendors will actually deliver, ideally in their first production =
releases of RTCweb.

Whilst a common codec is desirable for interop, it isn't essential, =
there can/will be transcoding proxies
which will paper over the differences and RTCweb can still succeed.

However if we see widespread use of transcoding, we will lose 2 of =
RTCweb's most exciting properties.

1) True P2P - i.e. the case where ICE can negotiate a direct connection =
behind a big NAT
2) security for calls over untrusted networks.

I am looking at use-cases for RTCweb where these are very important =
properties - think disaster relief for example,
without them RTCweb will be significantly harder to deploy over adhoc =
mesh wifis where the connection to the internet is very low bandwidth =
(if it exists at all) but where the meshed nodes can cover quite a big =
area with
decent, if variable, throughput.

For these cases it is essential we mandate a codec that will be =
delivered by _everyone_ . I don't care which it is.

The same arguments go for a mandated audio codec=20
- with the additional requirements that it needs to be resilient to
packetloss (probably have FEC) and able to dynamically adjust bitrate to =
cope with the mesh changing topology.

Tim.=

From basilgohar@librevideo.org  Sun Apr  1 12:27:06 2012
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Date: Sun, 01 Apr 2012 15:26:57 -0400
From: Basil Mohamed Gohar <basilgohar@librevideo.org>
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Subject: Re: [rtcweb] The importance of selecting a realistic mandatory video codec
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On 04/01/2012 01:35 PM, Tim Panton wrote:
> In the ITEF 83 meeting I put forward the importance of a common video codec, "even if it is a grotty one".
> What I tried to convey and may not have got over is that it needs to be a codec that all the browser
> vendors will actually deliver, ideally in their first production releases of RTCweb.
>
> Whilst a common codec is desirable for interop, it isn't essential, there can/will be transcoding proxies
> which will paper over the differences and RTCweb can still succeed.
>
> However if we see widespread use of transcoding, we will lose 2 of RTCweb's most exciting properties.
>
> 1) True P2P - i.e. the case where ICE can negotiate a direct connection behind a big NAT
> 2) security for calls over untrusted networks.
>
> I am looking at use-cases for RTCweb where these are very important properties - think disaster relief for example,
> without them RTCweb will be significantly harder to deploy over adhoc mesh wifis where the connection to the internet is very low bandwidth (if it exists at all) but where the meshed nodes can cover quite a big area with
> decent, if variable, throughput.
>
> For these cases it is essential we mandate a codec that will be delivered by _everyone_ . I don't care which it is.
>
> The same arguments go for a mandated audio codec
> - with the additional requirements that it needs to be resilient to
> packetloss (probably have FEC) and able to dynamically adjust bitrate to cope with the mesh changing topology.
>
> Tim.
Tim,

You bring up a very good point.  I will make and explain the case for my 
suggestions here.

For audio, there is a format that is in the final stages of being 
finalized that exceeds anything else out there in nearly all metrics, 
namely, Opus[1].  Opus is a royalty-free, free and open-source audio 
codec that features high quality, efficiency, and latency.  Several of 
the devs are on this list, so I will let them respond about the issue of 
FEC, but I think this is present as well.  It was desired with this kind 
of usage in mind, and is a merger of the CELT "music" codec and Skype's 
SILK voice codec.  In short, Opus is a shoe-in for audio for WebRTC, as 
it is superior in every category to anything that could come reasonably 
close to be sufficient for its needs.

For video, we get back to the same discussion as has come-up time and 
again.  The crux of your point is to avoid the need for a defacto 
deployment by implementors of wide-scale transcoding in conjunction with 
WebRTC as that would introduce undesirable side effects.  I don't think 
anyone here disagrees with you, as having as unhindered a connection as 
possible is ideal in communications.  The problem is there are vested 
interests that will choose one video format over another for reasons 
that are unsympathetic to all sides.  For this reason, it's important to 
fall back onto the reasoning for the success of the Internet as a 
universal medium for communication, and that has been due, in no small 
part, to the adoption of technologies that provide no legal barrier to 
their implementation.

As you can probably see, I am making the case for VP8 to be the format 
of choice for this reason.  VP8 provides absolutely no barrier to 
adoption to a new player in the field of realtime communications, and a 
minimal one to someone that's already a player.  Implementations in both 
hardware and software exist for VP8, and the developers have made 
realtime communications a focus for design choices, as well as quality, 
efficiency, and performance.  As a baseline video format, it doesn't get 
better than VP8.

Will this prevent others from implementing transcoding as part of their 
WebRTC infrastructure?  We cannot force nor guarantee that (except by 
explicitly stating that in the spec, which I think is a bad idea for the 
few cases where such a tranform would be needed and/or useful).  But we 
can at least lower the barrier of participation for new players so that 
they can implement WebRTC without requiring additional licenses.  VP8 
will allow for that, and make long-term, sustainable adoption of the 
standard that much easier.

-- 
Libre Video
http://librevideo.org


From juberti@google.com  Sun Apr  1 20:17:11 2012
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Subject: Re: [rtcweb] Interaction between MediaStream API and signaling
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On Sat, Mar 31, 2012 at 1:41 AM, Stefan Hakansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 03/31/2012 07:17 AM, Stefan Hakansson LK wrote:
>
>> On 03/30/2012 11:39 PM, Randell Jesup wrote:
>>
>>> On 3/30/2012 4:59 AM, Stefan Hakansson LK wrote:
>>>
>>>> The JS API has deals with MediaStreams (this is what you send and
>>>> receive using PeerConnection from an application perspective).
>>>>
>>>> A browser receiving RTP streams, needs side info to be able to
>>>> assemble those RTP streams into MediaStreams in a correct way. The
>>>> current model is that this is signaled using SDP exchanges (where
>>>> Haralds MSID proposal would tell which MediaStream an RTP stream
>>>> belongs to).
>>>>
>>>> As I brought up at the mike yesterday, I think we may have a race
>>>> condition for the responder.
>>>>
>>>> For the initiator side browser, this is clear: once an (PR-)ANSWER is
>>>> received, the responder has received the SDP, and hence can map
>>>> incoming RTP streams into MediaStreams.
>>>>
>>>> But for the responder side this is less clear to me. Imagine
>>>> applications where the responder just mirrors the initiator - if one
>>>> of the parties adds a MediaStream to PeerConnection, the other end
>>>> would add the corresponding MediaStream.
>>>>
>>>> This can happen any time in the session, so ICE can very well be up
>>>> and running. One example could be that the data channel is used for
>>>> text chat, when one side clicks a button to start video. And the
>>>> application can have asked for permission to use all input devices
>>>> earlier, so no user interaction may be involved.
>>>>
>>>> In this situation the responder's (added) RTP streams can very well
>>>> arrive before the ANSWER if I understand correctly.
>>>>
>>>
>>> Yes.  Just like in SIP.  And so when you send an OFFER (or modified
>>> re-OFFER), you must be ready to receive data per that offer even if no
>>> ANSWER has been received - just like in SIP.  And if its a re-offer, you
>>> need to accept the old, and accept the new (though you could probably
>>> use reception of obviously new-OFFER media to turn off
>>> decoding/rendering old-OFFER in preparation for the ANSWER).
>>>
>>> The flip side of this is the responder has to infer when the sender
>>> switches over to the result of the ANSWER from the media.  For example:
>>>
>>> A                                      B
>>> <--- H.261 --->
>>> re-OFFER(VP8) --->
>>> <-- ANSWER(VP8) (delayed in reception)
>>> <-----------VP8            (A should infer that B ANSWERed and accepted
>>> VP8)
>>>    ---------->   H.261
>>> <-- ANSWER(VP8) (received)
>>> <--------VP8---------->   (B should infer by reception of VP8 that ANSWER
>>> was received)
>>>
>>> (Personally, I hate inferences, but without a 3 (or 4) way handshake,
>>> you have to).  If you switches of codecs are staged, then this isn't
>>> (much) of a problem.  Either leave old codec on the list, or leave it on
>>> the list until accept, and then re-OFFER to remove the un-used codec.
>>>
>>
>> I think I understand what you mean, and this would work fine as long as
>> you just switch codecs that are used in already set-up MediaStreams.
>>
>> But if A in this case, as part of re-OFFERING the session, not only
>> offers a new codec (VP8) for the already flowing video but also adds a
>> new outgoing video stream (e.g. front cam), and then (without receiving
>> the ANSWER - delayed in reception) starts receiving VP8 video it could
>> not really know if this VP8 video is new video from the responders front
>> cam or just a new codec for the existing (back cam) video from the
>> responder to the sender.
>>
>
> This may have been a very bad example. Probably you can tell them apart on
> the SSRC. But even so, the A browser won't know what the VP8 stream (if it
> has an unknown SSRC) represents without receiving the ANSWER.


I think this is only an issue if you decide to add streams in the ANSWER.
But even so, eventually the ANSWER arrives and you can start
demuxing/decoding appropriately.

Regardless, if the app wants to require some sort of ACK message to begin
transmission, or perhaps require that the remote side ask for the media
streams it wants to be sent, I think this could be implemented in the
app-specific signaling layer; the streams could initially be added as
inactive, and only changed to sendrecv when the ACK arrives.


>
>
>
>>
>>> One problem is what to do in the switchover window when you might get a
>>> mixture of old and new media, especially if you moved them to different
>>> ports and so can't count on RTP sequence re-ordering to un-mix them; in
>>> the past I dealt with that (and long codec-switch times) by locking out
>>> codec changes for a fraction of a second after I do one.  Not a huge
>>> deal, however.
>>>
>>> My apologies if I've missed something in JSEP; I've been heads-down
>>> enough in Data Channels and bring-up that I could have a disconnect here
>>> and be saying something silly.
>>>
>>
>> Actually I don't think this is very JSEP related; it is the generic
>> problem that the browser receiving RTP streams need some side info about
>> them before being able to do anything sensible with them.
>>
>>
>>>  I think we need to find a way to handle this. One way is to add an
>>>> "ACK" that indicates to the responder that the initiator has received
>>>> the ANSWER, but I'm not sure that is the best way.
>>>>
>>>
>>> If you need to know that, you need a SIP-style ACK.
>>>
>>
>> As explained, I do think we need to know that.
>>
>>
>>>
>> ______________________________**_________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

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<br><br><div class=3D"gmail_quote">On Sat, Mar 31, 2012 at 1:41 AM, Stefan =
Hakansson LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@er=
icsson.com" target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</spa=
n> 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>On 03/31/2012 07:17 AM, Stefan Hakansson LK wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 03/30/2012 11:39 PM, Randell Jesup wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 3/30/2012 4:59 AM, Stefan Hakansson LK wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
The JS API has deals with MediaStreams (this is what you send and<br>
receive using PeerConnection from an application perspective).<br>
<br>
A browser receiving RTP streams, needs side info to be able to<br>
assemble those RTP streams into MediaStreams in a correct way. The<br>
current model is that this is signaled using SDP exchanges (where<br>
Haralds MSID proposal would tell which MediaStream an RTP stream<br>
belongs to).<br>
<br>
As I brought up at the mike yesterday, I think we may have a race<br>
condition for the responder.<br>
<br>
For the initiator side browser, this is clear: once an (PR-)ANSWER is<br>
received, the responder has received the SDP, and hence can map<br>
incoming RTP streams into MediaStreams.<br>
<br>
But for the responder side this is less clear to me. Imagine<br>
applications where the responder just mirrors the initiator - if one<br>
of the parties adds a MediaStream to PeerConnection, the other end<br>
would add the corresponding MediaStream.<br>
<br>
This can happen any time in the session, so ICE can very well be up<br>
and running. One example could be that the data channel is used for<br>
text chat, when one side clicks a button to start video. And the<br>
application can have asked for permission to use all input devices<br>
earlier, so no user interaction may be involved.<br>
<br>
In this situation the responder&#39;s (added) RTP streams can very well<br>
arrive before the ANSWER if I understand correctly.<br>
</blockquote>
<br>
Yes. =C2=A0Just like in SIP. =C2=A0And so when you send an OFFER (or modifi=
ed<br>
re-OFFER), you must be ready to receive data per that offer even if no<br>
ANSWER has been received - just like in SIP. =C2=A0And if its a re-offer, y=
ou<br>
need to accept the old, and accept the new (though you could probably<br>
use reception of obviously new-OFFER media to turn off<br>
decoding/rendering old-OFFER in preparation for the ANSWER).<br>
<br>
The flip side of this is the responder has to infer when the sender<br>
switches over to the result of the ANSWER from the media. =C2=A0For example=
:<br>
<br>
A =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=A0B<br>
&lt;--- H.261 ---&gt;<br>
re-OFFER(VP8) ---&gt;<br>
&lt;-- ANSWER(VP8) (delayed in reception)<br>
&lt;-----------VP8 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0(A should infer=
 that B ANSWERed and accepted VP8)<br>
 =C2=A0 =C2=A0----------&gt; =C2=A0 H.261<br>
&lt;-- ANSWER(VP8) (received)<br>
&lt;--------VP8----------&gt; =C2=A0 (B should infer by reception of VP8 th=
at ANSWER<br>
was received)<br>
<br>
(Personally, I hate inferences, but without a 3 (or 4) way handshake,<br>
you have to). =C2=A0If you switches of codecs are staged, then this isn&#39=
;t<br>
(much) of a problem. =C2=A0Either leave old codec on the list, or leave it =
on<br>
the list until accept, and then re-OFFER to remove the un-used codec.<br>
</blockquote>
<br>
I think I understand what you mean, and this would work fine as long as<br>
you just switch codecs that are used in already set-up MediaStreams.<br>
<br>
But if A in this case, as part of re-OFFERING the session, not only<br>
offers a new codec (VP8) for the already flowing video but also adds a<br>
new outgoing video stream (e.g. front cam), and then (without receiving<br>
the ANSWER - delayed in reception) starts receiving VP8 video it could<br>
not really know if this VP8 video is new video from the responders front<br=
>
cam or just a new codec for the existing (back cam) video from the<br>
responder to the sender.<br>
</blockquote>
<br></div></div>
This may have been a very bad example. Probably you can tell them apart on =
the SSRC. But even so, the A browser won&#39;t know what the VP8 stream (if=
 it has an unknown SSRC) represents without receiving the ANSWER.</blockquo=
te>


<div><br></div><div>I think this is only an issue if you decide to add stre=
ams in the ANSWER. But even so, eventually the ANSWER arrives and you can s=
tart demuxing/decoding appropriately.</div><div><br></div><div>Regardless, =
if the app wants to require some sort of ACK message to begin transmission,=
 or perhaps require that the remote side ask for the media streams it wants=
 to be sent, I think this could be implemented in the app-specific signalin=
g layer; the streams could initially be added as inactive, and only changed=
 to sendrecv when the ACK arrives.</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"><div><div><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
One problem is what to do in the switchover window when you might get a<br>
mixture of old and new media, especially if you moved them to different<br>
ports and so can&#39;t count on RTP sequence re-ordering to un-mix them; in=
<br>
the past I dealt with that (and long codec-switch times) by locking out<br>
codec changes for a fraction of a second after I do one. =C2=A0Not a huge<b=
r>
deal, however.<br>
<br>
My apologies if I&#39;ve missed something in JSEP; I&#39;ve been heads-down=
<br>
enough in Data Channels and bring-up that I could have a disconnect here<br=
>
and be saying something silly.<br>
</blockquote>
<br>
Actually I don&#39;t think this is very JSEP related; it is the generic<br>
problem that the browser receiving RTP streams need some side info about<br=
>
them before being able to do anything sensible with them.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I think we need to find a way to handle this. One way is to add an<br>
&quot;ACK&quot; that indicates to the responder that the initiator has rece=
ived<br>
the ANSWER, but I&#39;m not sure that is the best way.<br>
</blockquote>
<br>
If you need to know that, you need a SIP-style ACK.<br>
</blockquote>
<br>
As explained, I do think we need to know that.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
</blockquote>
<br>
______________________________<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>
</blockquote>
<br>
______________________________<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>

--20cf3005dc4029f92c04bca9a0be--

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From: Justin Uberti <juberti@google.com>
Date: Sun, 1 Apr 2012 23:21:49 -0400
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Subject: Re: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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--20cf3074da54138cd604bca9b220
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On Thu, Mar 29, 2012 at 3:57 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> WG, as individual
>
> As just discussed in the meeting Justin and Hadriel didn't think that
> you could clone PeerConnections properly. I do believe it is possible.
>
> If the API provides a method where you can clone a current
> peerConnection you can actually share the transport layer information at
> one end-point for multiple peer connections assuming that the other
> end-points transport addresses are different between the different
> peerConnections being cloned.
>
> This works as the complete ICE username is the concatenation of both
> sides, as long as an secondary answer contains a different username and
> has candidates with different transport addresses (address+port). Then
> end-point A receiving the second answer can have the ICE stack accept
> both usernames and if connectivity checks works then you create
> different 5-tuple filter to divide any media streams to the right peer
> connections media handler.
>
> When you clone a PC then must allocate the media decoding resources for
> each cloning. And that is something the cloning function may block on.
>
>
Thanks Magnus. This sounds reasonable; it might even be possible to make
cloning work using the existing API by taking the local description from
one PC and installing it as the local description for a new PC; the browser
could recognize that the candidates refer to existing ports and behave as
you mention above.

However, given that this seems to be mostly an edge case, I think we can
probably save this for v2.

--20cf3074da54138cd604bca9b220
Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Thu, Mar 29, 2012 at 3:57 AM, Magnus =
Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">

WG, as individual<br>
<br>
As just discussed in the meeting Justin and Hadriel didn&#39;t think that<b=
r>
you could clone PeerConnections properly. I do believe it is possible.<br>
<br>
If the API provides a method where you can clone a current<br>
peerConnection you can actually share the transport layer information at<br=
>
one end-point for multiple peer connections assuming that the other<br>
end-points transport addresses are different between the different<br>
peerConnections being cloned.<br>
<br>
This works as the complete ICE username is the concatenation of both<br>
sides, as long as an secondary answer contains a different username and<br>
has candidates with different transport addresses (address+port). Then<br>
end-point A receiving the second answer can have the ICE stack accept<br>
both usernames and if connectivity checks works then you create<br>
different 5-tuple filter to divide any media streams to the right peer<br>
connections media handler.<br>
<br>
When you clone a PC then must allocate the media decoding resources for<br>
each cloning. And that is something the cloning function may block on.<br>
<br></blockquote><div><br></div><div>Thanks Magnus. This sounds reasonable;=
 it might even be possible to make cloning work using the existing API by t=
aking the local description from one PC and installing it as the local desc=
ription for a new PC; the browser could recognize that the candidates refer=
 to existing ports and behave as you mention above.</div>

<div><br></div><div>However, given that this seems to be mostly an edge cas=
e, I think we can probably save this for v2.</div><div><br></div></div>

--20cf3074da54138cd604bca9b220--

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Stephan,

> The most commonly cited timeline for a widely in use technology to be
> "save" from a patent viewpoint, based on equitable defenses such as laches
> (in the US) is six years.  In some countries of significant size, this
> time is longer, and in others, equitable defenses do not exist.  (Very
> briefly, and perhaps incorrectly put, those equitable defenses allow a
> defendant to argue successfully that a patent cannot be enforced as the
> right holder knew that the patent claim was likely being infringed, and
> did not enforce the patent.).

Recall that Unisys forced people to pay royalties for using the GIF file
format long after it became widely popular.  The company asserted IPR claims
on GIF since it used a compression algorithm to which it acquired the
rights, and it started doing so right near the end of the 20-year period
during which a patent is considered valid.

In short, I really do not think one should ever assume it might be safe to
use any technology.

> In addition, as I pointed out in the meeting, the use of a video codec
> created by a body such as MPEG or ITU-T SG16 has the advantage of that the
> patents of all participating players are available at least under
> Reasonable and Non Discriminatory (RAND) terms.  This may sound like a Bad
> Thing if you operate under a business model that prevents you to pay
> anything for patent licenses, but it is surely a Good Thing if you are
> willing to dish out a moderate amount of money for a license.  RAND
> recently has gotten teeth, not so much in terms of the monetary
> compensation aspect, but in terms of difficulty (if not unavailability) to
> obtain injunctive relive, among others.  H.26x and the MPEG standards
> benefit from RAND commitments, VP8, AFAIK, does not.

I believe your point here is perhaps worth even more consideration.

I really know nothing about the IPR that exists or might be claimed on VP8.
That said, I know there has such an incredible amount of work by so many
companies to produce H.264 that I would be utterly surprised to find that
VP8 does not infringe on something.  All of the technology that went into
H.264 represents only a subset of all of the IPR that exists in the video
coding space.

It's the rest of the IPR, a bunch of IPR owned by companies who actually
have significant investment in video coding technologies, that I believe
people should worry about.  Everyone who worked on H.264 did so as part of
an open standards process, as you mention above.  They spent a lot of time
and energy in the process, coming to an agreement to license the technology
that went into H.264.  They did not agree to license technology that did not
go into H.264.  So, if one of those participating companies were to
subsequently sue over some IPR used in VP8, I would not dare call them a
troll.

Trolls always exist and may even lay claims to H.264, but I suspect it would
be much harder for a troll to lay claim to H.264.  That codec spent years in
development with input from tons of people.  If a troll were to come in and
lay claim to some part of H.264, I suspect there would be several companies
that would stand up and beat them down.  After all, a claim against H.264
would not only present a problem for the defendant in a lawsuit, but would
be an issue for every company with IPR on H.264.  Further, no matter what
part of H.264 such a troll might decide to pick on, there are a number of
world-class engineers who (as you indicated about H.263) could probably name
the person or persons who contributed the section of text, algorithm,
procedure, etc., all of which is covered with a known license.

VP8 does not have the benefit of such significant peer review and
collaboration, nor does it benefit from any IPR licensing agreements from
legitimate companies holding tons of video coding-related IPR.

Paul


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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Sun, 1 Apr 2012 22:28:34 -0700
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Subject: Re: [rtcweb] Proposal for H.263 baseline codec
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Note that H.264 isn't even under discussion here.

Also, my question on H.263 hasn't been answered yet. I do wonder about
the patent situation there!

Cheers,
Silvia.

On Sun, Apr 1, 2012 at 8:56 PM, Paul E. Jones <paulej@packetizer.com> wrote=
:
> Stephan,
>
>> The most commonly cited timeline for a widely in use technology to be
>> "save" from a patent viewpoint, based on equitable defenses such as lach=
es
>> (in the US) is six years. =A0In some countries of significant size, this
>> time is longer, and in others, equitable defenses do not exist. =A0(Very
>> briefly, and perhaps incorrectly put, those equitable defenses allow a
>> defendant to argue successfully that a patent cannot be enforced as the
>> right holder knew that the patent claim was likely being infringed, and
>> did not enforce the patent.).
>
> Recall that Unisys forced people to pay royalties for using the GIF file
> format long after it became widely popular. =A0The company asserted IPR c=
laims
> on GIF since it used a compression algorithm to which it acquired the
> rights, and it started doing so right near the end of the 20-year period
> during which a patent is considered valid.
>
> In short, I really do not think one should ever assume it might be safe t=
o
> use any technology.
>
>> In addition, as I pointed out in the meeting, the use of a video codec
>> created by a body such as MPEG or ITU-T SG16 has the advantage of that t=
he
>> patents of all participating players are available at least under
>> Reasonable and Non Discriminatory (RAND) terms. =A0This may sound like a=
 Bad
>> Thing if you operate under a business model that prevents you to pay
>> anything for patent licenses, but it is surely a Good Thing if you are
>> willing to dish out a moderate amount of money for a license. =A0RAND
>> recently has gotten teeth, not so much in terms of the monetary
>> compensation aspect, but in terms of difficulty (if not unavailability) =
to
>> obtain injunctive relive, among others. =A0H.26x and the MPEG standards
>> benefit from RAND commitments, VP8, AFAIK, does not.
>
> I believe your point here is perhaps worth even more consideration.
>
> I really know nothing about the IPR that exists or might be claimed on VP=
8.
> That said, I know there has such an incredible amount of work by so many
> companies to produce H.264 that I would be utterly surprised to find that
> VP8 does not infringe on something. =A0All of the technology that went in=
to
> H.264 represents only a subset of all of the IPR that exists in the video
> coding space.
>
> It's the rest of the IPR, a bunch of IPR owned by companies who actually
> have significant investment in video coding technologies, that I believe
> people should worry about. =A0Everyone who worked on H.264 did so as part=
 of
> an open standards process, as you mention above. =A0They spent a lot of t=
ime
> and energy in the process, coming to an agreement to license the technolo=
gy
> that went into H.264. =A0They did not agree to license technology that di=
d not
> go into H.264. =A0So, if one of those participating companies were to
> subsequently sue over some IPR used in VP8, I would not dare call them a
> troll.
>
> Trolls always exist and may even lay claims to H.264, but I suspect it wo=
uld
> be much harder for a troll to lay claim to H.264. =A0That codec spent yea=
rs in
> development with input from tons of people. =A0If a troll were to come in=
 and
> lay claim to some part of H.264, I suspect there would be several compani=
es
> that would stand up and beat them down. =A0After all, a claim against H.2=
64
> would not only present a problem for the defendant in a lawsuit, but woul=
d
> be an issue for every company with IPR on H.264. =A0Further, no matter wh=
at
> part of H.264 such a troll might decide to pick on, there are a number of
> world-class engineers who (as you indicated about H.263) could probably n=
ame
> the person or persons who contributed the section of text, algorithm,
> procedure, etc., all of which is covered with a known license.
>
> VP8 does not have the benefit of such significant peer review and
> collaboration, nor does it benefit from any IPR licensing agreements from
> legitimate companies holding tons of video coding-related IPR.
>
> Paul
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Silvia Pfeiffer'" <silviapfeiffer1@gmail.com>
References: <4F746163.5090506@hidayahonline.org> <CB9A367A.85338%stewe@stewe.org> <00fe01cd1084$9c9fac90$d5df05b0$@packetizer.com> <CAHp8n2nmhT2rGVOYG+usJUDnf4vKV5PWT2nxJuYCFRK3O_=p3Q@mail.gmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Proposal for H.263 baseline codec
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Silvia,

I believe H.263 would be covered if you license the MPEG-4 Visual =
"patent
pool".  I might be wrong, but you can certainly get clarification from =
MPEG
LA. You could also reach out to the list of companies who have claimed =
IPR
at the ITU (search for ITU IPR database).  I think there are about 14
companies who have IPR claims submitted.

As with H.264, it was a very open process.  Those involved are likely =
the
only ones with IPR claims.  Perhaps not, but given the pervasive use of
H.263 already on the Internet, one would think if there was some IPR in
hiding, it would have appeared already.

Paul

> -----Original Message-----
> From: Silvia Pfeiffer [mailto:silviapfeiffer1@gmail.com]
> Sent: Monday, April 02, 2012 1:29 AM
> To: Paul E. Jones
> Cc: Stephan Wenger; Basil Mohamed Gohar; rtcweb@ietf.org
> Subject: Re: [rtcweb] Proposal for H.263 baseline codec
>=20
> Note that H.264 isn't even under discussion here.
>=20
> Also, my question on H.263 hasn't been answered yet. I do wonder about =
the
> patent situation there!
>=20
> Cheers,
> Silvia.
>=20
> On Sun, Apr 1, 2012 at 8:56 PM, Paul E. Jones <paulej@packetizer.com>
> wrote:
> > Stephan,
> >
> >> The most commonly cited timeline for a widely in use technology to =
be
> >> "save" from a patent viewpoint, based on equitable defenses such as
> >> laches (in the US) is six years. =A0In some countries of =
significant
> >> size, this time is longer, and in others, equitable defenses do not
> >> exist. =A0(Very briefly, and perhaps incorrectly put, those =
equitable
> >> defenses allow a defendant to argue successfully that a patent =
cannot
> >> be enforced as the right holder knew that the patent claim was =
likely
> >> being infringed, and did not enforce the patent.).
> >
> > Recall that Unisys forced people to pay royalties for using the GIF
> > file format long after it became widely popular. =A0The company =
asserted
> > IPR claims on GIF since it used a compression algorithm to which it
> > acquired the rights, and it started doing so right near the end of =
the
> > 20-year period during which a patent is considered valid.
> >
> > In short, I really do not think one should ever assume it might be
> > safe to use any technology.
> >
> >> In addition, as I pointed out in the meeting, the use of a video
> >> codec created by a body such as MPEG or ITU-T SG16 has the =
advantage
> >> of that the patents of all participating players are available at
> >> least under Reasonable and Non Discriminatory (RAND) terms. =A0This =
may
> >> sound like a Bad Thing if you operate under a business model that
> >> prevents you to pay anything for patent licenses, but it is surely =
a
> >> Good Thing if you are willing to dish out a moderate amount of =
money
> >> for a license. =A0RAND recently has gotten teeth, not so much in =
terms
> >> of the monetary compensation aspect, but in terms of difficulty (if
> >> not unavailability) to obtain injunctive relive, among others. =
=A0H.26x
> >> and the MPEG standards benefit from RAND commitments, VP8, AFAIK, =
does
> not.
> >
> > I believe your point here is perhaps worth even more consideration.
> >
> > I really know nothing about the IPR that exists or might be claimed =
on
> VP8.
> > That said, I know there has such an incredible amount of work by so
> > many companies to produce H.264 that I would be utterly surprised to
> > find that
> > VP8 does not infringe on something. =A0All of the technology that =
went
> > into
> > H.264 represents only a subset of all of the IPR that exists in the
> > video coding space.
> >
> > It's the rest of the IPR, a bunch of IPR owned by companies who
> > actually have significant investment in video coding technologies,
> > that I believe people should worry about. =A0Everyone who worked on
> > H.264 did so as part of an open standards process, as you mention
> > above. =A0They spent a lot of time and energy in the process, coming =
to
> > an agreement to license the technology that went into H.264. =A0They =
did
> > not agree to license technology that did not go into H.264. =A0So, =
if
> > one of those participating companies were to subsequently sue over
> > some IPR used in VP8, I would not dare call them a troll.
> >
> > Trolls always exist and may even lay claims to H.264, but I suspect =
it
> > would be much harder for a troll to lay claim to H.264. =A0That =
codec
> > spent years in development with input from tons of people. =A0If a =
troll
> > were to come in and lay claim to some part of H.264, I suspect there
> > would be several companies that would stand up and beat them down.
> > After all, a claim against H.264 would not only present a problem =
for
> > the defendant in a lawsuit, but would be an issue for every company
> > with IPR on H.264. =A0Further, no matter what part of H.264 such a =
troll
> > might decide to pick on, there are a number of world-class engineers
> > who (as you indicated about H.263) could probably name the person or
> > persons who contributed the section of text, algorithm, procedure, =
etc.,
> all of which is covered with a known license.
> >
> > VP8 does not have the benefit of such significant peer review and
> > collaboration, nor does it benefit from any IPR licensing agreements
> > from legitimate companies holding tons of video coding-related IPR.
> >
> > Paul
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


From salvatore.loreto@ericsson.com  Mon Apr  2 03:55:29 2012
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [rtcweb] Query regarding rtcweb use cases and requirements.
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at moment the requirements/use case for Data are in the following draft
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-00


/Salvatore


On 3/27/12 1:36 PM, Harald Alvestrand wrote:
> On 03/27/2012 09:51 AM, ajrmeyn@gmail.com wrote:
>> Hi,
>>   
>> This is with regards to the following document:
>> Web Real-Time Communication Use-cases and Requirements
>> draft-ietf-rtcweb-use-cases-and-requirements-06.txt
>> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1
>> I have noticed that the use cases with regards to Data API of web 
>> real time communication, have not been addressed. For example: use 
>> cases that are related to file sharing between browsers.
>> is there any particular reason for the same?
> Requirement F23 is a data-passing requirement.
>
> For communications that did not need low latency, the WG felt that the 
> RTCWEB requirements were not different from the requirements of any 
> other application, so those could be fulfilled with other types of 
> interfaces, including server-mediated file transfers.
>
> The WG is suggesting data API/protocol solutions that can be used for 
> other things than fulfiling the most narrow interpretation of the 
> requirement listed, but did not feel a need to add a requirement 
> covering those.
>
> The purpose of the "use cases and requirements" is to find a minimum 
> set of things that the protocol suite has to be able to do in order to 
> be "good enough", not to delineate everything the protocol can be used 
> for.
>
> My interpretation....
>
>                       Harald
>
>
>
>
>> Regards,
>> Antony Meyn
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>


-- 
Salvatore Loreto, PhD
www.sloreto.com


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<html>
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    at moment the requirements/use case for Data are in the following
    draft<br>
    <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-00">http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-00</a><br>
    <br>
    <br>
    /Salvatore<br>
    <br>
    <br>
    On 3/27/12 1:36 PM, Harald Alvestrand wrote:
    <blockquote cite="mid:4F71982B.4010208@alvestrand.no" type="cite">
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      On 03/27/2012 09:51 AM, <a moz-do-not-send="true"
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      wrote:
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        type="cite">
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              <div dir="ltr">
                <div style="font-family: 'Arial'; color: rgb(0, 0, 0);
                  font-size: 10pt;">
                  <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">Hi,</pre>
                  <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">&nbsp;</pre>
                  <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">This is with regards to the following document:</pre>
                  <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">Web Real-Time Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-06.txt
</pre>
                  <a moz-do-not-send="true"
href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1"
                    target="_blank"><font face="Tahoma">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1</font></a><font
                    face="Tahoma"> </font>
                  <div><font face="Tahoma">I have noticed that the use
                      cases with regards to Data API of web real time
                      communication, have not been addressed. For
                      example: use cases that are related to file
                      sharing between browsers. </font></div>
                  <div>&nbsp;</div>
                  <div><font face="Tahoma">is there any particular
                      reason for the same?</font></div>
                </div>
              </div>
            </div>
          </div>
        </div>
      </blockquote>
      Requirement F23 is a data-passing requirement.<br>
      <br>
      For communications that did not need low latency, the WG felt that
      the RTCWEB requirements were not different from the requirements
      of any other application, so those could be fulfilled with other
      types of interfaces, including server-mediated file transfers.<br>
      <br>
      The WG is suggesting data API/protocol solutions that can be used
      for other things than fulfiling the most narrow interpretation of
      the requirement listed, but did not feel a need to add a
      requirement covering those.<br>
      <br>
      The purpose of the "use cases and requirements" is to find a
      minimum set of things that the protocol suite has to be able to do
      in order to be "good enough", not to delineate everything the
      protocol can be used for.<br>
      <br>
      My interpretation....<br>
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
      <br>
      <br>
      <br>
      <br>
      <blockquote cite="mid:FA86515700364A16B4EDF69E88E3FA1A@devbox"
        type="cite">
        <div dir="ltr">
          <div style="font-family: 'Arial'; color: rgb(0, 0, 0);
            font-size: 10pt;">
            <div style="font-style: normal; display: inline;
              font-family: 'Calibri'; color: rgb(0, 0, 0); font-size:
              small; font-weight: normal; text-decoration: none;">
              <div dir="ltr">
                <div style="font-family: 'Arial'; color: rgb(0, 0, 0);
                  font-size: 10pt;">
                  <div>&nbsp;</div>
                  <div><font face="Tahoma">Regards,</font></div>
                  <div><font face="Tahoma">Antony Meyn</font></div>
                </div>
              </div>
            </div>
          </div>
        </div>
        <pre wrap=""><fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
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>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Salvatore Loreto, PhD
<a class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
  </body>
</html>

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From oscar.ohlsson@ericsson.com  Mon Apr  2 05:17:13 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: Gregory Maxwell <gmaxwell@juniper.net>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 2 Apr 2012 14:17:09 +0200
Thread-Topic: [rtcweb] Support of SDES in WebRTC
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=20
> -----Original Message-----
> From: Gregory Maxwell [mailto:gmaxwell@juniper.net]=20
> Sent: Thursday, March 29, 2012 11:59 PM
> To: Oscar Ohlsson; I=F1aki Baz Castillo
> Cc: <rtcweb@ietf.org>
> Subject: RE: [rtcweb] Support of SDES in WebRTC
>=20
> Oscar Ohlsson [oscar.ohlsson@ericsson.com] wrote:
> > That's why I wrote "the entire webapp" below. If it was not clear I=20
> > meant that the
> > - main HTML page
> > - all external CSS files, JavaScript files, images, etc
> > - all XmlHttpRequests
> > - all WebSocket connections
> > are protected with TLS. This is obviously verifiable and=20
> it's a feature supported by all modern browsers (no mixed content).
>=20
> Even this is insufficient.
>=20
> First, even if it is in principle possible to provide=20
> adequate cryptographic security inside the JS applications a=20
> great many of them won't (for many reasons). We can't=20
> eliminate inconsistency in the security of webrtc=20
> applications- but we can certainly make it hard to go below
> a certain level of security especially by accident.   In particular,
> if JS applications provide only authentication services (e.g.=20
> by giving them access to hashes of the ephemeral session=20
> keys, but not the session keys themselves) then a=20
> cryptographically-incompetent application can only fail the=20
> user by failing to provide the authentication it promised=20
> over and above the platform baseline capability.

Yes, and that's exactly why DTLS-SRTP is offered by default.


>=20
> With HTTPS applications are currently free to do dumb things=20
> (mixed content, esp scripts) but at least the browser can=20
> detected this and alerts the user with varying degrees of=20
> loudness.  In a SDES/SRTP world the browser will not be=20
> generally able to detect insecure behavior by applications-=20
> creating something of a lemon market for webrtc based=20
> application security. As a resutl its important to narrow the=20
> amount of insecurity possible.

If mixed content is in use then SDES is not available. The user would never=
 be=20
queried since it's unlikely that he can make that kind of decision.

>=20
> This is effectively a repeat of the arguments against=20
> supporting plaintext. If plaintext is supported it will be=20
> commonly used because it's easiest, because users can't=20
> easily tell how insecure their connection is, and because=20
> users often don't have a good feel for the threat model (e.g.=20
> things like firesheep were _very_ surprising to most people)...
> Because of this the platform should provide the highest level=20
> of security that can reasonably be provided in the platform,=20
> and that means (among other things) perfect-forward-secrecy-=20
> while allowing applications to add things like authentication=20
> on top (because while strong ephemerally keyed crypto can be=20
> done very low in the stack, meaningful authentication needs=20
> layer-7/8 hooks).

I don't understand this argument. Are you saying that SDES would be more co=
mmoly used since it's easier? I seriously doubt that. The easiest is to go =
with default which is DTLS-SRTP. Only in the legacy interop case would it b=
e necessary to turn on SDES.

>=20
> There is also the application/library split issue- if a=20
> vulnerability is found in a common negotiation procedure what=20
> secures users better? Updating a few easily identified=20
> browsers / softphone apps?
> or a much larger base of webapps, many of which would be run=20
> by security ignorant/clueless people?  (and at least if the=20
> webserver is clueful but the client operator is not the app=20
> could yell about the insecure client,
> the other way around doesn't work as well).  -   Really, cryptographic
> negotiation is not properly an application feature, it=20
> belongs lower in the stack, and many applications that roll=20
> their own crypto have done a poor job of it.

The application is not performing any real cryptographic operations. It bas=
ically only forwards SDP blobs between the PeerConnection object and the TL=
S channel. So security is to a very large extent handled by lower layers.

>=20
> It's also inadequate on purely technical grounds: Javascript=20
> provides no mechanism for working with mlocked memory,  no=20
> mechanism to ensure that garbage collected data gets=20
> zeroized.  Your crypto app in JS could easily have its long=20
> term keying material pulled out of free ram by malware long=20
> after it runs, or pulled off the disk from swap.
> The breakneck pace of fancy JIT systems makes it seem=20
> unlikely to me that javascript will provide for that any time soon.

I think you're exaggerating here. I don't know the details of how memory ma=
nagment works in JavaScript but if you start considering malware applicatio=
ns and infected devices then there's probably very little we can do. Quotin=
g I.D.ietf-rtcweb-security-arch, Section 3:

"Conversely, if the browser is compromised, then no security guarantees are=
 possible."=20

I would appreciate if you could describe your attack in more detail and exp=
lain the assumptions that you make. I suspect that some parts are left out =
or have been greatly simplified.

Regards,

Oscar

>=20
>=20
>=20
> =

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To: Gregory Maxwell <gmaxwell@juniper.net>, Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Gregory Maxwell
> Sent: Friday, March 30, 2012 5:38 PM
> To: Randell Jesup; rtcweb@ietf.org
> Subject: Re: [rtcweb] SRTP and "marketing"
>=20
>=20
> A point which needs to be emphasized is that undetectable=20
> attacks are not at all the same thing as detectable attacks:=20
> Even when the chance of detection is somewhat low, if the=20
> cost of detection is high the possibility of it can be an=20
> effective deterrent.
>=20

It is definely important to distinguish between detectable and un-detectabl=
e interception. But as I mentioned earlier, I don't think enabling SDES in =
WebRTC will enable un-detectable interception. If both parties open up Tool=
s->PageInfo->Media->session-security and see something like:

Is SDES turned on? Yes
Is this endpoint capable of DTLS-SRTP? Yes

Then they could automatically assume that they're are beeing intercepted. I=
n my opinion this is not that much different from comparing fingerprints.

Regards,

Oscar





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On Mon, Apr 2, 2012 at 3:16 AM, Paul E. Jones <paulej@packetizer.com> wrote=
:
> Silvia,
>
> I believe H.263 would be covered if you license the MPEG-4 Visual "patent
> pool". =A0I might be wrong, but you can certainly get clarification from =
MPEG
> LA. You could also reach out to the list of companies who have claimed IP=
R
> at the ITU (search for ITU IPR database). =A0I think there are about 14
> companies who have IPR claims submitted.

According to this

http://www.nine-9s.com/h-263-codec-software.htm

Lucent is not covered by MPEG-LA, although it is not clear if they
actually claim IPR.

Regards
Marshall

>
> As with H.264, it was a very open process. =A0Those involved are likely t=
he
> only ones with IPR claims. =A0Perhaps not, but given the pervasive use of
> H.263 already on the Internet, one would think if there was some IPR in
> hiding, it would have appeared already.
>
> Paul
>
>> -----Original Message-----
>> From: Silvia Pfeiffer [mailto:silviapfeiffer1@gmail.com]
>> Sent: Monday, April 02, 2012 1:29 AM
>> To: Paul E. Jones
>> Cc: Stephan Wenger; Basil Mohamed Gohar; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Proposal for H.263 baseline codec
>>
>> Note that H.264 isn't even under discussion here.
>>
>> Also, my question on H.263 hasn't been answered yet. I do wonder about t=
he
>> patent situation there!
>>
>> Cheers,
>> Silvia.
>>
>> On Sun, Apr 1, 2012 at 8:56 PM, Paul E. Jones <paulej@packetizer.com>
>> wrote:
>> > Stephan,
>> >
>> >> The most commonly cited timeline for a widely in use technology to be
>> >> "save" from a patent viewpoint, based on equitable defenses such as
>> >> laches (in the US) is six years. =A0In some countries of significant
>> >> size, this time is longer, and in others, equitable defenses do not
>> >> exist. =A0(Very briefly, and perhaps incorrectly put, those equitable
>> >> defenses allow a defendant to argue successfully that a patent cannot
>> >> be enforced as the right holder knew that the patent claim was likely
>> >> being infringed, and did not enforce the patent.).
>> >
>> > Recall that Unisys forced people to pay royalties for using the GIF
>> > file format long after it became widely popular. =A0The company assert=
ed
>> > IPR claims on GIF since it used a compression algorithm to which it
>> > acquired the rights, and it started doing so right near the end of the
>> > 20-year period during which a patent is considered valid.
>> >
>> > In short, I really do not think one should ever assume it might be
>> > safe to use any technology.
>> >
>> >> In addition, as I pointed out in the meeting, the use of a video
>> >> codec created by a body such as MPEG or ITU-T SG16 has the advantage
>> >> of that the patents of all participating players are available at
>> >> least under Reasonable and Non Discriminatory (RAND) terms. =A0This m=
ay
>> >> sound like a Bad Thing if you operate under a business model that
>> >> prevents you to pay anything for patent licenses, but it is surely a
>> >> Good Thing if you are willing to dish out a moderate amount of money
>> >> for a license. =A0RAND recently has gotten teeth, not so much in term=
s
>> >> of the monetary compensation aspect, but in terms of difficulty (if
>> >> not unavailability) to obtain injunctive relive, among others. =A0H.2=
6x
>> >> and the MPEG standards benefit from RAND commitments, VP8, AFAIK, doe=
s
>> not.
>> >
>> > I believe your point here is perhaps worth even more consideration.
>> >
>> > I really know nothing about the IPR that exists or might be claimed on
>> VP8.
>> > That said, I know there has such an incredible amount of work by so
>> > many companies to produce H.264 that I would be utterly surprised to
>> > find that
>> > VP8 does not infringe on something. =A0All of the technology that went
>> > into
>> > H.264 represents only a subset of all of the IPR that exists in the
>> > video coding space.
>> >
>> > It's the rest of the IPR, a bunch of IPR owned by companies who
>> > actually have significant investment in video coding technologies,
>> > that I believe people should worry about. =A0Everyone who worked on
>> > H.264 did so as part of an open standards process, as you mention
>> > above. =A0They spent a lot of time and energy in the process, coming t=
o
>> > an agreement to license the technology that went into H.264. =A0They d=
id
>> > not agree to license technology that did not go into H.264. =A0So, if
>> > one of those participating companies were to subsequently sue over
>> > some IPR used in VP8, I would not dare call them a troll.
>> >
>> > Trolls always exist and may even lay claims to H.264, but I suspect it
>> > would be much harder for a troll to lay claim to H.264. =A0That codec
>> > spent years in development with input from tons of people. =A0If a tro=
ll
>> > were to come in and lay claim to some part of H.264, I suspect there
>> > would be several companies that would stand up and beat them down.
>> > After all, a claim against H.264 would not only present a problem for
>> > the defendant in a lawsuit, but would be an issue for every company
>> > with IPR on H.264. =A0Further, no matter what part of H.264 such a tro=
ll
>> > might decide to pick on, there are a number of world-class engineers
>> > who (as you indicated about H.263) could probably name the person or
>> > persons who contributed the section of text, algorithm, procedure, etc=
.,
>> all of which is covered with a known license.
>> >
>> > VP8 does not have the benefit of such significant peer review and
>> > collaboration, nor does it benefit from any IPR licensing agreements
>> > from legitimate companies holding tons of video coding-related IPR.
>> >
>> > Paul
>> >
>> > _______________________________________________
>> > 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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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-07.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Web Real-Time Communication Use-cases and Requirements
	Author(s)       : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-07.txt
	Pages           : 24
	Date            : 2012-04-02

   This document describes web based real-time communication use-cases.
   Based on the use-cases, the document also derives requirements
   related to the browser, and the API used by web applications to
   request and control media stream services provided by the browser.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-require=
ments-07.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirem=
ents-07.txt


From HKaplan@acmepacket.com  Mon Apr  2 06:02:40 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] Regarding cloning of PeerConnections when multiple answers	are received
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Hi,
I was thinking the tricky part is more about the local transport stuff gett=
ing cloned - i.e., imagine if you're using a TCP-based TURN connection.
But maybe it's a non-issue.

-hadriel


On Mar 29, 2012, at 3:57 AM, Magnus Westerlund wrote:

> WG, as individual
>=20
> As just discussed in the meeting Justin and Hadriel didn't think that
> you could clone PeerConnections properly. I do believe it is possible.
>=20
> If the API provides a method where you can clone a current
> peerConnection you can actually share the transport layer information at
> one end-point for multiple peer connections assuming that the other
> end-points transport addresses are different between the different
> peerConnections being cloned.
>=20
> This works as the complete ICE username is the concatenation of both
> sides, as long as an secondary answer contains a different username and
> has candidates with different transport addresses (address+port). Then
> end-point A receiving the second answer can have the ICE stack accept
> both usernames and if connectivity checks works then you create
> different 5-tuple filter to divide any media streams to the right peer
> connections media handler.
>=20
> When you clone a PC then must allocate the media decoding resources for
> each cloning. And that is something the cloning function may block on.
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> 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


From christer.holmberg@ericsson.com  Mon Apr  2 06:13:46 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 2 Apr 2012 15:13:42 +0200
Thread-Topic: Draft new version: draft-ietf-rtcweb-use-cases-and-requirements
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Hi,

We've submitted a new version of the use-case-and-reqs document.

The main reason for the new version is that the previous version was about =
to expire. The only change is that we have changed the numbering for 3 of t=
he requirements.

Regards,

Christer

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osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:=
//www.w3.org/TR/REC-html40"><head><meta http-equiv=3DContent-Type content=
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nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span lang=3DFI>=
Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DFI><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal>We&#8217;ve submitted a new version of =
the use-case-and-reqs document.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nb=
sp;</o:p></p><p class=3DMsoNormal>The main reason for the new version is th=
at the previous version was about to expire. The only change is that we hav=
e changed the numbering for 3 of the requirements.<o:p></o:p></p><p class=
=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Regards,<o:p></o:p><=
/p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Christer<=
o:p></o:p></p></div></body></html>=

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From stefan.lk.hakansson@ericsson.com  Mon Apr  2 06:55:38 2012
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On 04/02/2012 05:16 AM, Justin Uberti wrote:
>
>
> On Sat, Mar 31, 2012 at 1:41 AM, Stefan Hakansson LK
> <stefan.lk.hakansson@ericsson.com
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>
>     On 03/31/2012 07:17 AM, Stefan Hakansson LK wrote:
>
>         On 03/30/2012 11:39 PM, Randell Jesup wrote:
>
>             On 3/30/2012 4:59 AM, Stefan Hakansson LK wrote:
>
>                 The JS API has deals with MediaStreams (this is what you
>                 send and
>                 receive using PeerConnection from an application
>                 perspective).
>
>                 A browser receiving RTP streams, needs side info to be
>                 able to
>                 assemble those RTP streams into MediaStreams in a
>                 correct way. The
>                 current model is that this is signaled using SDP
>                 exchanges (where
>                 Haralds MSID proposal would tell which MediaStream an
>                 RTP stream
>                 belongs to).
>
>                 As I brought up at the mike yesterday, I think we may
>                 have a race
>                 condition for the responder.
>
>                 For the initiator side browser, this is clear: once an
>                 (PR-)ANSWER is
>                 received, the responder has received the SDP, and hence
>                 can map
>                 incoming RTP streams into MediaStreams.
>
>                 But for the responder side this is less clear to me. Imagine
>                 applications where the responder just mirrors the
>                 initiator - if one
>                 of the parties adds a MediaStream to PeerConnection, the
>                 other end
>                 would add the corresponding MediaStream.
>
>                 This can happen any time in the session, so ICE can very
>                 well be up
>                 and running. One example could be that the data channel
>                 is used for
>                 text chat, when one side clicks a button to start video.
>                 And the
>                 application can have asked for permission to use all
>                 input devices
>                 earlier, so no user interaction may be involved.
>
>                 In this situation the responder's (added) RTP streams
>                 can very well
>                 arrive before the ANSWER if I understand correctly.
>
>
>             Yes.  Just like in SIP.  And so when you send an OFFER (or
>             modified
>             re-OFFER), you must be ready to receive data per that offer
>             even if no
>             ANSWER has been received - just like in SIP.  And if its a
>             re-offer, you
>             need to accept the old, and accept the new (though you could
>             probably
>             use reception of obviously new-OFFER media to turn off
>             decoding/rendering old-OFFER in preparation for the ANSWER).
>
>             The flip side of this is the responder has to infer when the
>             sender
>             switches over to the result of the ANSWER from the media.
>               For example:
>
>             A                                      B
>             <--- H.261 --->
>             re-OFFER(VP8) --->
>             <-- ANSWER(VP8) (delayed in reception)
>             <-----------VP8            (A should infer that B ANSWERed
>             and accepted VP8)
>                 ---------->   H.261
>             <-- ANSWER(VP8) (received)
>             <--------VP8---------->   (B should infer by reception of
>             VP8 that ANSWER
>             was received)
>
>             (Personally, I hate inferences, but without a 3 (or 4) way
>             handshake,
>             you have to).  If you switches of codecs are staged, then
>             this isn't
>             (much) of a problem.  Either leave old codec on the list, or
>             leave it on
>             the list until accept, and then re-OFFER to remove the
>             un-used codec.
>
>
>         I think I understand what you mean, and this would work fine as
>         long as
>         you just switch codecs that are used in already set-up MediaStreams.
>
>         But if A in this case, as part of re-OFFERING the session, not only
>         offers a new codec (VP8) for the already flowing video but also
>         adds a
>         new outgoing video stream (e.g. front cam), and then (without
>         receiving
>         the ANSWER - delayed in reception) starts receiving VP8 video it
>         could
>         not really know if this VP8 video is new video from the
>         responders front
>         cam or just a new codec for the existing (back cam) video from the
>         responder to the sender.
>
>
>     This may have been a very bad example. Probably you can tell them
>     apart on the SSRC. But even so, the A browser won't know what the
>     VP8 stream (if it has an unknown SSRC) represents without receiving
>     the ANSWER.
>
>
> I think this is only an issue if you decide to add streams in the
> ANSWER. But even so, eventually the ANSWER arrives and you can start
> demuxing/decoding appropriately.

Yes, I agree to that it is only an issue if you add streams in the 
answer. Perhaps it is a model we should move away from - but that is the 
model used in the basic examples of the JSEP draft.

I think there is a risk of clipping if you start sending immediately, 
but can only start demuxing/decoding once an answer is received.

>
> Regardless, if the app wants to require some sort of ACK message to
> begin transmission, or perhaps require that the remote side ask for the
> media streams it wants to be sent, I think this could be implemented in
> the app-specific signaling layer; the streams could initially be added
> as inactive, and only changed to sendrecv when the ACK arrives.

To me it is the browser that would need the info to be able give 
something sensible to the app for playout - not the app itself.

>
>
>
>
>
>             One problem is what to do in the switchover window when you
>             might get a
>             mixture of old and new media, especially if you moved them
>             to different
>             ports and so can't count on RTP sequence re-ordering to
>             un-mix them; in
>             the past I dealt with that (and long codec-switch times) by
>             locking out
>             codec changes for a fraction of a second after I do one.
>               Not a huge
>             deal, however.
>
>             My apologies if I've missed something in JSEP; I've been
>             heads-down
>             enough in Data Channels and bring-up that I could have a
>             disconnect here
>             and be saying something silly.
>
>
>         Actually I don't think this is very JSEP related; it is the generic
>         problem that the browser receiving RTP streams need some side
>         info about
>         them before being able to do anything sensible with them.
>
>
>                 I think we need to find a way to handle this. One way is
>                 to add an
>                 "ACK" that indicates to the responder that the initiator
>                 has received
>                 the ANSWER, but I'm not sure that is the best way.
>
>
>             If you need to know that, you need a SIP-style ACK.
>
>
>         As explained, I do think we need to know that.
>
>
>
>         _________________________________________________
>         rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>         https://www.ietf.org/mailman/__listinfo/rtcweb
>         <https://www.ietf.org/mailman/listinfo/rtcweb>
>
>
>     _________________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/rtcweb
>     <https://www.ietf.org/mailman/listinfo/rtcweb>
>
>


From stefan.lk.hakansson@ericsson.com  Mon Apr  2 07:00:49 2012
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Subject: [rtcweb] Clarify "simultaneous video add" of JSEP draft
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Hi Justin,

I'm reading the JSEP draft 
(http://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/?include_text=1), and 
got to section 7.4 which is an example of simultaneous add of video.

I think it is a very good thing to support, as one side adding video 
would not delay the other one (if doing so more or less simultaneously). 
However, I'm not sure I understand the flow in the example.

I could make some guesses on what is happening, but I would rather not 
make a complete fool of myself - could you elaborate a little bit on it?

Br,
Stefan

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Subject: Re: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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--047d7b33cb6cdb7add04bcb469ad
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On Mon, Apr 2, 2012 at 9:02 AM, Hadriel Kaplan <HKaplan@acmepacket.com>wrote:

>
> Hi,
> I was thinking the tricky part is more about the local transport stuff
> getting cloned - i.e., imagine if you're using a TCP-based TURN connection.
> But maybe it's a non-issue.
>
>
This really depends on how the TURN support is implemented, but typically
using the same TCP connection or any other TURN allocation for multiple PC
should be no more of an issue then using same connection for multiple
streams within the PC.
_____________
Roman Shpount

--047d7b33cb6cdb7add04bcb469ad
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<br clear=3D"all"><div class=3D"gmail_quote">On Mon, Apr 2, 2012 at 9:02 AM=
, Hadriel Kaplan <span dir=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepacket=
.com">HKaplan@acmepacket.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">
<br>
Hi,<br>
I was thinking the tricky part is more about the local transport stuff gett=
ing cloned - i.e., imagine if you&#39;re using a TCP-based TURN connection.=
<br>
But maybe it&#39;s a non-issue.<br>
<br></blockquote><div>=A0</div><div>This really depends on how the TURN sup=
port is implemented, but typically using the same TCP connection or any oth=
er TURN allocation for multiple PC should be no more of an issue then using=
 same connection for multiple streams within the PC.<br>
</div></div>_____________<br>Roman Shpount<br>

--047d7b33cb6cdb7add04bcb469ad--

From randell-ietf@jesup.org  Mon Apr  2 10:25:27 2012
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Subject: Re: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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On 4/1/2012 11:21 PM, Justin Uberti wrote:

>     If the API provides a method where you can clone a current
>     peerConnection you can actually share the transport layer information at
>     one end-point for multiple peer connections assuming that the other
>     end-points transport addresses are different between the different
>     peerConnections being cloned.
>
>     This works as the complete ICE username is the concatenation of both
>     sides, as long as an secondary answer contains a different username and
>     has candidates with different transport addresses (address+port). Then
>     end-point A receiving the second answer can have the ICE stack accept
>     both usernames and if connectivity checks works then you create
>     different 5-tuple filter to divide any media streams to the right peer
>     connections media handler.


> Thanks Magnus. This sounds reasonable; it might even be possible to make
> cloning work using the existing API by taking the local description from
> one PC and installing it as the local description for a new PC; the
> browser could recognize that the candidates refer to existing ports and
> behave as you mention above.

Yes, this seems like it would work.

> However, given that this seems to be mostly an edge case, I think we can
> probably save this for v2.

So the question for me is "what scenarios/use-cases does this enable?"

1) Mesh Conferencing

While there are many ways to handle conferencing, if you have a mesh 
(full or partial), you need to create new PeerConnections for existing 
members of the mesh when someone joins.  There are two ways to do this:
a) somehow application-specific tell the new member who to send invites 
to, and then start N PeerConnection calls with some "I'm in the 
conference" token;
b)have them send N separate generic invites to the service provider 
which either forwards it to an existing member, or returns it with "it's 
ok, I've already forwarded invites for it);
c) somehow tell all the existing members to invite the new member; or
d) have the service provider fork the OFFER from the new member to all 
the existing members, and have all of them reply with ANSWERs.  This 
requires PeerConnection cloning to work well.

I strongly prefer d), and it also reduces conference join time (less 
round-trips, at least in theory - perhaps noticeably less by sharing the 
ICE candidate generation).

2) "Hanging" invites at a service provider

You send an OFFER, which is "held" at the service provider and then 
"forked"/forwarded to other clients when the server decides to.

This is of use in cases where there is external state (a game, virtual 
environment ala Second Life, virtual conference hall, etc) where 
something at the server decides when you should actually be connected 
and to whom (or multiple people).  Example (applies to several of 
these): as you move around a game/virtual-environment, your app is 
connected to other nearby players when they're in audible range.  This 
could be done by having the server tell your client to generate an OFFER 
(with attendant ICE discovery delay) on demand for each other person who 
comes in range, but that would be sub-optimal and again involve setup 
delay.  It also allows permission to be dealt-with up front and not 
delay actual connection.

I imagine there are others, but those are two which I think are pretty 
likely uses (we've been talking mesh conferences since the beginning, 
and this sort of environmental audio/teamspeak type stuff is an obvious 
use).  Our game-table demo showed chess, but other uses would involve 
more players (4 for bridge, etc) where you might want a full-mesh 
conference).

So, I'd really like to push to support cloning if possible.

-- 
Randell Jesup
randell-ietf@jesup.org

From salvatore.loreto@ericsson.com  Mon Apr  2 11:33:34 2012
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On 3/30/12 11:43 PM, Randell Jesup wrote:
>>> >>  2. Multihoming and interface switching: I don't suggest going for the
>>> >>  multipath support on the SCTP level. I think it would be best to deal
>>> >>  with this through ICE in the same way as for RTP, i.e. adding and
>>> >>  removing candidates as interfaces become available or go. Or let the
>>> >>  application handle it by creating a new data channel and continue from
>>> >>  the breakpoint. The most important requirement is that the application
>>> >>  gets notified about these changes downstrairs so it can act.
>> >  that is exact the proposal in the data-channel draft.
>> >  Anyway you can not use multihoming because of the fact that SCTP runs on
>> >  top of DTLS
> Correct, though there was some in-room discussion of running it over
> multiple DTLS connections to support smooth handoff across interfaces
> (think mobile and 4g<->WiFi switchover).
definitely, that is also something else worth to investigate more deeply


-- 
Salvatore Loreto, PhD
www.sloreto.com


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On Mar 28, 2012 2:08 PM, "Basil Mohamed Gohar" <
abu_hurayrah@hidayahonline.org> wrote:
>
> On 03/28/2012 04:41 PM, Roman Shpount wrote:
> > My main objection is that if an application developer does not take
> > care to develop a secure application, nothing you can do on the
> > standard side will make it a secure application. If I am building a
> > public voice blog that records a voice message that anybody can listen
> > to on the web site security is not needed. My assumption is that a
> > fair number of applications would be like this. So for such
> > applications this is an unnecessary feature.
> >
> > WebRTC will not exist in vacuum. It will communicate with other
> > systems. It is not limited to old SIP devices. It can be something new
> > like server side speech recognition that is integrated with web
> > application. For such application extra code and interop requirements
> > to support security will represent a real and significant cost. Any
> > requirement, unless absolutely necessary will create barriers to entry
> > for new applications. I would like to avoid as many of those as
> > possible.
> > _____________
> > Roman Shpount
> Roman,
>
> You make a lot of good points.  However, the inverse is true as well -
> namely, that is if encryption is not mandated, most implementations will
> likely leave it out,

That means SRTP unnecessary to fullfill major practical usage cases.

The decision to use telnet or ssh for remote desktop should be made by IT
department, not by standard committee.

- jesse

and adoption of secured communications would be
> stifled even longer.  I cannot speak about the implementation
> difficulties, but I can speak from the user side that most people will
> remain ignorant of the underlying technology and not know enough to
> demand nor enable a feature if it is optional to implement and/or use.
>
> As WebRTC is a new standard, requiring encryption will ensure that, at
> least going forward, the important concept of encryption is widely
> adopted correctly from the beginning.  Tacking it on later, no matter
> how much it is emphasized, will be difficult or impossible.
>
> The scope of WebRTC is broad enough to consider that we need to think
> about what's best going forward with regards to its implementation.
> Security by default is one of the best practices in general, the support
> from the browser community and others that are behind it will definitely
> ensure that adoption is widespread enough to make it easy enough to
> integrate into existing systems, as free software solutions will become
> available shortly after the standard emerges.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--e89a8fb1f350d33d2104bcbaa26b
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<p><br>
On Mar 28, 2012 2:08 PM, &quot;Basil Mohamed Gohar&quot; &lt;<a href=3D"mai=
lto:abu_hurayrah@hidayahonline.org">abu_hurayrah@hidayahonline.org</a>&gt; =
wrote:<br>
&gt;<br>
&gt; On 03/28/2012 04:41 PM, Roman Shpount wrote:<br>
&gt; &gt; My main objection is that if an application developer does not ta=
ke<br>
&gt; &gt; care to develop a secure application, nothing you can do on the<b=
r>
&gt; &gt; standard side will make it a secure application. If I am building=
 a<br>
&gt; &gt; public voice blog that records a voice message that anybody can l=
isten<br>
&gt; &gt; to on the web site security is not needed. My assumption is that =
a<br>
&gt; &gt; fair number of applications would be like this. So for such<br>
&gt; &gt; applications this is an unnecessary feature.<br>
&gt; &gt;<br>
&gt; &gt; WebRTC will not exist in vacuum. It will communicate with other<b=
r>
&gt; &gt; systems. It is not limited to old SIP devices. It can be somethin=
g new<br>
&gt; &gt; like server side speech recognition that is integrated with web<b=
r>
&gt; &gt; application. For such application extra code and interop requirem=
ents<br>
&gt; &gt; to support security will represent a real and significant cost. A=
ny<br>
&gt; &gt; requirement, unless absolutely necessary will create barriers to =
entry<br>
&gt; &gt; for new applications. I would like to avoid as many of those as<b=
r>
&gt; &gt; possible.<br>
&gt; &gt; _____________<br>
&gt; &gt; Roman Shpount<br>
&gt; Roman,<br>
&gt;<br>
&gt; You make a lot of good points. =A0However, the inverse is true as well=
 -<br>
&gt; namely, that is if encryption is not mandated, most implementations wi=
ll<br>
&gt; likely leave it out, </p>
<p>That means SRTP unnecessary to fullfill major practical usage cases. </p=
>
<p>The decision to use telnet or ssh for remote desktop should be made by I=
T department, not by standard committee.</p>
<p>- jesse</p>
<p>and adoption of secured communications would be<br>
&gt; stifled even longer. =A0I cannot speak about the implementation<br>
&gt; difficulties, but I can speak from the user side that most people will=
<br>
&gt; remain ignorant of the underlying technology and not know enough to<br=
>
&gt; demand nor enable a feature if it is optional to implement and/or use.=
<br>
&gt;<br>
&gt; As WebRTC is a new standard, requiring encryption will ensure that, at=
<br>
&gt; least going forward, the important concept of encryption is widely<br>
&gt; adopted correctly from the beginning. =A0Tacking it on later, no matte=
r<br>
&gt; how much it is emphasized, will be difficult or impossible.<br>
&gt;<br>
&gt; The scope of WebRTC is broad enough to consider that we need to think<=
br>
&gt; about what&#39;s best going forward with regards to its implementation=
.<br>
&gt; Security by default is one of the best practices in general, the suppo=
rt<br>
&gt; from the browser community and others that are behind it will definite=
ly<br>
&gt; ensure that adoption is widespread enough to make it easy enough to<br=
>
&gt; integrate into existing systems, as free software solutions will becom=
e<br>
&gt; available shortly after the standard emerges.<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>

--e89a8fb1f350d33d2104bcbaa26b--

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Mar 28, 2012 2:08 PM, "Basil Mohamed Gohar" <abu_hurayrah@hidayahonline.org>
wrote:

>
> That means SRTP unnecessary to fullfill major practical usage cases.
>
> The decision to use telnet or ssh for remote desktop should be made by IT
department, not by standard committee.

The more than 500 millions of Facebook users are end users, not an IT
department.

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<p>Mar 28, 2012 2:08 PM, &quot;Basil Mohamed Gohar&quot; &lt;<a href="mailto:abu_hurayrah@hidayahonline.org">abu_hurayrah@hidayahonline.org</a>&gt; wrote:</p>
<p>&gt;<br>
&gt; That means SRTP unnecessary to fullfill major practical usage cases.<br>
&gt;<br>
&gt; The decision to use telnet or ssh for remote desktop should be made by IT department, not by standard committee.</p>
<p>The more than 500 millions of Facebook users are end users, not an IT department.</p>

--20cf307d00f0d0616204bcbac9c8--

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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Consensus call regarding media security
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On Mon, Apr 2, 2012 at 7:45 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> The more than 500 millions of Facebook users are end users, not an IT
> department.
>

If Facebook does not want to provide security to its users, standards
committee will not make it secure.
_____________
Roman Shpount

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<br clear=3D"all"><br><div class=3D"gmail_quote">On Mon, Apr 2, 2012 at 7:4=
5 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@alia=
x.net">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">
The more than 500 millions of Facebook users are end users, not an IT depar=
tment.
<br></blockquote></div><br>If Facebook does not want to provide security to=
 its users, standards committee will not make it secure.<br>_____________<b=
r>Roman Shpount<br>
<br>

--f46d040f9bae5c423604bcbb9ac5--

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On Wed, Mar 28, 2012 at 5:08 PM, Basil Mohamed Gohar <
abu_hurayrah@hidayahonline.org> wrote:

> You make a lot of good points.  However, the inverse is true as well -
> namely, that is if encryption is not mandated, most implementations will
> likely leave it out, and adoption of secured communications would be
> stifled even longer.  I cannot speak about the implementation
> difficulties, but I can speak from the user side that most people will
> remain ignorant of the underlying technology and not know enough to
> demand nor enable a feature if it is optional to implement and/or use.
>
>
If application developer does not care about security nothing we do will
force the application to be secure. For instance, if SDES-SRTP is allowed,
I can implement something that will communicate with it with less then 1000
lines of code. I will drop the replay protection, real random key
generation and will just leave AES-CM and check sum calculation. This will
still successfully communicate with WebRTC end points but will not be
secure by a long stretch. We can provide the tools to secure the
communications, but unless application is properly developed, it will only
go through the motions of encryption without achieving any real results.
So, as a result, allowing non-encrypted communications will actually result
in more secure ecosystem. Where people who need security will support it,
and people who do not need security will clearly show that secure
communications are not implemented by their application.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 5:08 P=
M, Basil Mohamed Gohar <span dir=3D"ltr">&lt;<a href=3D"mailto:abu_hurayrah=
@hidayahonline.org">abu_hurayrah@hidayahonline.org</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">
You make a lot of good points. =A0However, the inverse is true as well -<br=
>
namely, that is if encryption is not mandated, most implementations will<br=
>
likely leave it out, and adoption of secured communications would be<br>
stifled even longer. =A0I cannot speak about the implementation<br>
difficulties, but I can speak from the user side that most people will<br>
remain ignorant of the underlying technology and not know enough to<br>
demand nor enable a feature if it is optional to implement and/or use.<br>
<br></blockquote></div><br>If application developer does not care about sec=
urity nothing we do will force the application to be secure. For instance, =
if SDES-SRTP is allowed, I can implement something that will communicate wi=
th it with less then 1000 lines of code. I will drop the replay protection,=
 real random key generation and will just leave AES-CM and check sum calcul=
ation. This will still successfully communicate with WebRTC end points but =
will not be secure by a long stretch. We can provide the tools to secure th=
e communications, but unless application is properly developed, it will onl=
y go through the motions of encryption without achieving any real results. =
So, as a result, allowing non-encrypted communications will actually result=
 in more secure ecosystem. Where people who need security will support it, =
and people who do not need security will clearly show that secure communica=
tions are not implemented by their application.<br>
_____________<br>Roman Shpount<br>
<br>

--e89a8ff1cce4afb4bc04bcbbba6f--

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Subject: [rtcweb] Which servers to trust (Re: Consensus call regarding media security)
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Changing the subject, since this message is not a position on the 
consensus call.

On 03/29/2012 07:31 PM, Fabio Pietrosanti (naif) wrote:
> On 3/29/12 7:02 PM, Ravindran, Parthasarathi wrote:
>> WebRTC trust model has to be considered as one of the main factor for deciding the key mechanism. AFAIK, SDES does not fit into WebRTC as Dr.Evil HTTPS RTCWeb server must be trusted in case of SDES. There is no means to track or analyze whether Dr.Evil involves in monitoring or recording or terminate the media traffic.  It will be good in case whoever advocate for SDES explain how SDES fits within WebRTC trust model.
> Sure!
>
> From:
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/?include_text=1
>
> "   The basic assumption of this architecture is that network resources
>     exist in a hierarchy of trust, rooted in the browser, which serves as
>     the user's TRUSTED COMPUTING BASE (TCB).  Any security property which
>     the user wishes to have enforced must be ultimately guaranteed by the
>     browser (or transitively by some property the browser verifies)."
>
> So, it means that if the browser already have a hierarchy of trust to
> use TLS for HTTPS, then SDES-SRTP will inherit the trust-properties of
> the HTTPS website from which it's loaded.
Ahem.... SDES-SRTP will "inherit" the property that as long as the HTTPS 
trust chain is intact, only attacks where the HTTPS website, or entities 
that the HTTPS website chooses to reveal the key to (voluntarily or 
involuntarily) can listen in on the call.

(The plenary at the IETF made interesting background for this claim.)
> It seems to me quite easy to fit SDES-SRTP into the browser model, as it
> allow you to assure that the communication path between the client and
> the server is secure.
>
> Do you expect WebRTC to be only peer-to-peer/client-to-client?
>
> I sincerly expect *a lot* of communications to goes trough SIP/RTP media
> proxy for security purpose, for billing purposes, for value added
> service purpose.
In the cases where a gateway needs to decrypt the packets, the gateway 
is the logical peer with the browser in the (DTLS-)SRTP trust relationship.
> SDES-SRTP provide a very reliable and simple way to let a WebRTC peer to
> establish security with the server, assuming that it already have
> established security trough HTTPS/TLS that's a consolidate trust method.
The term "the server" is a fallacy. The Web server and the media gateway 
(if there is one) are likely not the same server, and may even be 
operated by different entities.
SDES-SRTP forces you to trust both.

>


From ibc@aliax.net  Tue Apr  3 04:55:59 2012
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Subject: Re: [rtcweb] Which servers to trust (Re: Consensus call regarding media security)
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2012/4/3 Harald Alvestrand <harald@alvestrand.no>:
>> SDES-SRTP provide a very reliable and simple way to let a WebRTC peer to
>> establish security with the server, assuming that it already have
>> established security trough HTTPS/TLS that's a consolidate trust method.
>
> The term "the server" is a fallacy. The Web server and the media gateway =
(if
> there is one) are likely not the same server, and may even be operated by
> different entities.
> SDES-SRTP forces you to trust both.


If you trust the Web server (i.e. due to HTTPS usage with a valid
server certificate) then you will also trust that the Web server will
not signal your WebRTC communication to a malicious destination, am I
wrong?

Don't take me wrong but, which kind of security obsession are we
trying to satisfy in rtcweb? a media communication is not more
important than a web access to my back website in which I enter my
credit card PIN. Does IETF define "security standards" for POS ("Point
of sale terminal") for making a bank payment via a 3rd web
(e-commerce)? AFAIK not.

If the Web server (assuming HTTPS) is trusted and SDES-SRTP used, we
should trust the communication. If it fails that is because the Web
server has been attacked. If that occurs, it's really worse the case
in which my bank website has been attacked (I'm giving my credit card
PIN to the attacker).


Regards.


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

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2012/4/3 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> Don't take me wrong but, which kind of security obsession are we
> trying to satisfy in rtcweb? a media communication is not more
> important than a web access to my back website

typo: I meant "bank website"

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

From randell-ietf@jesup.org  Tue Apr  3 05:35:16 2012
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Subject: Re: [rtcweb] Which servers to trust (Re: Consensus call regarding media security)
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On 4/3/2012 7:55 AM, IÃ±aki Baz Castillo wrote:
> 2012/4/3 Harald Alvestrand<harald@alvestrand.no>:
>>> SDES-SRTP provide a very reliable and simple way to let a WebRTC peer to
>>> establish security with the server, assuming that it already have
>>> established security trough HTTPS/TLS that's a consolidate trust method.
>>
>> The term "the server" is a fallacy. The Web server and the media gateway (if
>> there is one) are likely not the same server, and may even be operated by
>> different entities.
>> SDES-SRTP forces you to trust both.
>
>
> If you trust the Web server (i.e. due to HTTPS usage with a valid
> server certificate) then you will also trust that the Web server will
> not signal your WebRTC communication to a malicious destination, am I
> wrong?

Partly.  First, as you mention, the site can be hacked.  Another issue 
is that you may NOT trust the website to avoid tapping your 
communication (and in fact they may be legally obligated to do so, and 
have no choice in the matter).  Also, as shown by Diginotar, website 
certs can be forged transparently if a CA is hacked.  Those are both 
real-world cases where you cannot trust the site you connect to via 
https to secure your data (in addition to "the site was hacked", even if 
they are "trustworthy".

> Don't take me wrong but, which kind of security obsession are we
> trying to satisfy in rtcweb? a media communication is not more
> important than a web access to my back website in which I enter my
> credit card PIN. Does IETF define "security standards" for POS ("Point
> of sale terminal") for making a bank payment via a 3rd web
> (e-commerce)? AFAIK not.

No, but other organizations define such standards in great detail - and 
still there are weekly breaches of security that require credit card 
companies to send out thousands or millions of replace cards & numbers.

And a site providing media communication will not have the ultra-strict 
security reviews and audits that come with processing credit card info, 
and so is much more likely to be hacked.

> If the Web server (assuming HTTPS) is trusted and SDES-SRTP used, we
> should trust the communication. If it fails that is because the Web
> server has been attacked. If that occurs, it's really worse the case
> in which my bank website has been attacked (I'm giving my credit card
> PIN to the attacker).

It can be far worse - most people are given guarantees about how much 
they can lose (often $0) - a hassle is all usually; sometimes a big 
hassle, but not harmful personally.  Media interception (especially 
video) can cause horrible repercussions in some cases - blackmail, 
embarrassment, scandal, and in countries like Iran or cases like the 
Arab Spring, jail or death.


-- 
Randell Jesup
randell-ietf@jesup.org

From harald@alvestrand.no  Tue Apr  3 05:58:58 2012
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] Identity and PSTN gateways
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One thing that has come up repeatedly in the discussion is the claim 
that "you can't have a verified identity when you talk to someone via a 
telephone gateway" (and therefore <insert your favourite security 
mechanism here> is not needed / not an added benefit / other claim).

I think this is a fallacy.

Sure, as people have commented numerous times, telephone numbers are 
identities; they're being used as such every time someone prints them on 
a business card or a billboard.

When you're connecting via a gateway to the PSTN, the gateway operator 
gives you a guarantee that you're being connected to the right person; 
that's what gateways are for.

This makes for a fairly simple mapping to the "identity / identity 
provider" model we've been bandying about for the "full-blown" IdP / 
endpoint case:

The identity is the telephone number.
The identity provider (one of many possible ones for the number) is the 
gateway operator.

Thus - if you call a telephone number via a gateway, you would perform a 
DTLS key exchange with the gateway, and an identity verification 
exchange with the gateway operator; you would then guarantee that the 
gateway operator vouches for this being a legitimate gateway function 
that you can reach for that number.

That's just about the best guarantee you can get when talking to the 
telephone system. But if we're using the IdP + DTLS-SRTP version, the 
exchange guarantees you that:
a) nobody is listening in between you and the gateway (even if they 
snooped your signalling)
b) the gateway operator vouches for the gateway being the right gateway 
to reach that number

Seems like a little bit better than what you get with SDES. Only a little.

                        Harald



From oej@edvina.net  Tue Apr  3 06:05:31 2012
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Subject: Re: [rtcweb] Identity and PSTN gateways
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3 apr 2012 kl. 14:58 skrev Harald Alvestrand:

> One thing that has come up repeatedly in the discussion is the claim =
that "you can't have a verified identity when you talk to someone via a =
telephone gateway" (and therefore <insert your favourite security =
mechanism here> is not needed / not an added benefit / other claim).
>=20
> I think this is a fallacy.
>=20
> Sure, as people have commented numerous times, telephone numbers are =
identities; they're being used as such every time someone prints them on =
a business card or a billboard.
>=20
> When you're connecting via a gateway to the PSTN, the gateway operator =
gives you a guarantee that you're being connected to the right person; =
that's what gateways are for.
>=20
> This makes for a fairly simple mapping to the "identity / identity =
provider" model we've been bandying about for the "full-blown" IdP / =
endpoint case:
>=20
> The identity is the telephone number.
> The identity provider (one of many possible ones for the number) is =
the gateway operator.
>=20
> Thus - if you call a telephone number via a gateway, you would perform =
a DTLS key exchange with the gateway, and an identity verification =
exchange with the gateway operator; you would then guarantee that the =
gateway operator vouches for this being a legitimate gateway function =
that you can reach for that number.
>=20
> That's just about the best guarantee you can get when talking to the =
telephone system. But if we're using the IdP + DTLS-SRTP version, the =
exchange guarantees you that:
> a) nobody is listening in between you and the gateway (even if they =
snooped your signalling)
> b) the gateway operator vouches for the gateway being the right =
gateway to reach that number
>=20
> Seems like a little bit better than what you get with SDES. Only a =
little.

Now we will have to separate "PSTN-emulating" gateways that accept calls =
to all phone numbers but play a prompt saying "You gotta be kidding me - =
calling a phone number?" from REAL gateways that have a connection to =
the PSTN world.=20

Will guys connecting with SS7 have a certificate signed by the ITU as a =
"TRUE" PSTN provider and the voip guy in the basement next door just =
have a "Best effort fourth-tier PSTN service" certificate?

I think that any identity of any PSTN gateway just identifies the =
gateway as a server. Not as a service.

/O=

From ibc@aliax.net  Tue Apr  3 06:13:41 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Apr 2012 15:13:20 +0200
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Subject: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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Hi all,

I've made two "pictures" showing WebRTC and SIP interop for two cases:

1) SDES-SRTP is allowed in WebRTC:
      http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.png

2) Just DTLS-ETK-SRTP is allowed in WebRTC [*]
      http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png

[*] slides 30-35 in
http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf


For those claiming to mandate *just* DTLS-EKT-SRTP in WebRTC, please
see the *cost* of such a decision, and also:

- Thanks for requiring a super Signaling+Media B2BUA/SBC in WebRTC/SIP
interop scenarios. Some vendors will be very happy and will become
very rich. Such a super device (also a DTLS to SDES conversor,
including DTLS key updates to re-INVITE) will be "a bit"... expensive.

- Thanks for disallowing *pure* SIP protocol usage (and instead
requiring SIP B2BUAs/SBCs or custom WebRTC signaling to SIP conversion
gateways). WebRTC is supposed to let the signaling protocol up to the
application, but pure SIP protocol will not be possible since a SIP
B2BUA/SBC is required, and those devices always break/limit the SIP
protocol (*always*).


So IMHO, option 2 ("just DTLS-EKT-SRTP is allowed in WebRTC") is The Barrie=
r.


Best regards.


PS: Note that the same is true for WebRTC/XMPP-Jingle interop.


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

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Subject: Re: [rtcweb] Identity and PSTN gateways
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2012/4/3 Harald Alvestrand <harald@alvestrand.no>:
> Thus - if you call a telephone number via a gateway, you would perform a
> DTLS key exchange with the gateway

A super gateway like in in this figure:

  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png

Taking into account that SIP-PSTN gateways into most of the SIP PSTN
providers don't support SRTP or ICE, how long will take the scenario
you mean to become real and widely adopted/implemented? IMHO 10 years.

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

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Subject: Re: [rtcweb] Identity and PSTN gateways
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On 04/03/2012 03:31 PM, IÃ±aki Baz Castillo wrote:
> 2012/4/3 Harald Alvestrand<harald@alvestrand.no>:
>> Thus - if you call a telephone number via a gateway, you would perform a
>> DTLS key exchange with the gateway
> A super gateway like in in this figure:
>
>    http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png
>
> Taking into account that SIP-PSTN gateways into most of the SIP PSTN
> providers don't support SRTP or ICE, how long will take the scenario
> you mean to become real and widely adopted/implemented? IMHO 10 years.
>
Real: 1 year.

Widely adopted: Just in time to see the PSTN shut down because nobody 
uses it any more :-)



From ibc@aliax.net  Tue Apr  3 06:53:04 2012
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Subject: Re: [rtcweb] Identity and PSTN gateways
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2012/4/3 Harald Alvestrand <harald@alvestrand.no>:
> Real: 1 year.
>
> Widely adopted: Just in time to see the PSTN shut down because nobody use=
s
> it any more :-)

Hope you are right :)

Anyhow I still want to interoperate with SIP devices, regardless they
are not boring PSTN gateways ;)

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

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It seems the same problem exists when a TURN relay is involved in an inter-=
browser call. The endpoints cannot verify each other's identity directly. T=
hey need to trust the relay to interconnect them and not to do anything evi=
l, such as snooping.

Huilan

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
> Olle E. Johansson
> Sent: Tuesday, April 03, 2012 9:05 AM
> To: Harald Alvestrand
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Identity and PSTN gateways
>=20
>=20
> 3 apr 2012 kl. 14:58 skrev Harald Alvestrand:
>=20
> > One thing that has come up repeatedly in the discussion is the claim th=
at
> "you can't have a verified identity when you talk to someone via a
> telephone gateway" (and therefore <insert your favourite security mechani=
sm
> here> is not needed / not an added benefit / other claim).
> >
> > I think this is a fallacy.
> >
> > Sure, as people have commented numerous times, telephone numbers are
> identities; they're being used as such every time someone prints them on =
a
> business card or a billboard.
> >
> > When you're connecting via a gateway to the PSTN, the gateway operator
> gives you a guarantee that you're being connected to the right person;
> that's what gateways are for.
> >
> > This makes for a fairly simple mapping to the "identity / identity
> provider" model we've been bandying about for the "full-blown" IdP /
> endpoint case:
> >
> > The identity is the telephone number.
> > The identity provider (one of many possible ones for the number) is the
> gateway operator.
> >
> > Thus - if you call a telephone number via a gateway, you would perform =
a
> DTLS key exchange with the gateway, and an identity verification exchange
> with the gateway operator; you would then guarantee that the gateway
> operator vouches for this being a legitimate gateway function that you ca=
n
> reach for that number.
> >
> > That's just about the best guarantee you can get when talking to the
> telephone system. But if we're using the IdP + DTLS-SRTP version, the
> exchange guarantees you that:
> > a) nobody is listening in between you and the gateway (even if they
> snooped your signalling)
> > b) the gateway operator vouches for the gateway being the right gateway
> to reach that number
> >
> > Seems like a little bit better than what you get with SDES. Only a litt=
le.
>=20
> Now we will have to separate "PSTN-emulating" gateways that accept calls =
to
> all phone numbers but play a prompt saying "You gotta be kidding me -
> calling a phone number?" from REAL gateways that have a connection to the
> PSTN world.
>=20
> Will guys connecting with SS7 have a certificate signed by the ITU as a
> "TRUE" PSTN provider and the voip guy in the basement next door just have=
 a
> "Best effort fourth-tier PSTN service" certificate?
>=20
> I think that any identity of any PSTN gateway just identifies the gateway
> as a server. Not as a service.
>=20
> /O
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ibc@aliax.net  Tue Apr  3 07:24:12 2012
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References: <4F732531.2030208@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E221877@inba-mail01.sonusnet.com> <4F749C82.4070305@infosecurity.ch> <4F7ACC96.90206@alvestrand.no> <CALiegf=jJ6SfQhbxPXKdDDKqp7bOrpRNVE=RfBs8Ah8zqy9ftQ@mail.gmail.com> <4F7AEDB6.8000907@jesup.org>
From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Apr 2012 16:23:50 +0200
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Which servers to trust (Re: Consensus call regarding media security)
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2012/4/3 Randell Jesup <randell-ietf@jesup.org>:
> On 4/3/2012 7:55 AM, I=C3=B1aki Baz Castillo wrote:
>> If you trust the Web server (i.e. due to HTTPS usage with a valid
>> server certificate) then you will also trust that the Web server will
>> not signal your WebRTC communication to a malicious destination, am I
>> wrong?
>
>
> Partly. =C2=A0First, as you mention, the site can be hacked.

If the site is hacked then it does not matter whether SDES or DTLS is
being used. Both can be hacked as demostrated in some draft (whose
name cannot remember now).


> Another issue is that you may NOT trust the website to avoid tapping your=
 communication

The communication is not just the RTP, but also the signaling. If my
wife (if I had) realizes that yesterday night I called to my
ex-girlfriend, that's important enough for me (regardless my wife does
not know what I talked with her). So what? no servers? no proxies?
pure P2P for all?


Also, as said in other mail (by other person in this maillist), if you
don't trust the web server, then we must also drop TURN from WebRTC
since TURN URI's are provided by the web server (and it can be hacked
and so...).


Regards.


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

From martin.thomson@gmail.com  Tue Apr  3 08:18:04 2012
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Subject: Re: [rtcweb] Identity and PSTN gateways
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On 3 April 2012 14:58, Harald Alvestrand <harald@alvestrand.no> wrote:
> The identity is the telephone number.
> The identity provider (one of many possible ones for the number) is the
> gateway operator.

The difference here is that the identity provider is not making an
authoritative statement about a namespace that it controls.  That
means that you need to trust the gateway provider.  Third-party,
non-authoritative assertions of this sort require an additional layer
of trust, which adds to their fragility.

When alvestrand.no asserts that harald@alvestrand.no is who he says he
is, then we are able to rely on that assertion because we have
identified alvestrand.no.  By definition, alvestrand.no controls that
namespace.

We could apply the same policy to a gateway and gain a lower value
assertion than the one you seek.  That is,
+1-234-555-6789@alvestrand.no is a valid identifier within the
namespace that alvestrand.no controls, but it is fundamentally
different to +1-234-555-6789@gateway.example.com.

The only problem is that I don't know how to explain this distinction
to my grandmother any more than I could explain that
john.smith@example.com and john.smith@example.org are different
people.

> That's just about the best guarantee you can get when talking to the
> telephone system. But if we're using the IdP + DTLS-SRTP version, the
> exchange guarantees you that:
> a) nobody is listening in between you and the gateway (even if they snooped
> your signalling)
> b) the gateway operator vouches for the gateway being the right gateway to
> reach that number

The problem here is that it requires that you trust the gateway
provider in order to believe b).

> Seems like a little bit better than what you get with SDES. Only a little.

If you can get b), then it's arguably better.  If not, then you get
the same level of assurance, though eavesdropping requires insertion
of a media gateway (which is made easier with EKT).

--Martin

From martin.thomson@gmail.com  Tue Apr  3 08:21:08 2012
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On 3 April 2012 16:11, Lu, Hui-Lan (Huilan)
<huilan.lu@alcatel-lucent.com> wrote:
> It seems the same problem exists when a TURN relay is involved in an inter-browser call. The endpoints cannot verify each other's identity directly. They need to trust the relay to interconnect them and not to do anything evil, such as snooping.

TURN relays UDP packets.  Since DTLS operates at a higher layer, it
passes through the relay unmodified.  TURN isn't a problem.

Sure, the relay can copy packets, but it is no better off than any
other on-path attacker.

--Martin

From pravindran@sonusnet.com  Tue Apr  3 08:30:37 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Apr 2012 17:50:18 +0200
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Subject: Re: [rtcweb] Identity and PSTN gateways
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2012/4/3 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> Inaki,
>
> I remember that you have mentioned that "WebRTC is not SIP" and "IMHO we =
should stop talking about SIP in this WG" and your related mail thread is h=
ttp://www.ietf.org/mail-archive/web/rtcweb/current/msg02744.html. =C2=A0Of =
course, lot of +1 for your mail ;-)

Sure I remember :)

And I still say the same, sure. Security is a need for WebRTC and
interop with SIP should not make it worse. But IMHO SDES-SRTP (with
the adition of secure signaling HTTPS/WSS) is a good solution, the
very same for ICE which is ***poorly*** implemented in SIP world.

Of course interoperability between WebRTC and SIP is a good point, and
that's a good reason to allow SDES-SRTP.


> AFAIK in WebRTC, SRTP-DTLS makes more sense. =C2=A0Having said that, you =
can prove web folks that SRTP-DTLS is not suitable for web and SDES is bett=
er :-)

I don't say that SRTP-DTLS is bad, not at all. I agree that it MUST be
the preferred option for a WebRTC communication.



> Apart from this, I wish single secured RTP key-exchange mechanism for Web=
RTC as it reduces the number of combination in WebRTC gateways design as we=
ll.

You probably would be happy building the
Super-SIP-Media-B2BUA-DTLS-EKT-SDES-ICE-Capable-WebRTC-Gateway, but
not me ;)


Regards.



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

From roman@telurix.com  Tue Apr  3 08:51:17 2012
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Date: Tue, 3 Apr 2012 11:51:13 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Identity and PSTN gateways
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--e89a8fb208dcc6ce7004bcc84696
Content-Type: text/plain; charset=ISO-8859-1

On Tue, Apr 3, 2012 at 9:05 AM, Olle E. Johansson <oej@edvina.net> wrote:

>
> Now we will have to separate "PSTN-emulating" gateways that accept calls
> to all phone numbers but play a prompt saying "You gotta be kidding me -
> calling a phone number?" from REAL gateways that have a connection to the
> PSTN world.
>
> Will guys connecting with SS7 have a certificate signed by the ITU as a
> "TRUE" PSTN provider and the voip guy in the basement next door just have a
> "Best effort fourth-tier PSTN service" certificate?
>
> I think that any identity of any PSTN gateway just identifies the gateway
> as a server. Not as a service.
>
> I agree with you that you can only identify the gateway. Above this, I
think the whole discussion is pointless since there are no security
guarantees within PSTN. A million of people can be listening in. You can be
connected to a completely different number then the one you've dialed due
to LNP, call routing rules, call forwarding, or anything else. If you are
dialing internationally your traffic often goes over unsecured public
internet. So far, 99.999% of all phone calls were unsecured, tapped into,
recorded and listen by anybody who possessed even the moderate desire to do
so. If you start talking about calls coming from PSTN, you have even less
guarantees about accuracy of the caller ID information. You are currently
trying to secure the edge and provide identity on top of this mess.
_____________
Roman Shpount

--e89a8fb208dcc6ce7004bcc84696
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<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Apr 3, 2012 at 9:05 AM=
, Olle E. Johansson <span dir=3D"ltr">&lt;<a href=3D"mailto:oej@edvina.net"=
>oej@edvina.net</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">
<br>
Now we will have to separate &quot;PSTN-emulating&quot; gateways that accep=
t calls to all phone numbers but play a prompt saying &quot;You gotta be ki=
dding me - calling a phone number?&quot; from REAL gateways that have a con=
nection to the PSTN world.<br>

<br>
Will guys connecting with SS7 have a certificate signed by the ITU as a &qu=
ot;TRUE&quot; PSTN provider and the voip guy in the basement next door just=
 have a &quot;Best effort fourth-tier PSTN service&quot; certificate?<br>

<br>
I think that any identity of any PSTN gateway just identifies the gateway a=
s a server. Not as a service.<br>
<font color=3D"#888888"></font><br></blockquote><div>I agree with you that =
you can only identify the gateway. Above this, I think the whole discussion=
 is pointless since there are no security guarantees within PSTN. A million=
 of people can be listening in. You can be connected to a completely differ=
ent number then the one you&#39;ve dialed due to LNP, call routing rules, c=
all forwarding, or anything else. If you are dialing internationally your t=
raffic often goes over unsecured public internet. So far, 99.999% of all ph=
one calls were unsecured, tapped into, recorded and listen by anybody who p=
ossessed even the moderate desire to do so. If you start talking about call=
s coming from PSTN, you have even less guarantees about accuracy of the cal=
ler ID information. You are currently trying to secure the edge and provide=
 identity on top of this mess. <br>
</div></div>_____________<br>Roman Shpount<br>

--e89a8fb208dcc6ce7004bcc84696--

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2012/4/3 Roman Shpount <roman@telurix.com>:
> On Tue, Apr 3, 2012 at 9:05 AM, Olle E. Johansson <oej@edvina.net> wrote:
>>
>>
>> Now we will have to separate "PSTN-emulating" gateways that accept calls
>> to all phone numbers but play a prompt saying "You gotta be kidding me -
>> calling a phone number?" from REAL gateways that have a connection to th=
e
>> PSTN world.
>>
>> Will guys connecting with SS7 have a certificate signed by the ITU as a
>> "TRUE" PSTN provider and the voip guy in the basement next door just hav=
e a
>> "Best effort fourth-tier PSTN service" certificate?
>>
>> I think that any identity of any PSTN gateway just identifies the gatewa=
y
>> as a server. Not as a service.
>>
> I agree with you that you can only identify the gateway. Above this, I th=
ink
> the whole discussion is pointless since there are no security guarantees
> within PSTN. A million of people can be listening in. You can be connecte=
d
> to a completely different number then the one you've dialed due to LNP, c=
all
> routing rules, call forwarding, or anything else. If you are dialing
> internationally your traffic often goes over unsecured public internet. S=
o
> far, 99.999% of all phone calls were unsecured, tapped into, recorded and
> listen by anybody who possessed even the moderate desire to do so. If you
> start talking about calls coming from PSTN, you have even less guarantees
> about accuracy of the caller ID information. You are currently trying to
> secure the edge and provide identity on top of this mess.


I don't understand why we are trying to resolve eternal PSTN problems in rt=
cweb.



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

From roman@telurix.com  Tue Apr  3 08:58:16 2012
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From: Roman Shpount <roman@telurix.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Identity and PSTN gateways
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--047d7b33cb6cdaf23c04bcc85f60
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On Tue, Apr 3, 2012 at 11:50 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> I don't say that SRTP-DTLS is bad, not at all. I agree that it MUST be
> the preferred option for a WebRTC communication.
>

I actually would say that SRTP-DTLS is bad. It adds latency to call setup,
provides a bunch of features nobody needs, and does not map to SDES-SRTP
without re-encrypting the traffic. It has no deployed base and makes
interop a lot harder. Unfortunately this is the ony thing that provides
anything even remotely qualifying as secure communications right now. I
would prefer to see a new key exchange protocol designed instead of
DTLS-SRTP. I know that there is effort to patch DTLS-SRTP to fix call setup
delays, but all those patches will only make interop harder, since they
increase the number of necessary tests.
_____________
Roman Shpount

--047d7b33cb6cdaf23c04bcc85f60
Content-Type: text/html; charset=ISO-8859-1
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<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Apr 3, 2012 at 11:50 A=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</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">
I don&#39;t say that SRTP-DTLS is bad, not at all. I agree that it MUST be<=
br>
the preferred option for a WebRTC communication.<br></blockquote></div><br>=
I actually would say that SRTP-DTLS is bad. It adds latency to call setup, =
provides a bunch of features nobody needs, and does not map to SDES-SRTP wi=
thout re-encrypting the traffic. It has no deployed base and makes interop =
a lot harder. Unfortunately this is the ony thing that provides anything ev=
en remotely qualifying as secure communications right now. I would prefer t=
o see a new key exchange protocol designed instead of DTLS-SRTP. I know tha=
t there is effort to patch DTLS-SRTP to fix call setup delays, but all thos=
e patches will only make interop harder, since they increase the number of =
necessary tests. <br>
_____________<br>Roman Shpount<br>
<br>

--047d7b33cb6cdaf23c04bcc85f60--

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Subject: Re: [rtcweb] Identity and PSTN gateways
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On Tue, Apr 3, 2012 at 11:53 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2012/4/3 Roman Shpount <roman@telurix.com>:
> > I agree with you that you can only identify the gateway. Above this, I
> think
> > the whole discussion is pointless since there are no security guarantee=
s
> > within PSTN. A million of people can be listening in. You can be
> connected
> > to a completely different number then the one you've dialed due to LNP,
> call
> > routing rules, call forwarding, or anything else. If you are dialing
> > internationally your traffic often goes over unsecured public internet.
> So
> > far, 99.999% of all phone calls were unsecured, tapped into, recorded a=
nd
> > listen by anybody who possessed even the moderate desire to do so. If y=
ou
> > start talking about calls coming from PSTN, you have even less guarante=
es
> > about accuracy of the caller ID information. You are currently trying t=
o
> > secure the edge and provide identity on top of this mess.
>
>
> I don't understand why we are trying to resolve eternal PSTN problems in
> rtcweb.
>
> I do not, even though you repeatedly say that you want to fix the Interne=
t
via WebRTC. All I am saying that the PSTN gateway cannot provide identity
on behalf of the phone number. PSTN phone number is not an identity and we
should never display something that says "You have a secure communication
with +1800YOURBANK". You can only say "You have a secure communication with
ACME Telecom Phone Gateway"
_____________
Roman Shpount

--047d7b15ac43cfba8304bcc86b89
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<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Apr 3, 2012 at 11:53 A=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</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">
2012/4/3 Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telur=
ix.com</a>&gt;:<br>
<div><div></div><div class=3D"h5">&gt; I agree with you that you can only i=
dentify the gateway. Above this, I think<br>
&gt; the whole discussion is pointless since there are no security guarante=
es<br>
&gt; within PSTN. A million of people can be listening in. You can be conne=
cted<br>
&gt; to a completely different number then the one you&#39;ve dialed due to=
 LNP, call<br>
&gt; routing rules, call forwarding, or anything else. If you are dialing<b=
r>
&gt; internationally your traffic often goes over unsecured public internet=
. So<br>
&gt; far, 99.999% of all phone calls were unsecured, tapped into, recorded =
and<br>
&gt; listen by anybody who possessed even the moderate desire to do so. If =
you<br>
&gt; start talking about calls coming from PSTN, you have even less guarant=
ees<br>
&gt; about accuracy of the caller ID information. You are currently trying =
to<br>
&gt; secure the edge and provide identity on top of this mess.<br>
<br>
<br>
</div></div>I don&#39;t understand why we are trying to resolve eternal PST=
N problems in rtcweb.<br>
<div><div></div><div class=3D"h5"><br></div></div></blockquote><div>I do no=
t, even though you repeatedly say that you want to fix the Internet via Web=
RTC. All I am saying that the PSTN gateway cannot provide identity on behal=
f of the phone number. PSTN phone number is not an identity and we should n=
ever display something that says &quot;You have a secure communication with=
 +1800YOURBANK&quot;. You can only say &quot;You have a secure communicatio=
n with ACME Telecom Phone Gateway&quot; <br>
</div><div>_____________<br>Roman Shpount<br>=A0
<br></div></div>

--047d7b15ac43cfba8304bcc86b89--

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Subject: Re: [rtcweb] Identity and PSTN gateways
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3 apr 2012 kl. 18:01 skrev Roman Shpount:

>=20
> On Tue, Apr 3, 2012 at 11:53 AM, I=F1aki Baz Castillo <ibc@aliax.net> =
wrote:
> 2012/4/3 Roman Shpount <roman@telurix.com>:
> > I agree with you that you can only identify the gateway. Above this, =
I think
> > the whole discussion is pointless since there are no security =
guarantees
> > within PSTN. A million of people can be listening in. You can be =
connected
> > to a completely different number then the one you've dialed due to =
LNP, call
> > routing rules, call forwarding, or anything else. If you are dialing
> > internationally your traffic often goes over unsecured public =
internet. So
> > far, 99.999% of all phone calls were unsecured, tapped into, =
recorded and
> > listen by anybody who possessed even the moderate desire to do so. =
If you
> > start talking about calls coming from PSTN, you have even less =
guarantees
> > about accuracy of the caller ID information. You are currently =
trying to
> > secure the edge and provide identity on top of this mess.
>=20
>=20
> I don't understand why we are trying to resolve eternal PSTN problems =
in rtcweb.
>=20
> I do not, even though you repeatedly say that you want to fix the =
Internet via WebRTC. All I am saying that the PSTN gateway cannot =
provide identity on behalf of the phone number. PSTN phone number is not =
an identity and we should never display something that says "You have a =
secure communication with +1800YOURBANK". You can only say "You have a =
secure communication with ACME Telecom Phone Gateway"=20

Exactly. Thanks.

/O


From ibc@aliax.net  Tue Apr  3 10:00:11 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 3 Apr 2012 18:59:49 +0200
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Identity and PSTN gateways
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2012/4/3 Roman Shpount <roman@telurix.com>:
> On Tue, Apr 3, 2012 at 11:53 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:

>> I don't understand why we are trying to resolve eternal PSTN problems in
>> rtcweb.
>>
> I do not, even though you repeatedly say that you want to fix the Interne=
t
> via WebRTC.

Disallowing plain and hyper-unsecure RTP is not fixing Internet, IMHO ;)




> All I am saying that the PSTN gateway cannot provide identity on
> behalf of the phone number. PSTN phone number is not an identity and we
> should never display something that says "You have a secure communication
> with +1800YOURBANK". You can only say "You have a secure communication wi=
th
> ACME Telecom Phone Gateway"

And for sure you are 100% right. PSTN is insecure by nature (signed: a
telco worker).



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

From igor.faynberg@alcatel-lucent.com  Tue Apr  3 10:12:09 2012
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Subject: Re: [rtcweb] Identity and PSTN gateways
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I cannot quite agree with IÃ±aki, although I do need to qualify this. 
(Maybe we don't disagree after all.)

POTS has been as secure as anything could be (short of attacks involving 
splicing the twisted pair or cable), but all kinds of games played with 
gateways have made it vulnerable as far as caller identification is 
concerned.  But here--at least in the US--there are strong regulations 
that protect the POTS end-points (i.e., consumers).

The major point though is that wireless networking has been sufficiently 
secure in 3G and 4G. To this end, a mobile phone with the SIM 
application is a strong authentication device, and there are many 
interesting way to bootstrap this into IdP services.  We had a very 
interesting discussion during the BrowserID session, which came I think 
very closely to this point.

Igor

On 4/3/2012 12:59 PM, IÃ±aki Baz Castillo wrote:
> ...
> And for sure you are 100% right. PSTN is insecure by nature (signed: a
> telco worker).
>
>
>

From sergel@google.com  Tue Apr  3 10:55:28 2012
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From: Serge Lachapelle <sergel@google.com>
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Subject: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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--14dae9c09bc201ddde04bcca0336
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*[Forking the thread]*
*
*
*Hello folks,*

Google confirms that the VP8 patent grant applies to both third-party
hardware and software implementations of VP8.
*
*
*Google encourages the community to create hardware implementations of VP8,
and has recently blogged about a number of new hardware implementations on
the WebM blog (
http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware.html).
*
*
*
*Google is quite proud of what the community has done with VP8 and looks
forward to seeing more implementations of VP8 in both hardware and
software. *
*
*
*Regards,*
*
*
*/Serge Lachapelle, Google, Stockholm*

On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org> wrote:

>
>
> On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org>
> wrote:
>
> >On 03/29/2012 10:20 AM, Stephan Wenger wrote:
> >> The second part of your sentence may or may not be true, depending on
> >>your
> >> relationship with google, your willingness to use the webm
> >>implementation
> >> in unchanged form, and other factors.  Please see the webm license
> >> conditions, which AFAIK can be found here:
> >> http://www.webmproject.org/license/additional/
> >Correct.  I think you are referring to this part, explicitly:
> >> If you or your agent or exclusive licensee institute or order or agree
> >> to the institution of patent litigation against any entity (including
> >> a cross-claim or counterclaim in a lawsuit) alleging that this
> >> implementation of VP8 or any code incorporated within this
> >> implementation of VP8 constitutes direct or contributory patent
> >> infringement, or inducement of patent infringement, then any patent
> >> rights granted to you under this License for this implementation of
> >> VP8 shall terminate as of the date such litigation is filed.
> >Perhaps I assumed that that is a very reasonable part of the license.
> >That is, if you are suing someone alleging a patent infringement within
> >VP8, you are no longer granted the license to use VP8's patented
> >technologies that Google owns.
>
> Yes, that's one issue.  Call it personal preference for different type of
> reciprocity conditions :-)  (I could rant about it for hours, but let's
> continue to pretend that this is mostly a technical mailing list)
>
> The other issue, though (the fact that the license grant extends only to
> the VP8 implementation as provided by google, and does not extent to
> derivative works such as hardware implementations) should be moderately
> alarming even for an open source person.  With respect to this clause, I
> will note that I criticized the licensing conditions in private and in
> public (IETF mike) several times, months ago, and nothing happened.
> Suggests to me one of three things: (1) google is a large company and
> decisions take time, or (2) google's legal is currently occupied with
> other stuff, or (3) that the choice of language is intentional, and
> intended to prevent forks.  Take your pick.
> Stephan
>
> >_______________________________________________
> >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
> Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880
>
> Apparently, this footer is required in Europe. Apologies. This email may
> be confidential or privileged.  If you received this communication by
> mistake, please don't forward it to anyone else,please erase all copies and
> attachments, and please let me know that it went to the wrong person.
>  Thanks.
>  <https://www.ietf.org/mailman/listinfo/rtcweb>



-- 
Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32


-- 
Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32
Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880

Apparently, this footer is required in Europe. Apologies. This email may be
confidential or privileged.  If you received this communication by mistake,
please don't forward it to anyone else,please erase all copies and
attachments, and please let me know that it went to the wrong person.
 Thanks.

--14dae9c09bc201ddde04bcca0336
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<div><b style=3D"font-family:Times;font-size:medium"><span style=3D"font-si=
ze:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-=
space:pre-wrap">[Forking the thread]</span></b></div><div><b style=3D"font-=
family:Times;font-size:medium"><span style=3D"font-size:15px;font-family:Ar=
ial;font-weight:normal;vertical-align:baseline;white-space:pre-wrap"><br>

</span></b></div><div><b id=3D"internal-source-marker_0.23840911174193025" =
style=3D"font-family:Times;font-size:medium"><span style=3D"font-size:15px;=
font-family:Arial;font-weight:normal;vertical-align:baseline;white-space:pr=
e-wrap">Hello folks,</span></b></div>

<div><span id=3D"internal-source-marker_0.23840911174193025" style=3D"font-=
family:Times;font-size:medium"><span style=3D"font-size:15px;font-family:Ar=
ial;font-weight:normal;vertical-align:baseline;white-space:pre-wrap"></span=
><br>

<span style=3D"font-size:15px;font-family:Arial;font-weight:normal;vertical=
-align:baseline;white-space:pre-wrap">Google confirms that the VP8 patent g=
rant applies to both third-party hardware and software implementations of V=
P8.=A0</span></span></div>

<div><b style=3D"font-family:Times;font-size:medium"><span style=3D"font-si=
ze:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-=
space:pre-wrap"><br></span></b></div><div><b style=3D"font-family:Times;fon=
t-size:medium"><span style=3D"font-size:15px;font-family:Arial;font-weight:=
normal;vertical-align:baseline;white-space:pre-wrap">Google encourages the =
community to create hardware implementations of VP8, and has recently blogg=
ed about a number of new hardware implementations on the WebM blog ( <a hre=
f=3D"http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware.=
html">http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware=
.html</a> ).=A0</span></b></div>

<div><b style=3D"font-family:Times;font-size:medium"><span style=3D"font-si=
ze:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-=
space:pre-wrap"><br></span></b></div><div><b style=3D"font-family:Times;fon=
t-size:medium"><span style=3D"font-size:15px;font-family:Arial;font-weight:=
normal;vertical-align:baseline;white-space:pre-wrap">Google is quite proud =
of what the community has done with VP8 and looks forward to seeing more im=
plementations of VP8 in both hardware and software.=A0</span></b></div>

<div><b style=3D"font-family:Times;font-size:medium"><span style=3D"font-si=
ze:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-=
space:pre-wrap"><br></span></b></div><div><b style=3D"font-family:Times;fon=
t-size:medium"><span style=3D"font-size:15px;font-family:Arial;font-weight:=
normal;vertical-align:baseline;white-space:pre-wrap">Regards,</span></b></d=
iv>

<div><b style=3D"font-family:Times;font-size:medium"><span style=3D"font-si=
ze:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-=
space:pre-wrap"><br></span></b></div><div><b style=3D"font-family:Times;fon=
t-size:medium"><span style=3D"font-size:15px;font-family:Arial;font-weight:=
normal;vertical-align:baseline;white-space:pre-wrap">/Serge Lachapelle,
Google, Stockholm</span></b></div><div><br></div><div><div class=3D"gmail_q=
uote">On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <span dir=3D"ltr">&lt;<=
a href=3D"mailto:stewe@stewe.org" target=3D"_blank">stewe@stewe.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">
<br>
<br>
On 3.29.2012 16:24 , &quot;Basil Mohamed Gohar&quot; &lt;<a href=3D"mailto:=
basilgohar@librevideo.org" target=3D"_blank">basilgohar@librevideo.org</a>&=
gt;<br>
wrote:<br>
<div><div><br>
&gt;On 03/29/2012 10:20 AM, Stephan Wenger wrote:<br>
&gt;&gt; The second part of your sentence may or may not be true, depending=
 on<br>
&gt;&gt;your<br>
&gt;&gt; relationship with google, your willingness to use the webm<br>
&gt;&gt;implementation<br>
&gt;&gt; in unchanged form, and other factors. =A0Please see the webm licen=
se<br>
&gt;&gt; conditions, which AFAIK can be found here:<br>
&gt;&gt; <a href=3D"http://www.webmproject.org/license/additional/" target=
=3D"_blank">http://www.webmproject.org/license/additional/</a><br>
&gt;Correct. =A0I think you are referring to this part, explicitly:<br>
&gt;&gt; If you or your agent or exclusive licensee institute or order or a=
gree<br>
&gt;&gt; to the institution of patent litigation against any entity (includ=
ing<br>
&gt;&gt; a cross-claim or counterclaim in a lawsuit) alleging that this<br>
&gt;&gt; implementation of VP8 or any code incorporated within this<br>
&gt;&gt; implementation of VP8 constitutes direct or contributory patent<br=
>
&gt;&gt; infringement, or inducement of patent infringement, then any paten=
t<br>
&gt;&gt; rights granted to you under this License for this implementation o=
f<br>
&gt;&gt; VP8 shall terminate as of the date such litigation is filed.<br>
&gt;Perhaps I assumed that that is a very reasonable part of the license.<b=
r>
&gt;That is, if you are suing someone alleging a patent infringement within=
<br>
&gt;VP8, you are no longer granted the license to use VP8&#39;s patented<br=
>
&gt;technologies that Google owns.<br>
<br>
</div></div>Yes, that&#39;s one issue. =A0Call it personal preference for d=
ifferent type of<br>
reciprocity conditions :-) =A0(I could rant about it for hours, but let&#39=
;s<br>
continue to pretend that this is mostly a technical mailing list)<br>
<br>
The other issue, though (the fact that the license grant extends only to<br=
>
the VP8 implementation as provided by google, and does not extent to<br>
derivative works such as hardware implementations) should be moderately<br>
alarming even for an open source person. =A0With respect to this clause, I<=
br>
will note that I criticized the licensing conditions in private and in<br>
public (IETF mike) several times, months ago, and nothing happened.<br>
Suggests to me one of three things: (1) google is a large company and<br>
decisions take time, or (2) google&#39;s legal is currently occupied with<b=
r>
other stuff, or (3) that the choice of language is intentional, and<br>
intended to prevent forks. =A0Take your pick.<br>
<span><font color=3D"#888888">Stephan<br>
</font></span><div><div><br>
&gt;_______________________________________________<br>
&gt;rtcweb mailing list<br>
&gt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bla=
nk">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<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></div></div><a href=3D"http=
s://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank"><font color=3D"=
#999999" size=3D"1">Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | O=
rg. nr. 556656-6880=A0</font><br>


<br><font color=3D"#999999" size=3D"1">Apparently, this footer is required =
in Europe. Apologies. This email may be confidential or privileged. =A0If y=
ou received this communication by mistake, please don&#39;t forward it to a=
nyone else,please erase all copies and attachments, and please let me know =
that it went to the wrong person. =A0Thanks.</font><br>



</a></blockquote></div></div>
<br clear=3D"all"><div><br></div>-- <br><span style=3D"color:rgb(85,85,85);=
font-family:sans-serif;line-height:20px"><span style=3D"border-top-width:2p=
x;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;bord=
er-top-style:solid;border-right-style:solid;border-bottom-style:solid;borde=
r-left-style:solid;border-top-color:rgb(213,15,37);border-right-color:rgb(2=
13,15,37);border-bottom-color:rgb(213,15,37);border-left-color:rgb(213,15,3=
7);padding-top:2px;margin-top:2px">Serge Lachapelle=A0|</span><span style=
=3D"border-top-width:2px;border-right-width:0px;border-bottom-width:0px;bor=
der-left-width:0px;border-top-style:solid;border-right-style:solid;border-b=
ottom-style:solid;border-left-style:solid;border-top-color:rgb(51,105,232);=
border-right-color:rgb(51,105,232);border-bottom-color:rgb(51,105,232);bord=
er-left-color:rgb(51,105,232);padding-top:2px;margin-top:2px">=A0Product Ma=
nager=A0|</span><span style=3D"border-top-width:2px;border-right-width:0px;=
border-bottom-width:0px;border-left-width:0px;border-top-style:solid;border=
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-top-color:rgb(0,153,57);border-right-color:rgb(0,153,57);border-bottom-col=
or:rgb(0,153,57);border-left-color:rgb(0,153,57);padding-top:2px;margin-top=
:2px">=A0<a href=3D"mailto:sergel@google.com" target=3D"_blank">sergel@goog=
le.com</a>=A0|</span><span style=3D"border-top-width:2px;border-right-width=
:0px;border-bottom-width:0px;border-left-width:0px;border-top-style:solid;b=
order-right-style:solid;border-bottom-style:solid;border-left-style:solid;b=
order-top-color:rgb(238,178,17);border-right-color:rgb(238,178,17);border-b=
ottom-color:rgb(238,178,17);border-left-color:rgb(238,178,17);padding-top:2=
px;margin-top:2px">=A0+46 732 01 22 32</span></span><br>


<br clear=3D"all"><div><br></div>-- <br><meta charset=3D"utf-8"><div><meta =
charset=3D"utf-8"><div style=3D"line-height:1.5em;padding-top:10px;margin-t=
op:10px;color:rgb(85,85,85);font-family:sans-serif;font-size:small"><span s=
tyle=3D"border-top-width:2px;border-right-width:0px;border-bottom-width:0px=
;border-left-width:0px;border-top-style:solid;border-right-style:solid;bord=
er-bottom-style:solid;border-left-style:solid;border-top-color:rgb(213,15,3=
7);border-right-color:rgb(213,15,37);border-bottom-color:rgb(213,15,37);bor=
der-left-color:rgb(213,15,37);padding-top:2px;margin-top:2px">Serge Lachape=
lle=A0|</span><span style=3D"border-top-width:2px;border-right-width:0px;bo=
rder-bottom-width:0px;border-left-width:0px;border-top-style:solid;border-r=
ight-style:solid;border-bottom-style:solid;border-left-style:solid;border-t=
op-color:rgb(51,105,232);border-right-color:rgb(51,105,232);border-bottom-c=
olor:rgb(51,105,232);border-left-color:rgb(51,105,232);padding-top:2px;marg=
in-top:2px">=A0Product Manager=A0|</span><span style=3D"border-top-width:2p=
x;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;bord=
er-top-style:solid;border-right-style:solid;border-bottom-style:solid;borde=
r-left-style:solid;border-top-color:rgb(0,153,57);border-right-color:rgb(0,=
153,57);border-bottom-color:rgb(0,153,57);border-left-color:rgb(0,153,57);p=
adding-top:2px;margin-top:2px">=A0<a href=3D"mailto:sergel@google.com" targ=
et=3D"_blank">sergel@google.com</a>=A0|</span><span style=3D"border-top-wid=
th:2px;border-right-width:0px;border-bottom-width:0px;border-left-width:0px=
;border-top-style:solid;border-right-style:solid;border-bottom-style:solid;=
border-left-style:solid;border-top-color:rgb(238,178,17);border-right-color=
:rgb(238,178,17);border-bottom-color:rgb(238,178,17);border-left-color:rgb(=
238,178,17);padding-top:2px;margin-top:2px">=A0+46 732 01 22 32</span></div=
>

</div><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">Google =
Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880=A0</fon=
t><br><br><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">App=
arently, this footer is required in Europe. Apologies. This email may be co=
nfidential or privileged. =A0If you received this communication by mistake,=
 please don&#39;t forward it to anyone else,please erase all copies and att=
achments, and please let me know that it went to the wrong person. =A0Thank=
s.</font><br>



--14dae9c09bc201ddde04bcca0336--

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Date: Tue, 03 Apr 2012 20:00:33 +0200
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Subject: [rtcweb] On DTLS-SRTP trust model (and consideration for SDES-SRTP)
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>From trust model SDES-SRTP it's very similar with DTLS-SRTP with the
regards to the need to trust third party with no real end-to-end
authentication/encryption support.

Let me explain this assumption:

- DTLS-SRTP key exchange without fingerprint/identity verification mean
"false sense of security" = no security

- DTLS-SRTP to verify fingerprint require identity services
- Identity services rely on trusted third parties
- Trusted third party identity services rely on TLS (https)

So basically in any case the 'root of trust' is based on TLS.

>From http://www.ietf.org/id/draft-ietf-rtcweb-security-arch-01.txt :

4.1.  Initial Signaling
[...]

   Based purely on the interface provided
   by the signaling server, they know that the signaling server claims
   that the call is from Alice to Bob.

4.3.  DTLS Handshake
[...]

   If Alice and Bob authenticated via their IdPs,
   then they also know that the signaling service is not attacking them.
   Even if they do not use an IdP, as long as they have minimal trust in
   the signaling service not to perform a man-in-the-middle attack, they
   know that their communications are secure against the signaling
   service as well.

5.1.  Origin and Web Security Issues

   The basic unit of permissions for RTCWEB is the origin [RFC6454].
   Because the security of the origin depends on being able to
   authenticate content from that origin, the origin can only be
   securely established if data is transferred over HTTPS [RFC2818].


So, while i totally agree to get DTLS-SRTP into the standard but please
don't consider it the "panacea" as it's trust model, for how it's proposed:
- DTLS-SRTP work securely when relying on trusted third party for idP
- DTLS-SRTP work securely when you trust the signaling services
- DTLS-SRTP work securely only when you trust third party that are trued
via other TLS/CA based trust chain:
  - signaling service (web server)
  - identity provider

So to have a "secure DTLS-SRTP" infrastructure that guarantee end-to-end
encryption the user is not independent (like for ZRTP SAS that the
authors of rtcweb-security-arch doesn't consider a valuable point) but
must rely on at least 2 separate third party services that rely on TLS
trust model.

With SDES-SRTP you get "less security" than what DTLS-SRTP can provide
but with an "order of magnitude" of simplicity because:
- SDES-SRTP work securely when relying on a single party for TLS

So again summarizing:

- DTLS-SRTP "can" provide end-to-end crypto...
...ONLY IF the user trust 2 services that rely on TLS
 (the user trust at least 2 different services, signaling + idP)

- SDES-SRTP "can" provide end-to-site crypto...
..ONLY IF the user trust 1 single service with TLS

That's my point in arguing that SDES-SRTP MUST be implemented for it's
strongest simplicity in implementation and in providing end-to-site
encryption without all the high complexity DTLS-SRTP.

But always consider that DTLS-SRTP, in order to be secure, still require
the user to trust some third party and it's not able to independently
verify the trust-condition of the other user.

It's a matter of considering which trust scenario you want to protect
about.

DTLS-SRTP represent the "elephant in a room" concept, where a *huge
level of complexity* is under the hood to be able to provide even a
basic security level (even end-to-site encryption), and the end-user
doesn't have a real way to understand "which security level" he is.

At least with SDES-SRTP the user know that there is less, but simpler to
be guaranteed, specific end-to-site security model (i connect to gmail,
i trust gmail snooping my email).

Please consider that as the stratification of complexity for DTLS-SRTP
will probably result in practical less-reduced-security.

IMHO any technology must provide protection against a specific risk
model, here in this WG we are saying that DTLS-SRTP is the panacea for
all the risk-model.

That's unfortunately not true.

There is "the right technology" to face "the right risk context" and
when you try to make the use of a "specific technology" to satisfy all
the risk context, the results are not always very good, due to the
complexity of different factors to be considered.

If we think that just providing something that can deliver "unsecure
end-to-end encryption" by providing a "false sense of security" is a
good things, let's goes on.

But imho every technology must be specifically adapted to a specific
risk scenario.

If we want to really consider DTLS-SRTP's end-to-end encryption, the
fingerprint checking must be completely in the hand of the user, with
ZRTP's like SAS checking and not rely on third party services for trust.

Fabio

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Subject: Re: [rtcweb] Support of SDES in WebRTC (Javascript Crypto Extensions)
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On 3/29/12 11:59 PM, Gregory Maxwell wrote:
> Oscar Ohlsson [oscar.ohlsson@ericsson.com] wrote:
>> That's why I wrote "the entire webapp" below. If it was not clear I meant that the
>> - main HTML page
>> - all external CSS files, JavaScript files, images, etc
>> - all XmlHttpRequests
>> - all WebSocket connections
>> are protected with TLS. This is obviously verifiable and it's a feature supported by all modern browsers (no mixed content).
> 
> Even this is insufficient.
> 
> First, even if it is in principle possible to provide adequate
> cryptographic security inside the JS applications a great many of them
> won't (for many reasons). 

That's a on-going debate that for many years kept researcher from
working on JS-crypto, but nowdays the value of JS-crypto it's well
recognized with valuable projects working on it.

I think that in 2012 the old
http://www.matasano.com/articles/javascript-cryptography/ on Javascript
crypto can be considered superseded by
http://hellais.wordpress.com/2011/12/27/how-to-improve-javascript-cryptography/
and by the discussion that arised from the OpenPGPJS mailing lists
https://github.com/tanx/SafeWith.me/wiki/FAQ .

The overall result is that it's possible to do securely enough crypto in
javascript and that, as everything, it depends on the way you use it.

So this is a typical out-of-date javascript crypto security
consideration that is practically discussed in all the mailing lists i'm
subscribed about javascript crypto.

But from a pragmatic point of view there are very valuable way to
implement secure Javascript Crypto and browser vendors are supporting
and driving the path of JS-crypto securization.

The cryptographic negotiation can be, and imho should be, pluggable, to
be able to work at application level.
Thunderbird doesn't provide OpenPGP encryption, but Enigmail (as a
plugin) does.

So imho we should leave to the application the ability to improve and
adapt the security mechanisms, given all the context and use-cases that
this WG cannot even forecast.

Do we want rtcweb security working as "a blackbox" or do we want to make
it working also like an "open platform" where it may be possible.

This is not the strongest argument to have also support for SDES-SRTP,
but it's for sure an argument, where the two strongest factor of SDES are:
- simplicity
- interoperability
- opennes (so ability to hook custom crypto key exchange methods, due to
it's simplicity in handling keys relying on TLS transport)

> It's also inadequate on purely technical grounds: Javascript provides
> no mechanism for working with mlocked memory,  no mechanism to ensure
> that garbage collected data gets zeroized.  Your crypto app in JS could
> easily have its long term keying material pulled out of free ram by
> malware long after it runs, or pulled off the disk from swap.
> The breakneck pace of fancy JIT systems makes it seem unlikely to me
> that javascript will provide for that any time soon.

That's a not real issue that cannot be fixed in any case, also in case
you are using a bullet-proof native-code VoIP client.
If you local machine compromised there's no mlock/garbage collection
that can protect against memory dumping and memory injection.

Javascript encryption is secure, for most context, as native encryption
and many attacks considered to javascript crypto are attacks present in
native implementation of crypto.

Fabio

From basilgohar@librevideo.org  Tue Apr  3 11:37:20 2012
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Serge,

Thank you very much for this.  To go a step forward, I would encourage
that this be made more legally obvious, because a reference to a single
e-mail on an "obscure" mailing list (relative to what most people are
familiar with) will not provide the deepest levels of assurance to those
not actively involved with the WebRTC project.

I would humbly request an unambiguous statement somewhere on the WebM
project website that makes this assertion clear, because what is
publicly available is, at the very least, ambiguous, and at the worst,
"evidence of Google's nefarious scheme to pounce on hardware
implementers after wide adoption".

On 04/03/2012 01:54 PM, Serge Lachapelle wrote:
> *[Forking the thread]*
> *
>
>
> *
> *Hello folks,*
>
> Google confirms that the VP8 patent grant applies to both third-party
> hardware and software implementations of VP8. 
> *
> *
> *Google encourages the community to create hardware implementations of
> VP8, and has recently blogged about a number of new hardware
> implementations on the WebM blog (
> http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware.html
> ). *
> *
> *
> *Google is quite proud of what the community has done with VP8 and
> looks forward to seeing more implementations of VP8 in both hardware
> and software. *
> *
> *
> *Regards,*
> *
> *
> */Serge Lachapelle,
> Google, Stockholm*
>
> On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org
> <mailto:stewe@stewe.org>> wrote:
>
>
>
>     On 3.29.2012 16:24 , "Basil Mohamed Gohar"
>     <basilgohar@librevideo.org <mailto:basilgohar@librevideo.org>>
>     wrote:
>
>     >On 03/29/2012 10:20 AM, Stephan Wenger wrote:
>     >> The second part of your sentence may or may not be true,
>     depending on
>     >>your
>     >> relationship with google, your willingness to use the webm
>     >>implementation
>     >> in unchanged form, and other factors.  Please see the webm license
>     >> conditions, which AFAIK can be found here:
>     >> http://www.webmproject.org/license/additional/
>     >Correct.  I think you are referring to this part, explicitly:
>     >> If you or your agent or exclusive licensee institute or order
>     or agree
>     >> to the institution of patent litigation against any entity
>     (including
>     >> a cross-claim or counterclaim in a lawsuit) alleging that this
>     >> implementation of VP8 or any code incorporated within this
>     >> implementation of VP8 constitutes direct or contributory patent
>     >> infringement, or inducement of patent infringement, then any patent
>     >> rights granted to you under this License for this implementation of
>     >> VP8 shall terminate as of the date such litigation is filed.
>     >Perhaps I assumed that that is a very reasonable part of the license.
>     >That is, if you are suing someone alleging a patent infringement
>     within
>     >VP8, you are no longer granted the license to use VP8's patented
>     >technologies that Google owns.
>
>     Yes, that's one issue.  Call it personal preference for different
>     type of
>     reciprocity conditions :-)  (I could rant about it for hours, but
>     let's
>     continue to pretend that this is mostly a technical mailing list)
>
>     The other issue, though (the fact that the license grant extends
>     only to
>     the VP8 implementation as provided by google, and does not extent to
>     derivative works such as hardware implementations) should be
>     moderately
>     alarming even for an open source person.  With respect to this
>     clause, I
>     will note that I criticized the licensing conditions in private and in
>     public (IETF mike) several times, months ago, and nothing happened.
>     Suggests to me one of three things: (1) google is a large company and
>     decisions take time, or (2) google's legal is currently occupied with
>     other stuff, or (3) that the choice of language is intentional, and
>     intended to prevent forks.  Take your pick.
>     Stephan
>
>     >_______________________________________________
>     >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
>     Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr.
>     556656-6880 
>
>     Apparently, this footer is required in Europe. Apologies. This
>     email may be confidential or privileged.  If you received this
>     communication by mistake, please don't forward it to anyone
>     else,please erase all copies and attachments, and please let me
>     know that it went to the wrong person.  Thanks.
>     <https://www.ietf.org/mailman/listinfo/rtcweb>
>
>
>
> -- 
> Serge Lachapelle | Product Manager | sergel@google.com
> <mailto:sergel@google.com> | +46 732 01 22 32
>
>
> -- 
> Serge Lachapelle | Product Manager | sergel@google.com
> <mailto:sergel@google.com> | +46 732 01 22 32
> Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr.
> 556656-6880 
>
> Apparently, this footer is required in Europe. Apologies. This email
> may be confidential or privileged.  If you received this communication
> by mistake, please don't forward it to anyone else,please erase all
> copies and attachments, and please let me know that it went to the
> wrong person.  Thanks.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Libre Video
http://librevideo.org


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Subject: Re: [rtcweb] Identity and PSTN gateways
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On 4/3/2012 10:11 AM, Lu, Hui-Lan (Huilan) wrote:
> It seems the same problem exists when a TURN relay is involved in an inter-browser call. The endpoints cannot verify each other's identity directly. They need to trust the relay to interconnect them and not to do anything evil, such as snooping.

That shouldn't be the case if you're using DTLS-SRTP (i.e. TURN server 
doesn't decrypt/re-encrypt; if it does then it's a MITM).  If it's 
SDES-SRTP, then the TURN server could tap off the encrypted traffic for 
later analysis by whomever has the key, or be supplied with the key to 
do realtime decode-and-forward, analysis or traffic modification.

-- 
Randell Jesup
randell-ietf@jesup.org


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2012/4/3 Randell Jesup <randell-ietf@jesup.org>:
> On 4/3/2012 10:11 AM, Lu, Hui-Lan (Huilan) wrote:
>>
>> It seems the same problem exists when a TURN relay is involved in an
>> inter-browser call. The endpoints cannot verify each other's identity
>> directly. They need to trust the relay to interconnect them and not to d=
o
>> anything evil, such as snooping.
>
>
> That shouldn't be the case if you're using DTLS-SRTP (i.e. TURN server
> doesn't decrypt/re-encrypt; if it does then it's a MITM).

> If it's
> SDES-SRTP, then the TURN server could tap off the encrypted traffic for
> later analysis by whomever has the key,

And who has the key? The trusted HTTPS server.


> or be supplied with the key to do
> realtime decode-and-forward, analysis or traffic modification.

And who has the key? The trusted HTTPS server.


So what is the real problem? the trusted HTTPS server when attacked.
The *same* is true in case of DTLS-SRTP when the HTTPS server or the
identity provider (via TLS again) is attacked.


Please read the mail "On DTLS-SRTP trust model (and consideration for
SDES-SRTP)" from Fabio. DTLS is not the panacea.

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

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From: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Identity and PSTN gateways
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On 4/3/2012 4:47 PM, IÃ±aki Baz Castillo wrote:
> 2012/4/3 Randell Jesup<randell-ietf@jesup.org>:

>> If it's
>> SDES-SRTP, then the TURN server could tap off the encrypted traffic for
>> later analysis by whomever has the key,
>
> And who has the key? The trusted HTTPS server.
>
>> or be supplied with the key to do
>> realtime decode-and-forward, analysis or traffic modification.
>
> And who has the key? The trusted HTTPS server.
>
> So what is the real problem? the trusted HTTPS server when attacked.
> The *same* is true in case of DTLS-SRTP when the HTTPS server or the
> identity provider (via TLS again) is attacked.

Ah, but Identity Provider != Service Provider.  (It may be the same, but 
for smaller entities it wouldn't be - typically the Idp's will be "big 
sites" that act as primary email hosts, like Google.)  Yes, it's 
well-known in the discussion that if the service provider == Idp then 
the Idp can MITM you (or at least make things more complex; it may be 
possible to use key-chaining to reduce risk some).

> Please read the mail "On DTLS-SRTP trust model (and consideration for
> SDES-SRTP)" from Fabio. DTLS is not the panacea.

Nothing is.  :-)


-- 
Randell Jesup
randell-ietf@jesup.org

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Serge Lachapelle'" <sergel@google.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Will Google indemnify all implementers against any IPR claims?  Do that, and
I think the debate would be over.

 

Paul

 

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Serge Lachapelle
Sent: Tuesday, April 03, 2012 1:55 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal
for H.263 baseline codec]

 

[Forking the thread]

 

Hello folks,


Google confirms that the VP8 patent grant applies to both third-party
hardware and software implementations of VP8. 

 

Google encourages the community to create hardware implementations of VP8,
and has recently blogged about a number of new hardware implementations on
the WebM blog (
http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware.html
). 

 

Google is quite proud of what the community has done with VP8 and looks
forward to seeing more implementations of VP8 in both hardware and software.


 

Regards,

 

/Serge Lachapelle, Google, Stockholm

 

On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org> wrote:



On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org>
wrote:


>On 03/29/2012 10:20 AM, Stephan Wenger wrote:
>> The second part of your sentence may or may not be true, depending on
>>your
>> relationship with google, your willingness to use the webm
>>implementation
>> in unchanged form, and other factors.  Please see the webm license
>> conditions, which AFAIK can be found here:
>> http://www.webmproject.org/license/additional/
>Correct.  I think you are referring to this part, explicitly:
>> If you or your agent or exclusive licensee institute or order or agree
>> to the institution of patent litigation against any entity (including
>> a cross-claim or counterclaim in a lawsuit) alleging that this
>> implementation of VP8 or any code incorporated within this
>> implementation of VP8 constitutes direct or contributory patent
>> infringement, or inducement of patent infringement, then any patent
>> rights granted to you under this License for this implementation of
>> VP8 shall terminate as of the date such litigation is filed.
>Perhaps I assumed that that is a very reasonable part of the license.
>That is, if you are suing someone alleging a patent infringement within
>VP8, you are no longer granted the license to use VP8's patented
>technologies that Google owns.

Yes, that's one issue.  Call it personal preference for different type of
reciprocity conditions :-)  (I could rant about it for hours, but let's
continue to pretend that this is mostly a technical mailing list)

The other issue, though (the fact that the license grant extends only to
the VP8 implementation as provided by google, and does not extent to
derivative works such as hardware implementations) should be moderately
alarming even for an open source person.  With respect to this clause, I
will note that I criticized the licensing conditions in private and in
public (IETF mike) several times, months ago, and nothing happened.
Suggests to me one of three things: (1) google is a large company and
decisions take time, or (2) google's legal is currently occupied with
other stuff, or (3) that the choice of language is intentional, and
intended to prevent forks.  Take your pick.
Stephan


>_______________________________________________
>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

 <https://www.ietf.org/mailman/listinfo/rtcweb> Google Sweden AB | Kungsbron
2, SE-111 22 Stockholm | Org. nr. 556656-6880 

Apparently, this footer is required in Europe. Apologies. This email may be
confidential or privileged.  If you received this communication by mistake,
please don't forward it to anyone else,please erase all copies and
attachments, and please let me know that it went to the wrong person.
Thanks.





 

-- 
Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32



 

-- 

Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32

Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880 

Apparently, this footer is required in Europe. Apologies. This email may be
confidential or privileged.  If you received this communication by mistake,
please don't forward it to anyone else,please erase all copies and
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Will Google indemnify all implementers against any IPR claims?&nbsp; =
Do that, and I think the debate would be over.<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><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Paul<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-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Serge Lachapelle<br><b>Sent:</b> Tuesday, April 03, 2012 1:55 =
PM<br><b>To:</b> rtcweb@ietf.org<br><b>Subject:</b> [rtcweb] Google VP8 =
Patent Grant for third parties [Was Re:Proposal for H.263 baseline =
codec]<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal><span =
style=3D'font-size:11.5pt;font-family:"Arial","sans-serif"'>[Forking the =
thread]</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:11.5pt;font-family:"Arial","sans-serif"'>Hello =
folks,</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:13.5pt;font-family:"Times","serif"'><br></span><span =
style=3D'font-size:11.5pt;font-family:"Arial","sans-serif"'>Google =
confirms that the VP8 patent grant applies to both third-party hardware =
and software implementations of =
VP8.&nbsp;</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:11.5pt;font-family:"Arial","sans-serif"'>Google =
encourages the community to create hardware implementations of VP8, and =
has recently blogged about a number of new hardware implementations on =
the WebM blog ( <a =
href=3D"http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hard=
ware.html">http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-h=
ardware.html</a> ).&nbsp;</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:11.5pt;font-family:"Arial","sans-serif"'>Google is =
quite proud of what the community has done with VP8 and looks forward to =
seeing more implementations of VP8 in both hardware and =
software.&nbsp;</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:11.5pt;font-family:"Arial","sans-serif"'>Regards,</spa=
n><o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:11.5pt;font-family:"Arial","sans-serif"'>/Serge =
Lachapelle, Google, Stockholm</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><div><p =
class=3DMsoNormal>On Thu, Mar 29, 2012 at 16:53, Stephan Wenger &lt;<a =
href=3D"mailto:stewe@stewe.org" =
target=3D"_blank">stewe@stewe.org</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal><br><br>On 3.29.2012 16:24 , &quot;Basil Mohamed =
Gohar&quot; &lt;<a href=3D"mailto:basilgohar@librevideo.org" =
target=3D"_blank">basilgohar@librevideo.org</a>&gt;<br>wrote:<o:p></o:p><=
/p><div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><br>&gt;On 03/29/2012 10:20 AM, Stephan =
Wenger wrote:<br>&gt;&gt; The second part of your sentence may or may =
not be true, depending on<br>&gt;&gt;your<br>&gt;&gt; relationship with =
google, your willingness to use the =
webm<br>&gt;&gt;implementation<br>&gt;&gt; in unchanged form, and other =
factors. &nbsp;Please see the webm license<br>&gt;&gt; conditions, which =
AFAIK can be found here:<br>&gt;&gt; <a =
href=3D"http://www.webmproject.org/license/additional/" =
target=3D"_blank">http://www.webmproject.org/license/additional/</a><br>&=
gt;Correct. &nbsp;I think you are referring to this part, =
explicitly:<br>&gt;&gt; If you or your agent or exclusive licensee =
institute or order or agree<br>&gt;&gt; to the institution of patent =
litigation against any entity (including<br>&gt;&gt; a cross-claim or =
counterclaim in a lawsuit) alleging that this<br>&gt;&gt; implementation =
of VP8 or any code incorporated within this<br>&gt;&gt; implementation =
of VP8 constitutes direct or contributory patent<br>&gt;&gt; =
infringement, or inducement of patent infringement, then any =
patent<br>&gt;&gt; rights granted to you under this License for this =
implementation of<br>&gt;&gt; VP8 shall terminate as of the date such =
litigation is filed.<br>&gt;Perhaps I assumed that that is a very =
reasonable part of the license.<br>&gt;That is, if you are suing someone =
alleging a patent infringement within<br>&gt;VP8, you are no longer =
granted the license to use VP8's patented<br>&gt;technologies that =
Google owns.<o:p></o:p></p></div></div><p class=3DMsoNormal>Yes, that's =
one issue. &nbsp;Call it personal preference for different type =
of<br>reciprocity conditions :-) &nbsp;(I could rant about it for hours, =
but let's<br>continue to pretend that this is mostly a technical mailing =
list)<br><br>The other issue, though (the fact that the license grant =
extends only to<br>the VP8 implementation as provided by google, and =
does not extent to<br>derivative works such as hardware implementations) =
should be moderately<br>alarming even for an open source person. =
&nbsp;With respect to this clause, I<br>will note that I criticized the =
licensing conditions in private and in<br>public (IETF mike) several =
times, months ago, and nothing happened.<br>Suggests to me one of three =
things: (1) google is a large company and<br>decisions take time, or (2) =
google's legal is currently occupied with<br>other stuff, or (3) that =
the choice of language is intentional, and<br>intended to prevent forks. =
&nbsp;Take your pick.<br><span =
style=3D'color:#888888'>Stephan</span><o:p></o:p></p><div><div><p =
class=3DMsoNormal><br>&gt;_______________________________________________=
<br>&gt;rtcweb mailing list<br>&gt;<a href=3D"mailto:rtcweb@ietf.org" =
target=3D"_blank">rtcweb@ietf.org</a><br>&gt;<a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>&gt=
;<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><o:p></=
o:p></p></div></div><p class=3DMsoNormal><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank"><span style=3D'font-size:7.5pt;color:#999999'>Google =
Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. =
556656-6880&nbsp;</span><br><br><span =
style=3D'font-size:7.5pt;color:#999999'>Apparently, this footer is =
required in Europe. Apologies. This email may be confidential or =
privileged. &nbsp;If you received this communication by mistake, please =
don't forward it to anyone else,please erase all copies and attachments, =
and please let me know that it went to the wrong person. =
&nbsp;Thanks.</span><br></a><o:p></o:p></p></div></div><p =
class=3DMsoNormal><br clear=3Dall><o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>-- =
<br><span =
style=3D'font-family:"Arial","sans-serif";color:#555555;border:solid =
#D50F25 1.5pt;padding:2.0pt'>Serge Lachapelle&nbsp;|</span><span =
style=3D'font-family:"Arial","sans-serif";color:#555555;border:solid =
#3369E8 1.5pt;padding:2.0pt'>&nbsp;Product Manager&nbsp;|</span><span =
style=3D'font-family:"Arial","sans-serif";color:#555555;border:solid =
#009939 1.5pt;padding:2.0pt'>&nbsp;<a href=3D"mailto:sergel@google.com" =
target=3D"_blank">sergel@google.com</a>&nbsp;|</span><span =
style=3D'font-family:"Arial","sans-serif";color:#555555;border:solid =
#EEB211 1.5pt;padding:2.0pt'>&nbsp;+46 732 01 22 32</span><br><br =
clear=3Dall><o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>-- =
<o:p></o:p></p><div><div style=3D'margin-top:7.5pt'><p class=3DMsoNormal =
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From basilgohar@librevideo.org  Tue Apr  3 21:25:04 2012
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Message-ID: <4F7BCD1A.7020508@librevideo.org>
Date: Wed, 04 Apr 2012 00:24:58 -0400
From: Basil Mohamed Gohar <basilgohar@librevideo.org>
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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While it would be great if such a thing were possible, even the MPEG-LA 
does not indemnify users of H.264 from any IPR claims.  Expecting Google 
to do so is unreasonable.  Moreover, there are other practical issues 
such as NDAs that patent litigation frequently invokes that would still 
allow someone to be sued even if Google did offer indemnification.

The strongest thing Google can do, I think, is what they already do in 
relation to nullifying all grants to their patents that cover VP8 should 
someone engage in a lawsuit against Google.  The software patent 
landscape, sadly, does not leave many more options on the table.  
Perhaps someone more familiar with patent law can expand and/or correct 
what I've said.

It should go without saying that IANAL.

-- 
Libre Video
http://librevideo.org


On 04/03/2012 10:45 PM, Paul E. Jones wrote:
>
> Will Google indemnify all implementers against any IPR claims?  Do 
> that, and I think the debate would be over.
>
> Paul
>
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On 
> Behalf Of *Serge Lachapelle
> *Sent:* Tuesday, April 03, 2012 1:55 PM
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] Google VP8 Patent Grant for third parties [Was 
> Re:Proposal for H.263 baseline codec]
>
> [Forking the thread]
>
> Hello folks,
>
>
> Google confirms that the VP8 patent grant applies to both third-party 
> hardware and software implementations of VP8.
>
> Google encourages the community to create hardware implementations of 
> VP8, and has recently blogged about a number of new hardware 
> implementations on the WebM blog ( 
> http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware.html 
> ).
>
> Google is quite proud of what the community has done with VP8 and 
> looks forward to seeing more implementations of VP8 in both hardware 
> and software.
>
> Regards,
>
> /Serge Lachapelle, Google, Stockholm
>
> On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org 
> <mailto:stewe@stewe.org>> wrote:
>
>
>
> On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org 
> <mailto:basilgohar@librevideo.org>>
> wrote:
>
>
> >On 03/29/2012 10:20 AM, Stephan Wenger wrote:
> >> The second part of your sentence may or may not be true, depending on
> >>your
> >> relationship with google, your willingness to use the webm
> >>implementation
> >> in unchanged form, and other factors.  Please see the webm license
> >> conditions, which AFAIK can be found here:
> >> http://www.webmproject.org/license/additional/
> >Correct.  I think you are referring to this part, explicitly:
> >> If you or your agent or exclusive licensee institute or order or agree
> >> to the institution of patent litigation against any entity (including
> >> a cross-claim or counterclaim in a lawsuit) alleging that this
> >> implementation of VP8 or any code incorporated within this
> >> implementation of VP8 constitutes direct or contributory patent
> >> infringement, or inducement of patent infringement, then any patent
> >> rights granted to you under this License for this implementation of
> >> VP8 shall terminate as of the date such litigation is filed.
> >Perhaps I assumed that that is a very reasonable part of the license.
> >That is, if you are suing someone alleging a patent infringement within
> >VP8, you are no longer granted the license to use VP8's patented
> >technologies that Google owns.
>
> Yes, that's one issue.  Call it personal preference for different type of
> reciprocity conditions :-)  (I could rant about it for hours, but let's
> continue to pretend that this is mostly a technical mailing list)
>
> The other issue, though (the fact that the license grant extends only to
> the VP8 implementation as provided by google, and does not extent to
> derivative works such as hardware implementations) should be moderately
> alarming even for an open source person.  With respect to this clause, I
> will note that I criticized the licensing conditions in private and in
> public (IETF mike) several times, months ago, and nothing happened.
> Suggests to me one of three things: (1) google is a large company and
> decisions take time, or (2) google's legal is currently occupied with
> other stuff, or (3) that the choice of language is intentional, and
> intended to prevent forks.  Take your pick.
> Stephan
>
>

From paulej@packetizer.com  Tue Apr  3 22:23:30 2012
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Basil,

I disagree.  If VP8 is believed so strongly to be free of any IPR, then
Google could boldly indemnify a company.  Personally, I have my doubts that
it is.

MPEG-LA is not in the business of indemnifying people, nor do I believe they
would make a claim that H.264 is not covered by another patent holder.  That
said, I do believe that there is a significantly increased probability that
all patent holders either license through MPEG-LA or have listed IPR on
H.264 in the ITU's IPR database.

I would have a high degree of confidence that I could identify all H.264 IPR
holders.  I have no degree of confidence I could do the same for VP8.

Paul

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Basil Mohamed Gohar
> Sent: Wednesday, April 04, 2012 12:25 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was
> Re:Proposal for H.263 baseline codec]
> 
> While it would be great if such a thing were possible, even the MPEG-LA
> does not indemnify users of H.264 from any IPR claims.  Expecting Google
> to do so is unreasonable.  Moreover, there are other practical issues such
> as NDAs that patent litigation frequently invokes that would still allow
> someone to be sued even if Google did offer indemnification.
> 
> The strongest thing Google can do, I think, is what they already do in
> relation to nullifying all grants to their patents that cover VP8 should
> someone engage in a lawsuit against Google.  The software patent
> landscape, sadly, does not leave many more options on the table.
> Perhaps someone more familiar with patent law can expand and/or correct
> what I've said.
> 
> It should go without saying that IANAL.
> 
> --
> Libre Video
> http://librevideo.org
> 
> 
> On 04/03/2012 10:45 PM, Paul E. Jones wrote:
> >
> > Will Google indemnify all implementers against any IPR claims?  Do
> > that, and I think the debate would be over.
> >
> > Paul
> >
> > *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> > Behalf Of *Serge Lachapelle
> > *Sent:* Tuesday, April 03, 2012 1:55 PM
> > *To:* rtcweb@ietf.org
> > *Subject:* [rtcweb] Google VP8 Patent Grant for third parties [Was
> > Re:Proposal for H.263 baseline codec]
> >
> > [Forking the thread]
> >
> > Hello folks,
> >
> >
> > Google confirms that the VP8 patent grant applies to both third-party
> > hardware and software implementations of VP8.
> >
> > Google encourages the community to create hardware implementations of
> > VP8, and has recently blogged about a number of new hardware
> > implementations on the WebM blog (
> > http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-
> hardware.html
> > ).
> >
> > Google is quite proud of what the community has done with VP8 and
> > looks forward to seeing more implementations of VP8 in both hardware
> > and software.
> >
> > Regards,
> >
> > /Serge Lachapelle, Google, Stockholm
> >
> > On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org
> > <mailto:stewe@stewe.org>> wrote:
> >
> >
> >
> > On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org
> > <mailto:basilgohar@librevideo.org>>
> > wrote:
> >
> >
> > >On 03/29/2012 10:20 AM, Stephan Wenger wrote:
> > >> The second part of your sentence may or may not be true, depending on
> > >>your
> > >> relationship with google, your willingness to use the webm
> > >>implementation
> > >> in unchanged form, and other factors.  Please see the webm license
> > >> conditions, which AFAIK can be found here:
> > >> http://www.webmproject.org/license/additional/
> > >Correct.  I think you are referring to this part, explicitly:
> > >> If you or your agent or exclusive licensee institute or order or
> agree
> > >> to the institution of patent litigation against any entity (including
> > >> a cross-claim or counterclaim in a lawsuit) alleging that this
> > >> implementation of VP8 or any code incorporated within this
> > >> implementation of VP8 constitutes direct or contributory patent
> > >> infringement, or inducement of patent infringement, then any patent
> > >> rights granted to you under this License for this implementation of
> > >> VP8 shall terminate as of the date such litigation is filed.
> > >Perhaps I assumed that that is a very reasonable part of the license.
> > >That is, if you are suing someone alleging a patent infringement within
> > >VP8, you are no longer granted the license to use VP8's patented
> > >technologies that Google owns.
> >
> > Yes, that's one issue.  Call it personal preference for different type
> of
> > reciprocity conditions :-)  (I could rant about it for hours, but let's
> > continue to pretend that this is mostly a technical mailing list)
> >
> > The other issue, though (the fact that the license grant extends only to
> > the VP8 implementation as provided by google, and does not extent to
> > derivative works such as hardware implementations) should be moderately
> > alarming even for an open source person.  With respect to this clause, I
> > will note that I criticized the licensing conditions in private and in
> > public (IETF mike) several times, months ago, and nothing happened.
> > Suggests to me one of three things: (1) google is a large company and
> > decisions take time, or (2) google's legal is currently occupied with
> > other stuff, or (3) that the choice of language is intentional, and
> > intended to prevent forks.  Take your pick.
> > Stephan
> >
> >
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From basilgohar@librevideo.org  Wed Apr  4 06:42:18 2012
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Paul,

What you are asking for is something no one in the media industry offers
for something comparable.  As was pointed out elsewhere on this list,
even the H.264 standard has been updated to include additional patents
over time deemed to be "essential".  One could be non-infringing in the
past and suddenly find yourself infringing.

I don't know what all of Google's public statements about VP8 IPR status
are, but we know that the patents they own are granted via the license,
and we know that a public call for the formation of a patent pool over a
year ago has yielded not even a follow-up announcement from MPEG-LA.

Furthermore, not a single patent has been brought forward publicly as
being even remotely doubtful as to its application to VP8.

So, until there is some actual evidence that there is an IPR threat,
then with all due respect, statements such as yours must be classified
as spreading FUD, because they serve to drive adoption away from a free,
open, unencumbered standard to one that is on baseless claims.

-- 
Libre Video
http://librevideo.org



On 04/04/2012 01:23 AM, Paul E. Jones wrote:
> Basil,
>
> I disagree.  If VP8 is believed so strongly to be free of any IPR, then
> Google could boldly indemnify a company.  Personally, I have my doubts that
> it is.
>
> MPEG-LA is not in the business of indemnifying people, nor do I believe they
> would make a claim that H.264 is not covered by another patent holder.  That
> said, I do believe that there is a significantly increased probability that
> all patent holders either license through MPEG-LA or have listed IPR on
> H.264 in the ITU's IPR database.
>
> I would have a high degree of confidence that I could identify all H.264 IPR
> holders.  I have no degree of confidence I could do the same for VP8.
>
> Paul
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Basil Mohamed Gohar
>> Sent: Wednesday, April 04, 2012 12:25 AM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was
>> Re:Proposal for H.263 baseline codec]
>>
>> While it would be great if such a thing were possible, even the MPEG-LA
>> does not indemnify users of H.264 from any IPR claims.  Expecting Google
>> to do so is unreasonable.  Moreover, there are other practical issues such
>> as NDAs that patent litigation frequently invokes that would still allow
>> someone to be sued even if Google did offer indemnification.
>>
>> The strongest thing Google can do, I think, is what they already do in
>> relation to nullifying all grants to their patents that cover VP8 should
>> someone engage in a lawsuit against Google.  The software patent
>> landscape, sadly, does not leave many more options on the table.
>> Perhaps someone more familiar with patent law can expand and/or correct
>> what I've said.
>>
>> It should go without saying that IANAL.
>>
>> --
>> Libre Video
>> http://librevideo.org
>>
>>
>> On 04/03/2012 10:45 PM, Paul E. Jones wrote:
>>> Will Google indemnify all implementers against any IPR claims?  Do
>>> that, and I think the debate would be over.
>>>
>>> Paul
>>>
>>> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
>>> Behalf Of *Serge Lachapelle
>>> *Sent:* Tuesday, April 03, 2012 1:55 PM
>>> *To:* rtcweb@ietf.org
>>> *Subject:* [rtcweb] Google VP8 Patent Grant for third parties [Was
>>> Re:Proposal for H.263 baseline codec]
>>>
>>> [Forking the thread]
>>>
>>> Hello folks,
>>>
>>>
>>> Google confirms that the VP8 patent grant applies to both third-party
>>> hardware and software implementations of VP8.
>>>
>>> Google encourages the community to create hardware implementations of
>>> VP8, and has recently blogged about a number of new hardware
>>> implementations on the WebM blog (
>>> http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-
>> hardware.html
>>> ).
>>>
>>> Google is quite proud of what the community has done with VP8 and
>>> looks forward to seeing more implementations of VP8 in both hardware
>>> and software.
>>>
>>> Regards,
>>>
>>> /Serge Lachapelle, Google, Stockholm
>>>
>>> On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org
>>> <mailto:stewe@stewe.org>> wrote:
>>>
>>>
>>>
>>> On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org
>>> <mailto:basilgohar@librevideo.org>>
>>> wrote:
>>>
>>>
>>>> On 03/29/2012 10:20 AM, Stephan Wenger wrote:
>>>>> The second part of your sentence may or may not be true, depending on
>>>>> your
>>>>> relationship with google, your willingness to use the webm
>>>>> implementation
>>>>> in unchanged form, and other factors.  Please see the webm license
>>>>> conditions, which AFAIK can be found here:
>>>>> http://www.webmproject.org/license/additional/
>>>> Correct.  I think you are referring to this part, explicitly:
>>>>> If you or your agent or exclusive licensee institute or order or
>> agree
>>>>> to the institution of patent litigation against any entity (including
>>>>> a cross-claim or counterclaim in a lawsuit) alleging that this
>>>>> implementation of VP8 or any code incorporated within this
>>>>> implementation of VP8 constitutes direct or contributory patent
>>>>> infringement, or inducement of patent infringement, then any patent
>>>>> rights granted to you under this License for this implementation of
>>>>> VP8 shall terminate as of the date such litigation is filed.
>>>> Perhaps I assumed that that is a very reasonable part of the license.
>>>> That is, if you are suing someone alleging a patent infringement within
>>>> VP8, you are no longer granted the license to use VP8's patented
>>>> technologies that Google owns.
>>> Yes, that's one issue.  Call it personal preference for different type
>> of
>>> reciprocity conditions :-)  (I could rant about it for hours, but let's
>>> continue to pretend that this is mostly a technical mailing list)
>>>
>>> The other issue, though (the fact that the license grant extends only to
>>> the VP8 implementation as provided by google, and does not extent to
>>> derivative works such as hardware implementations) should be moderately
>>> alarming even for an open source person.  With respect to this clause, I
>>> will note that I criticized the licensing conditions in private and in
>>> public (IETF mike) several times, months ago, and nothing happened.
>>> Suggests to me one of three things: (1) google is a large company and
>>> decisions take time, or (2) google's legal is currently occupied with
>>> other stuff, or (3) that the choice of language is intentional, and
>>> intended to prevent forks.  Take your pick.
>>> Stephan
>>>
>>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb



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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 4 Apr 2012 18:43:47 +0200
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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2012/4/3 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> Hi all,
>
> I've made two "pictures" showing WebRTC and SIP interop for two cases:
>
> 1) SDES-SRTP is allowed in WebRTC:
> =C2=A0 =C2=A0 =C2=A0http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDE=
S-SRTP.png
>
> 2) Just DTLS-ETK-SRTP is allowed in WebRTC [*]
> =C2=A0 =C2=A0 =C2=A0http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTL=
S-EKT-SRTP.png
>
> [*] slides 30-35 in
> http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf
>
>
> For those claiming to mandate *just* DTLS-EKT-SRTP in WebRTC, please
> see the *cost* of such a decision, and also:
>
> - Thanks for requiring a super Signaling+Media B2BUA/SBC in WebRTC/SIP
> interop scenarios. Some vendors will be very happy and will become
> very rich. Such a super device (also a DTLS to SDES conversor,
> including DTLS key updates to re-INVITE) will be "a bit"... expensive.
>
> - Thanks for disallowing *pure* SIP protocol usage (and instead
> requiring SIP B2BUAs/SBCs or custom WebRTC signaling to SIP conversion
> gateways). WebRTC is supposed to let the signaling protocol up to the
> application, but pure SIP protocol will not be possible since a SIP
> B2BUA/SBC is required, and those devices always break/limit the SIP
> protocol (*always*).
>
>
> So IMHO, option 2 ("just DTLS-EKT-SRTP is allowed in WebRTC") is The Barr=
ier.


Hi, nobody cares about the implications of option 2 ???

Do all the people planning to interop with SIP assume that they'll
need the super B2BUA in the second image (without the possibility of
using a pure SIP proxy)?:

  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png


BTW Fabio has sent two magistral mails explaining why DTLS is not the
panacea and why SDES can be good enough to satisfy the same level of
security than DTLS. No comments?


Regards.


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

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--e89a8fb208dcbf527804bcddb251
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On Wed, Apr 4, 2012 at 12:43 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Hi, nobody cares about the implications of option 2 ???
>
> Do all the people planning to interop with SIP assume that they'll
> need the super B2BUA in the second image (without the possibility of
> using a pure SIP proxy)?:
>
>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png
>
>
<sarcasm>I guess we should look at this as a business opportunity ;) I am
not sure why you assume that building such gateway would take 10 years. I
can sell a gateway like this to anybody who needs it right now.</sarcasm>

My assumption is that IP phones will migrate to WebRTC effectively
eliminating SIP in end user devices. The only place where SIP will remain
would be federation, which commonly uses some sort of SBC anyway. This SBC
will need to be extended to support ICE anyway. Might as well through in
support for DTLS-EKV-SRTP. I doubt it will map DTLS key updates to
re-invites. Most probably it will simply re-encode.

______________
Roman Shpount

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<div class=3D"gmail_quote">On Wed, Apr 4, 2012 at 12:43 PM, I=F1aki Baz Cas=
tillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net<=
/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">
Hi, nobody cares about the implications of option 2 ???<br>
<br>
Do all the people planning to interop with SIP assume that they&#39;ll<br>
need the super B2BUA in the second image (without the possibility of<br>
using a pure SIP proxy)?:<br>
<br>
 =A0<a href=3D"http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-S=
RTP.png" target=3D"_blank">http://public.aliax.net/WebRTC/WebRTC_SIP_Intero=
p_DTLS-EKT-SRTP.png</a><br>
<br></blockquote></div><br>&lt;sarcasm&gt;I guess we should look at this as=
 a business opportunity ;) I am not sure why you assume that building such =
gateway would take 10 years. I can sell a gateway like this to anybody who =
needs it right now.&lt;/sarcasm&gt;<br>
<br>My assumption is that IP phones will migrate to WebRTC effectively elim=
inating SIP in end user devices. The only place where SIP will remain would=
 be federation, which commonly uses some sort of SBC anyway. This SBC will =
need to be extended to support ICE anyway. Might as well through in support=
 for DTLS-EKV-SRTP. I doubt it will map DTLS key updates to re-invites. Mos=
t probably it will simply re-encode.<br>
=A0<br>______________<br>Roman Shpount<br>

--e89a8fb208dcbf527804bcddb251--

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2012/4/4 Roman Shpount <roman@telurix.com>:
> On Wed, Apr 4, 2012 at 12:43 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:
>>
>> Hi, nobody cares about the implications of option 2 ???
>>
>> Do all the people planning to interop with SIP assume that they'll
>> need the super B2BUA in the second image (without the possibility of
>> using a pure SIP proxy)?:
>>
>> =C2=A0http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.pn=
g
>>
>
> <sarcasm>I guess we should look at this as a business opportunity ;) I am
> not sure why you assume that building such gateway would take 10 years. I
> can sell a gateway like this to anybody who needs it right now.</sarcasm>
>
> My assumption is that IP phones will migrate to WebRTC effectively
> eliminating SIP in end user devices. The only place where SIP will remain
> would be federation, which commonly uses some sort of SBC anyway. This SB=
C
> will need to be extended to support ICE anyway. Might as well through in
> support for DTLS-EKV-SRTP. I doubt it will map DTLS key updates to
> re-invites. Most probably it will simply re-encode.

Roman... I hope you missed the </sarcasm> closing tag just HERE, am I
right? XDDD

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

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From: Serge Lachapelle <sergel@google.com>
Date: Wed, 4 Apr 2012 20:44:34 +0200
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Thank you for your feedback Basil. Google posted this as a result:

http://www.webmproject.org/about/faq/#does-the-vp8-patent-grant-also-extend-to-hardware-implementations

/Serge

On Tue, Apr 3, 2012 at 20:37, Basil Mohamed Gohar <basilgohar@librevideo.org
> wrote:

> Serge,
>
> Thank you very much for this.  To go a step forward, I would encourage
> that this be made more legally obvious, because a reference to a single
> e-mail on an "obscure" mailing list (relative to what most people are
> familiar with) will not provide the deepest levels of assurance to those
> not actively involved with the WebRTC project.
>
> I would humbly request an unambiguous statement somewhere on the WebM
> project website that makes this assertion clear, because what is
> publicly available is, at the very least, ambiguous, and at the worst,
> "evidence of Google's nefarious scheme to pounce on hardware
> implementers after wide adoption".
>
> On 04/03/2012 01:54 PM, Serge Lachapelle wrote:
> > *[Forking the thread]*
> > *
> >
> >
> > *
> > *Hello folks,*
> >
> > Google confirms that the VP8 patent grant applies to both third-party
> > hardware and software implementations of VP8.
> > *
> > *
> > *Google encourages the community to create hardware implementations of
> > VP8, and has recently blogged about a number of new hardware
> > implementations on the WebM blog (
> >
> http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware.html
> > ). *
> > *
> > *
> > *Google is quite proud of what the community has done with VP8 and
> > looks forward to seeing more implementations of VP8 in both hardware
> > and software. *
> > *
> > *
> > *Regards,*
> > *
> > *
> > */Serge Lachapelle,
> > Google, Stockholm*
> >
> > On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org
> > <mailto:stewe@stewe.org>> wrote:
> >
> >
> >
> >     On 3.29.2012 16:24 , "Basil Mohamed Gohar"
> >     <basilgohar@librevideo.org <mailto:basilgohar@librevideo.org>>
> >     wrote:
> >
> >     >On 03/29/2012 10:20 AM, Stephan Wenger wrote:
> >     >> The second part of your sentence may or may not be true,
> >     depending on
> >     >>your
> >     >> relationship with google, your willingness to use the webm
> >     >>implementation
> >     >> in unchanged form, and other factors.  Please see the webm license
> >     >> conditions, which AFAIK can be found here:
> >     >> http://www.webmproject.org/license/additional/
> >     >Correct.  I think you are referring to this part, explicitly:
> >     >> If you or your agent or exclusive licensee institute or order
> >     or agree
> >     >> to the institution of patent litigation against any entity
> >     (including
> >     >> a cross-claim or counterclaim in a lawsuit) alleging that this
> >     >> implementation of VP8 or any code incorporated within this
> >     >> implementation of VP8 constitutes direct or contributory patent
> >     >> infringement, or inducement of patent infringement, then any
> patent
> >     >> rights granted to you under this License for this implementation
> of
> >     >> VP8 shall terminate as of the date such litigation is filed.
> >     >Perhaps I assumed that that is a very reasonable part of the
> license.
> >     >That is, if you are suing someone alleging a patent infringement
> >     within
> >     >VP8, you are no longer granted the license to use VP8's patented
> >     >technologies that Google owns.
> >
> >     Yes, that's one issue.  Call it personal preference for different
> >     type of
> >     reciprocity conditions :-)  (I could rant about it for hours, but
> >     let's
> >     continue to pretend that this is mostly a technical mailing list)
> >
> >     The other issue, though (the fact that the license grant extends
> >     only to
> >     the VP8 implementation as provided by google, and does not extent to
> >     derivative works such as hardware implementations) should be
> >     moderately
> >     alarming even for an open source person.  With respect to this
> >     clause, I
> >     will note that I criticized the licensing conditions in private and
> in
> >     public (IETF mike) several times, months ago, and nothing happened.
> >     Suggests to me one of three things: (1) google is a large company and
> >     decisions take time, or (2) google's legal is currently occupied with
> >     other stuff, or (3) that the choice of language is intentional, and
> >     intended to prevent forks.  Take your pick.
> >     Stephan
> >
> >     >_______________________________________________
> >     >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
> >     Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr.
> >     556656-6880
> >
> >     Apparently, this footer is required in Europe. Apologies. This
> >     email may be confidential or privileged.  If you received this
> >     communication by mistake, please don't forward it to anyone
> >     else,please erase all copies and attachments, and please let me
> >     know that it went to the wrong person.  Thanks.
> >     <https://www.ietf.org/mailman/listinfo/rtcweb>
> >
> >
> >
> > --
> > Serge Lachapelle | Product Manager | sergel@google.com
> > <mailto:sergel@google.com> | +46 732 01 22 32
> >
> >
> > --
> > Serge Lachapelle | Product Manager | sergel@google.com
> > <mailto:sergel@google.com> | +46 732 01 22 32
> > Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr.
> > 556656-6880
> >
> > Apparently, this footer is required in Europe. Apologies. This email
> > may be confidential or privileged.  If you received this communication
> > by mistake, please don't forward it to anyone else,please erase all
> > copies and attachments, and please let me know that it went to the
> > wrong person.  Thanks.
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> --
> Libre Video
> http://librevideo.org
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



-- 
Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32
Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880

Apparently, this footer is required in Europe. Apologies. This email may be
confidential or privileged.  If you received this communication by mistake,
please don't forward it to anyone else,please erase all copies and
attachments, and please let me know that it went to the wrong person.
 Thanks.

--e89a8f503c5853e33204bcded2d8
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Thank you for your feedback Basil. Google posted this as a result:<div><br>=
</div><div><a href=3D"http://www.webmproject.org/about/faq/#does-the-vp8-pa=
tent-grant-also-extend-to-hardware-implementations">http://www.webmproject.=
org/about/faq/#does-the-vp8-patent-grant-also-extend-to-hardware-implementa=
tions</a></div>

<div><br></div><div>/Serge<br><div><br><div class=3D"gmail_quote">On Tue, A=
pr 3, 2012 at 20:37, Basil Mohamed Gohar <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:basilgohar@librevideo.org">basilgohar@librevideo.org</a>&gt;</span> w=
rote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Serge,<br>
<br>
Thank you very much for this. =A0To go a step forward, I would encourage<br=
>
that this be made more legally obvious, because a reference to a single<br>
e-mail on an &quot;obscure&quot; mailing list (relative to what most people=
 are<br>
familiar with) will not provide the deepest levels of assurance to those<br=
>
not actively involved with the WebRTC project.<br>
<br>
I would humbly request an unambiguous statement somewhere on the WebM<br>
project website that makes this assertion clear, because what is<br>
publicly available is, at the very least, ambiguous, and at the worst,<br>
&quot;evidence of Google&#39;s nefarious scheme to pounce on hardware<br>
implementers after wide adoption&quot;.<br>
<br>
On 04/03/2012 01:54 PM, Serge Lachapelle wrote:<br>
&gt; *[Forking the thread]*<br>
&gt; *<br>
&gt;<br>
&gt;<br>
&gt; *<br>
&gt; *Hello folks,*<br>
<div class=3D"im">&gt;<br>
&gt; Google confirms that the VP8 patent grant applies to both third-party<=
br>
&gt; hardware and software implementations of VP8.<br>
</div>&gt; *<br>
&gt; *<br>
&gt; *Google encourages the community to create hardware implementations of=
<br>
<div class=3D"im">&gt; VP8, and has recently blogged about a number of new =
hardware<br>
&gt; implementations on the WebM blog (<br>
&gt; <a href=3D"http://blog.webmproject.org/2012/03/webm-gaining-momentum-i=
n-hardware.html" target=3D"_blank">http://blog.webmproject.org/2012/03/webm=
-gaining-momentum-in-hardware.html</a><br>
</div>&gt; ). *<br>
&gt; *<br>
&gt; *<br>
&gt; *Google is quite proud of what the community has done with VP8 and<br>
<div class=3D"im">&gt; looks forward to seeing more implementations of VP8 =
in both hardware<br>
</div>&gt; and software. *<br>
&gt; *<br>
&gt; *<br>
&gt; *Regards,*<br>
&gt; *<br>
&gt; *<br>
&gt; */Serge Lachapelle,<br>
&gt; Google, Stockholm*<br>
<div class=3D"im">&gt;<br>
&gt; On Thu, Mar 29, 2012 at 16:53, Stephan Wenger &lt;<a href=3D"mailto:st=
ewe@stewe.org">stewe@stewe.org</a><br>
</div><div class=3D"im">&gt; &lt;mailto:<a href=3D"mailto:stewe@stewe.org">=
stewe@stewe.org</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 On 3.29.2012 16:24 , &quot;Basil Mohamed Gohar&quot;<br>
</div>&gt; =A0 =A0 &lt;<a href=3D"mailto:basilgohar@librevideo.org">basilgo=
har@librevideo.org</a> &lt;mailto:<a href=3D"mailto:basilgohar@librevideo.o=
rg">basilgohar@librevideo.org</a>&gt;&gt;<br>
<div><div class=3D"h5">&gt; =A0 =A0 wrote:<br>
&gt;<br>
&gt; =A0 =A0 &gt;On 03/29/2012 10:20 AM, Stephan Wenger wrote:<br>
&gt; =A0 =A0 &gt;&gt; The second part of your sentence may or may not be tr=
ue,<br>
&gt; =A0 =A0 depending on<br>
&gt; =A0 =A0 &gt;&gt;your<br>
&gt; =A0 =A0 &gt;&gt; relationship with google, your willingness to use the=
 webm<br>
&gt; =A0 =A0 &gt;&gt;implementation<br>
&gt; =A0 =A0 &gt;&gt; in unchanged form, and other factors. =A0Please see t=
he webm license<br>
&gt; =A0 =A0 &gt;&gt; conditions, which AFAIK can be found here:<br>
&gt; =A0 =A0 &gt;&gt; <a href=3D"http://www.webmproject.org/license/additio=
nal/" target=3D"_blank">http://www.webmproject.org/license/additional/</a><=
br>
&gt; =A0 =A0 &gt;Correct. =A0I think you are referring to this part, explic=
itly:<br>
&gt; =A0 =A0 &gt;&gt; If you or your agent or exclusive licensee institute =
or order<br>
&gt; =A0 =A0 or agree<br>
&gt; =A0 =A0 &gt;&gt; to the institution of patent litigation against any e=
ntity<br>
&gt; =A0 =A0 (including<br>
&gt; =A0 =A0 &gt;&gt; a cross-claim or counterclaim in a lawsuit) alleging =
that this<br>
&gt; =A0 =A0 &gt;&gt; implementation of VP8 or any code incorporated within=
 this<br>
&gt; =A0 =A0 &gt;&gt; implementation of VP8 constitutes direct or contribut=
ory patent<br>
&gt; =A0 =A0 &gt;&gt; infringement, or inducement of patent infringement, t=
hen any patent<br>
&gt; =A0 =A0 &gt;&gt; rights granted to you under this License for this imp=
lementation of<br>
&gt; =A0 =A0 &gt;&gt; VP8 shall terminate as of the date such litigation is=
 filed.<br>
&gt; =A0 =A0 &gt;Perhaps I assumed that that is a very reasonable part of t=
he license.<br>
&gt; =A0 =A0 &gt;That is, if you are suing someone alleging a patent infrin=
gement<br>
&gt; =A0 =A0 within<br>
&gt; =A0 =A0 &gt;VP8, you are no longer granted the license to use VP8&#39;=
s patented<br>
&gt; =A0 =A0 &gt;technologies that Google owns.<br>
&gt;<br>
&gt; =A0 =A0 Yes, that&#39;s one issue. =A0Call it personal preference for =
different<br>
&gt; =A0 =A0 type of<br>
&gt; =A0 =A0 reciprocity conditions :-) =A0(I could rant about it for hours=
, but<br>
&gt; =A0 =A0 let&#39;s<br>
&gt; =A0 =A0 continue to pretend that this is mostly a technical mailing li=
st)<br>
&gt;<br>
&gt; =A0 =A0 The other issue, though (the fact that the license grant exten=
ds<br>
&gt; =A0 =A0 only to<br>
&gt; =A0 =A0 the VP8 implementation as provided by google, and does not ext=
ent to<br>
&gt; =A0 =A0 derivative works such as hardware implementations) should be<b=
r>
&gt; =A0 =A0 moderately<br>
&gt; =A0 =A0 alarming even for an open source person. =A0With respect to th=
is<br>
&gt; =A0 =A0 clause, I<br>
&gt; =A0 =A0 will note that I criticized the licensing conditions in privat=
e and in<br>
&gt; =A0 =A0 public (IETF mike) several times, months ago, and nothing happ=
ened.<br>
&gt; =A0 =A0 Suggests to me one of three things: (1) google is a large comp=
any and<br>
&gt; =A0 =A0 decisions take time, or (2) google&#39;s legal is currently oc=
cupied with<br>
&gt; =A0 =A0 other stuff, or (3) that the choice of language is intentional=
, and<br>
&gt; =A0 =A0 intended to prevent forks. =A0Take your pick.<br>
&gt; =A0 =A0 Stephan<br>
&gt;<br>
&gt; =A0 =A0 &gt;_______________________________________________<br>
&gt; =A0 =A0 &gt;rtcweb mailing list<br>
</div></div>&gt; =A0 =A0 &gt;<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>
<div class=3D"im">&gt; =A0 =A0 &gt;<a href=3D"https://www.ietf.org/mailman/=
listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rt=
cweb</a><br>
&gt; =A0 =A0 &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">rtcweb@ietf.org</a> &=
lt;mailto:<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<div class=3D"im">&gt; =A0 =A0 <a href=3D"https://www.ietf.org/mailman/list=
info/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb=
</a><br>
&gt; =A0 =A0 Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr.=
<br>
&gt; =A0 =A0 556656-6880<br>
&gt;<br>
&gt; =A0 =A0 Apparently, this footer is required in Europe. Apologies. This=
<br>
&gt; =A0 =A0 email may be confidential or privileged. =A0If you received th=
is<br>
&gt; =A0 =A0 communication by mistake, please don&#39;t forward it to anyon=
e<br>
&gt; =A0 =A0 else,please erase all copies and attachments, and please let m=
e<br>
&gt; =A0 =A0 know that it went to the wrong person. =A0Thanks.<br>
</div>&gt; =A0 =A0 &lt;<a href=3D"https://www.ietf.org/mailman/listinfo/rtc=
web" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a>&gt;=
<br>
<div class=3D"im">&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Serge Lachapelle | Product Manager | <a href=3D"mailto:sergel@google.c=
om">sergel@google.com</a><br>
</div>&gt; &lt;mailto:<a href=3D"mailto:sergel@google.com">sergel@google.co=
m</a>&gt; | <a href=3D"tel:%2B46%20732%2001%2022%2032" value=3D"+4673201223=
2">+46 732 01 22 32</a><br>
<div class=3D"im">&gt;<br>
&gt;<br>
&gt; --<br>
&gt; Serge Lachapelle | Product Manager | <a href=3D"mailto:sergel@google.c=
om">sergel@google.com</a><br>
</div>&gt; &lt;mailto:<a href=3D"mailto:sergel@google.com">sergel@google.co=
m</a>&gt; | <a href=3D"tel:%2B46%20732%2001%2022%2032" value=3D"+4673201223=
2">+46 732 01 22 32</a><br>
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ckholm | Org. nr.<br>
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&gt; copies and attachments, and please let me know that it went to the<br>
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&gt;<br>
&gt;<br>
</div><div class=3D"im HOEnZb">&gt; _______________________________________=
________<br>
&gt; rtcweb mailing list<br>
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<br>
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Libre Video<br>
<a href=3D"http://librevideo.org" target=3D"_blank">http://librevideo.org</=
a><br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
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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>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<meta charset=3D"utf-8"><div><meta charset=3D"utf-8"><div style=3D"line-hei=
ght:1.5em;padding-top:10px;margin-top:10px;color:rgb(85,85,85);font-family:=
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<span style=3D"border-top-width:2px;border-right-width:0px;border-bottom-wi=
dth:0px;border-left-width:0px;border-top-style:solid;border-right-style:sol=
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37);border-left-color:rgb(213,15,37);padding-top:2px;margin-top:2px">Serge =
Lachapelle=A0|</span><span style=3D"border-top-width:2px;border-right-width=
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ottom-color:rgb(51,105,232);border-left-color:rgb(51,105,232);padding-top:2=
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d;border-left-style:solid;border-top-color:rgb(0,153,57);border-right-color=
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m" target=3D"_blank">sergel@google.com</a>=A0|</span><span style=3D"border-=
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t-color:rgb(238,178,17);border-bottom-color:rgb(238,178,17);border-left-col=
or:rgb(238,178,17);padding-top:2px;margin-top:2px">=A0+46 732 01 22 32</spa=
n></div>

</div><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">Google =
Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880=A0</fon=
t><br><br><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">App=
arently, this footer is required in Europe. Apologies. This email may be co=
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 please don&#39;t forward it to anyone else,please erase all copies and att=
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s.</font><br>


</div></div>

--e89a8f503c5853e33204bcded2d8--

From basilgohar@librevideo.org  Wed Apr  4 11:53:50 2012
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Date: Wed, 04 Apr 2012 14:53:44 -0400
From: Basil Mohamed Gohar <basilgohar@librevideo.org>
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Serge,

Thank you very much for this much-needed, public clarification.  Just
one more bit of FUD knocked down!

-- 
Libre Video
http://librevideo.org


On 04/04/2012 02:44 PM, Serge Lachapelle wrote:
> Thank you for your feedback Basil. Google posted this as a result:
>
> http://www.webmproject.org/about/faq/#does-the-vp8-patent-grant-also-extend-to-hardware-implementations
>
> /Serge
>
> On Tue, Apr 3, 2012 at 20:37, Basil Mohamed Gohar
> <basilgohar@librevideo.org <mailto:basilgohar@librevideo.org>> wrote:
>
>     Serge,
>
>     Thank you very much for this.  To go a step forward, I would encourage
>     that this be made more legally obvious, because a reference to a
>     single
>     e-mail on an "obscure" mailing list (relative to what most people are
>     familiar with) will not provide the deepest levels of assurance to
>     those
>     not actively involved with the WebRTC project.
>
>     I would humbly request an unambiguous statement somewhere on the WebM
>     project website that makes this assertion clear, because what is
>     publicly available is, at the very least, ambiguous, and at the worst,
>     "evidence of Google's nefarious scheme to pounce on hardware
>     implementers after wide adoption".
>
>     On 04/03/2012 01:54 PM, Serge Lachapelle wrote:
>     > *[Forking the thread]*
>     > *
>     >
>     >
>     > *
>     > *Hello folks,*
>     >
>     > Google confirms that the VP8 patent grant applies to both
>     third-party
>     > hardware and software implementations of VP8.
>     > *
>     > *
>     > *Google encourages the community to create hardware
>     implementations of
>     > VP8, and has recently blogged about a number of new hardware
>     > implementations on the WebM blog (
>     >
>     http://blog.webmproject.org/2012/03/webm-gaining-momentum-in-hardware.html
>     > ). *
>     > *
>     > *
>     > *Google is quite proud of what the community has done with VP8 and
>     > looks forward to seeing more implementations of VP8 in both hardware
>     > and software. *
>     > *
>     > *
>     > *Regards,*
>     > *
>     > *
>     > */Serge Lachapelle,
>     > Google, Stockholm*
>     >
>     > On Thu, Mar 29, 2012 at 16:53, Stephan Wenger <stewe@stewe.org
>     <mailto:stewe@stewe.org>
>     > <mailto:stewe@stewe.org <mailto:stewe@stewe.org>>> wrote:
>     >
>     >
>     >
>     >     On 3.29.2012 16:24 , "Basil Mohamed Gohar"
>     >     <basilgohar@librevideo.org
>     <mailto:basilgohar@librevideo.org>
>     <mailto:basilgohar@librevideo.org <mailto:basilgohar@librevideo.org>>>
>     >     wrote:
>     >
>     >     >On 03/29/2012 10:20 AM, Stephan Wenger wrote:
>     >     >> The second part of your sentence may or may not be true,
>     >     depending on
>     >     >>your
>     >     >> relationship with google, your willingness to use the webm
>     >     >>implementation
>     >     >> in unchanged form, and other factors.  Please see the
>     webm license
>     >     >> conditions, which AFAIK can be found here:
>     >     >> http://www.webmproject.org/license/additional/
>     >     >Correct.  I think you are referring to this part, explicitly:
>     >     >> If you or your agent or exclusive licensee institute or order
>     >     or agree
>     >     >> to the institution of patent litigation against any entity
>     >     (including
>     >     >> a cross-claim or counterclaim in a lawsuit) alleging that
>     this
>     >     >> implementation of VP8 or any code incorporated within this
>     >     >> implementation of VP8 constitutes direct or contributory
>     patent
>     >     >> infringement, or inducement of patent infringement, then
>     any patent
>     >     >> rights granted to you under this License for this
>     implementation of
>     >     >> VP8 shall terminate as of the date such litigation is filed.
>     >     >Perhaps I assumed that that is a very reasonable part of
>     the license.
>     >     >That is, if you are suing someone alleging a patent
>     infringement
>     >     within
>     >     >VP8, you are no longer granted the license to use VP8's
>     patented
>     >     >technologies that Google owns.
>     >
>     >     Yes, that's one issue.  Call it personal preference for
>     different
>     >     type of
>     >     reciprocity conditions :-)  (I could rant about it for
>     hours, but
>     >     let's
>     >     continue to pretend that this is mostly a technical mailing
>     list)
>     >
>     >     The other issue, though (the fact that the license grant extends
>     >     only to
>     >     the VP8 implementation as provided by google, and does not
>     extent to
>     >     derivative works such as hardware implementations) should be
>     >     moderately
>     >     alarming even for an open source person.  With respect to this
>     >     clause, I
>     >     will note that I criticized the licensing conditions in
>     private and in
>     >     public (IETF mike) several times, months ago, and nothing
>     happened.
>     >     Suggests to me one of three things: (1) google is a large
>     company and
>     >     decisions take time, or (2) google's legal is currently
>     occupied with
>     >     other stuff, or (3) that the choice of language is
>     intentional, and
>     >     intended to prevent forks.  Take your pick.
>     >     Stephan
>     >
>     >     >_______________________________________________
>     >     >rtcweb mailing list
>     >     >rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     <mailto:rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>     >     >https://www.ietf.org/mailman/listinfo/rtcweb
>     >     >
>     >
>     >
>     >     _______________________________________________
>     >     rtcweb mailing list
>     >     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     <mailto:rtcweb@ietf.org <mailto:rtcweb@ietf.org>>
>     >     https://www.ietf.org/mailman/listinfo/rtcweb
>     >     Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr.
>     >     556656-6880
>     >
>     >     Apparently, this footer is required in Europe. Apologies. This
>     >     email may be confidential or privileged.  If you received this
>     >     communication by mistake, please don't forward it to anyone
>     >     else,please erase all copies and attachments, and please let me
>     >     know that it went to the wrong person.  Thanks.
>     >     <https://www.ietf.org/mailman/listinfo/rtcweb>
>     >
>     >
>     >
>     > --
>     > Serge Lachapelle | Product Manager | sergel@google.com
>     <mailto:sergel@google.com>
>     > <mailto:sergel@google.com <mailto:sergel@google.com>> | +46 732
>     01 22 32 <tel:%2B46%20732%2001%2022%2032>
>     >
>     >
>     > --
>     > Serge Lachapelle | Product Manager | sergel@google.com
>     <mailto:sergel@google.com>
>     > <mailto:sergel@google.com <mailto:sergel@google.com>> | +46 732
>     01 22 32 <tel:%2B46%20732%2001%2022%2032>
>     > Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr.
>     > 556656-6880
>     >
>     > Apparently, this footer is required in Europe. Apologies. This email
>     > may be confidential or privileged.  If you received this
>     communication
>     > by mistake, please don't forward it to anyone else,please erase all
>     > copies and attachments, and please let me know that it went to the
>     > wrong person.  Thanks.
>     >
>     >
>     > _______________________________________________
>     > rtcweb mailing list
>     > rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     > https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>     --
>     Libre Video
>     http://librevideo.org
>

-- 
Libre Video
http://librevideo.org


From harald@alvestrand.no  Wed Apr  4 15:08:24 2012
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On 04/04/2012 07:23 AM, Paul E. Jones wrote:
> I would have a high degree of confidence that I could identify all H.264 IPR
> holders.
Question .... how would you do that?
I'm genuinely curious. The subject has come up in other discussions, 
including in ISO in connection with the WebVC and IVC projects.

The database in question is an ISO database, not an ITU database, and 
the format is most un-database-like; it is here:

http://isotc.iso.org/livelink/livelink/fetch/2000/2122/3770791/JTC1_Patents_database.html?nodeid=3777806&vernum=-2 
<http://isotc.iso.org/livelink/livelink/fetch/2000/2122/3770791/JTC1_Patents_database.html?nodeid=3777806&vernum=-2>




From chat2jesse@gmail.com  Wed Apr  4 17:42:46 2012
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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On Apr 4, 2012 10:24 AM, "Roman Shpount" <roman@telurix.com> wrote:
>
> On Wed, Apr 4, 2012 at 12:43 PM, I=F1aki Baz Castillo <ibc@aliax.net> wro=
te:
>>
>> Hi, nobody cares about the implications of option 2 ???
>>
>> Do all the people planning to interop with SIP assume that they'll
>> need the super B2BUA in the second image (without the possibility of
>> using a pure SIP proxy)?:
>>
>>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png
>>
>
> <sarcasm>I guess we should look at this as a business opportunity ;) I am
not sure why you assume that building such gateway would take 10 years. I
can sell a gateway like this to anybody who needs it right now.</sarcasm>
>
> My assumption is that IP phones will migrate to WebRTC effectively
eliminating SIP in end user devices.

That is your academic assumption, why does IT department need to spend
extra money to either desert or upgrade its existing SIP devices without
substantial gain? After all, tons of users are still using window xp.

Backward capability is a key to the success of new technology.

-Jesse

The only place where SIP will remain would be federation, which commonly
uses some sort of SBC anyway. This SBC will need to be extended to support
ICE anyway. Might as well through in support for DTLS-EKV-SRTP. I doubt it
will map DTLS key updates to re-invites. Most probably it will simply
re-encode.
>
> ______________
> Roman Shpount
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<p><br>
On Apr 4, 2012 10:24 AM, &quot;Roman Shpount&quot; &lt;<a href=3D"mailto:ro=
man@telurix.com">roman@telurix.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On Wed, Apr 4, 2012 at 12:43 PM, I=F1aki Baz Castillo &lt;<a href=3D"m=
ailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Hi, nobody cares about the implications of option 2 ???<br>
&gt;&gt;<br>
&gt;&gt; Do all the people planning to interop with SIP assume that they&#3=
9;ll<br>
&gt;&gt; need the super B2BUA in the second image (without the possibility =
of<br>
&gt;&gt; using a pure SIP proxy)?:<br>
&gt;&gt;<br>
&gt;&gt; =A0<a href=3D"http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DT=
LS-EKT-SRTP.png">http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT=
-SRTP.png</a><br>
&gt;&gt;<br>
&gt;<br>
&gt; &lt;sarcasm&gt;I guess we should look at this as a business opportunit=
y ;) I am not sure why you assume that building such gateway would take 10 =
years. I can sell a gateway like this to anybody who needs it right now.&lt=
;/sarcasm&gt;<br>

&gt;<br>
&gt; My assumption is that IP phones will migrate to WebRTC effectively eli=
minating SIP in end user devices.<br></p>
<p>That is your academic assumption, why does IT department need to spend e=
xtra money to either desert or upgrade its existing SIP devices without sub=
stantial gain? After all, tons of users are still using window xp.</p>

<p>Backward capability is a key to the success of new technology.</p>
<p>-Jesse</p>
<p> The only place where SIP will remain would be federation, which commonl=
y uses some sort of SBC anyway. This SBC will need to be extended to suppor=
t ICE anyway. Might as well through in support for DTLS-EKV-SRTP. I doubt i=
t will map DTLS key updates to re-invites. Most probably it will simply re-=
encode.<br>

&gt; =A0<br>
&gt; ______________<br>
&gt; Roman Shpount<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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On Wed, Apr 4, 2012 at 8:42 PM, jesse <chat2jesse@gmail.com> wrote:

>
> On Apr 4, 2012 10:24 AM, "Roman Shpount" <roman@telurix.com> wrote:
> >
> > My assumption is that IP phones will migrate to WebRTC effectively
> eliminating SIP in end user devices.
>
> That is your academic assumption, why does IT department need to spend
> extra money to either desert or upgrade its existing SIP devices without
> substantial gain? After all, tons of users are still using window xp.
>
> Backward capability is a key to the success of new technology.
>
Please note the <sarcasm> tags.

On the more serious note, very few SIP end points offer working ICE
support. So, in a large sense, interop with them is not an option. Out of
the ones that do support ICE and SRTP, very few are actually connected
directly to a public internet. Most of them are connected to some sort of
PBX or an IP PBX type service. So, in reality you do not need to bridge
every IP phone with WebRTC. If a few PBX and hosted centrex vendors will
add support for WebRTC required features, we will get compatibility with
existing end points. To support the rest you will need to deploy some sort
of gateway.

In my personal opinion, tons of users use XP due to the fact that never MS
OS versions do not offer a significant reason to upgrade, while introducing
enough usability and operational issues to create a significant upgrade
hurdle (I am trying to be politically correct here). WebRTC enabled end
points, on the other hand, will offer significant benefits to traditional
SIP phones, since they will allow development of higher quality integrated
real time communications services. I hope this will drive a much quicker
standard adoption.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Apr 4, 2012 at 8:42 PM=
, jesse <span dir=3D"ltr">&lt;<a href=3D"mailto:chat2jesse@gmail.com">chat2=
jesse@gmail.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 class=3D"im"><p><br>
On Apr 4, 2012 10:24 AM, &quot;Roman Shpount&quot; &lt;<a href=3D"mailto:ro=
man@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:<br>
&gt;<br>
&gt; My assumption is that IP phones will migrate to WebRTC effectively eli=
minating SIP in end user devices.<br></p>
</div><p>That is your academic assumption, why does IT department need to s=
pend extra money to either desert or upgrade its existing SIP devices witho=
ut substantial gain? After all, tons of users are still using window xp.</p=
>


<p>Backward capability is a key to the success of new technology.</p></bloc=
kquote></div>Please note the &lt;sarcasm&gt; tags. <br><br>On the more seri=
ous note, very few SIP end points offer working ICE support. So, in a large=
 sense, interop with them is not an option. Out of the ones that do support=
 ICE and SRTP, very few are actually connected directly to a public interne=
t. Most of them are connected to some sort of PBX or an IP PBX type service=
. So, in reality you do not need to bridge every IP phone with WebRTC. If a=
 few PBX and hosted centrex vendors will add support for WebRTC required fe=
atures, we will get compatibility with existing end points. To support the =
rest you will need to deploy some sort of gateway.<br>
<br>In my personal opinion, tons of users use XP due to the fact that never=
 MS OS versions do not offer a significant reason to upgrade, while introdu=
cing enough usability and operational issues to create a significant upgrad=
e hurdle (I am trying to be politically correct here). WebRTC enabled end p=
oints, on the other hand, will offer significant benefits to traditional SI=
P phones, since they will allow development of higher quality integrated re=
al time communications services. I hope this will drive a much quicker stan=
dard adoption.<br>
_____________<br>Roman Shpount<br>
<br>

--047d7b339ccffa53d604bce51752--

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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Basil,

> What you are asking for is something no one in the media industry offers
> for something comparable.

I know, but the reluctance to do so means that Google knows that there *may
be* IPR on VP8.  This is just the facts of life.  I believe it's extremely
important that people know and understand that nobody knows the IPR
situation with VP8.  If they did, they would offer indemnity, because there
is nothing to worry about, but nobody can.

> As was pointed out elsewhere on this list, even
> the H.264 standard has been updated to include additional patents over
> time deemed to be "essential".  One could be non-infringing in the past
> and suddenly find yourself infringing.

Indeed, but I have a significantly higher level of confidence that I can
identify all of the legitimate companies with IPR on H.264, whereas I
haven't a clue where to start (outside of Google) for VP8. 
 
> I don't know what all of Google's public statements about VP8 IPR status
> are, but we know that the patents they own are granted via the license,
> and we know that a public call for the formation of a patent pool over a
> year ago has yielded not even a follow-up announcement from MPEG-LA.
> Furthermore, not a single patent has been brought forward publicly as
> being even remotely doubtful as to its application to VP8.

If I owned IPR on VP8 (which I don't personally; can't speak for my
employer), I certainly would not tell you and I would not join a patent
pool, either.  I would wait until you adopt VP8, build it into software and
hardware products, have it massively deployed, and then I'd come along and
collect my royalties.  There is absolutely no financial incentive for an IPR
holder to join a patent pool.

H.264, on the other hand, is different.  H.264 was developed jointly by
video coding experts in ISO and ITU.  As an part of that participation, they
are requested to disclose IPR.  And, respectable companies will disclose
their IPR.   Every company participating in that work has a vested interest
in the success of H.264.  Some are looking for royalties and some are
looking for the technology to enable their products.  Thus, the formation of
the patent pool is in the best interest of all involved so that people will
adopt the technology.  While there are those not in the pool, they have
submitted IPR statements to the ITU.
 
> So, until there is some actual evidence that there is an IPR threat, then
> with all due respect, statements such as yours must be classified as
> spreading FUD, because they serve to drive adoption away from a free,
> open, unencumbered standard to one that is on baseless claims.

My statement is intended to spread FUD, but not maliciously; I'm trying to
be helpful.  You SHOULD be fearful of not only patent trolls, but also
legitimate companies who are not obligated to disclose their IPR and who
have a vested interest in H.264 and who will take full advantage of their
position should VP8 be widely deployed.  You SHOULD feel some level of
uncertainty about the IPR situation surrounding VP8.  And, you SHOULD doubt
any claim made that there is no IPR on VP8.

While there may be undisclosed IPR on H.264, I find it hard to believe that
there would be much, if any, at this point.  Keep in mind that H.264 is
product of the joint effort of a bunch of people who are experts in the
field.  I would be suspicious of any company not involved in the development
of H.264 claiming to have IPR, because somebody in the joint committee
probably owns that IPR.  Sure, there might be something ... something very
minor.  Perhaps.

And what if there is something that comes up?  Every company participating
in the work on H.264 would have a vested interest in disproving the claims,
particularly if whomever makes claims refuses to license or tries to demand
insane royalties.  I could imagine a stream of expert witnesses coming into
a courtroom to defend H.264.  With no vested interest in VP8, who cares if
you get sued for 50% of product revenues?

Understand, all of this commentary has nothing to do with the technical
merits of VP8 vs. H.264.  There are risks with selecting either technology.
That said, I personally have a much higher degree of confidence that I will
not get sued out of business if I were to select H.264, contact all of the
IPR holders registered in the ITU and ISO patent databases, and negotiate a
license directly (or via MPEG-LA when possible).  I have no degree of
confidence whatsoever with VP8, except that Google was kind enough to make
its IPR available at no cost.  I just don't know who else is out there, and
it might be 10 or 15 H.264 patent holders, all with an interest to exercise
their rights when the time comes.
 
Paul



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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was	Re:Proposal for H.263 baseline codec]
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Harald,

> On 04/04/2012 07:23 AM, Paul E. Jones wrote:
> > I would have a high degree of confidence that I could identify all
> > H.264 IPR holders.
> Question .... how would you do that?
> I'm genuinely curious. The subject has come up in other discussions,
> including in ISO in connection with the WebVC and IVC projects.
> 
> The database in question is an ISO database, not an ITU database, and the
> format is most un-database-like; it is here:
> 
> http://isotc.iso.org/livelink/livelink/fetch/2000/2122/3770791/JTC1_Patent
> s_database.html?nodeid=3777806&vernum=-2
> <http://isotc.iso.org/livelink/livelink/fetch/2000/2122/3770791/JTC1_Paten
> ts_database.html?nodeid=3777806&vernum=-2>

There are two databases.  One with ISO (and you have the link) and the other
is ITU:
http://www.itu.int/ipr/IPRSearch.aspx?iprtype=PS

Just enter "H.264" into the box under "Recommendation No" and hit ENTER.
There are 327 filed IPR claims.

Of course, this is not all of the IPR that may exist, but my point is that
it's highly unlikely that more exists from companies not listed given the
ITU and ISO's patent policies, participating companies' mutual desire to see
the technology succeed, etc.  Virtually every concept you can imagine that
found its way into H.264 is either prior art or new IPR introduced by
participating companies, all of which SHOULD be listed.  If a company
participated and failed to disclose IPR... that's really bad.  Not only
would the defendant be unhappy, but said IPR holder might very well be
blocked from further participation.  Something would be done.  And, they
might have a tough time arguing in court, knowing they were withholding
information and, thus, misleading the defendant who could demonstrate an
effort was made to negotiate with all IPR holders.  If I were a judge, I
would not take kindly to such subversive, destructive behavior in an
standards organization.  That leaves only patent trolls who did not
participate in the H.264 work.  There is little we can do with such entities
who, quite likely, never invented anything in H.264, but have a patent
worded just well enough to give the illusion that they invented something in
the standard.  This is an unfortunate reality in our world with all
technology.

Now, as I look at the 327 patents filed on H.264 (in the ITU database
alone), I'm left wondering... nobody in the whole world has even one on VP8
outside of On2 / Google?  I'd bet there are.  I personally would not want to
take the risk, but that's me.

Paul



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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 11:04:17 +0200
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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2012/4/5 Roman Shpount <roman@telurix.com>:
> On the more serious note, very few SIP end points offer working ICE suppo=
rt.
> So, in a large sense, interop with them is not an option.

For me there is a BIG difference when a kind of B2BUA is required
(which involves signaling "transaction"). That's the barrier IMHO.

ICE support can be implemented in a ICE-Lite RTP/SRTP proxy, see:

  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.png

The problem arises when media encrypt/decrypt is required, and evenr
more when a key update in RTP (like the DTLS EKT update) must be
converted into a signaling re-INVITE by a super Signaling+Media B2BUA:

  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png



> Out of the ones
> that do support ICE and SRTP, very few are actually connected directly to=
 a
> public internet. Most of them are connected to some sort of PBX or an IP =
PBX
> type service. So, in reality you do not need to bridge every IP phone wit=
h
> WebRTC.

That is a *very* limited scope of what WebRTC can provide. An IT
department should be able to deploy its own WebRTC infrastructure (a
Web+WebSocket server) within its "local" network, so browsers
accessing to such a local website share the network with SIP/XMPP
phones/devices/softphones.

Please don't imagine WebRTC and SIP interop as the communication
between two islands ;)



> If a few PBX and hosted centrex vendors will add support for WebRTC
> required features, we will get compatibility with existing end points.

Pure SIP proxies do exist. A PBX is not always needed, so again, SIP
world does not require to be "an island".


> To support the rest you will need to deploy some sort of gateway.

Not in the case SDES+SRTP is allowed in WebRTC (see Fabius's recent
mails about SDES and DTLS).



> hurdle (I am trying to be politically correct here). WebRTC enabled end
> points, on the other hand, will offer significant benefits to traditional
> SIP phones, since they will allow development of higher quality integrate=
d
> real time communications services. I hope this will drive a much quicker
> standard adoption.

The problem is that there is a *NEW* SRTP related spec for WebRTC:

  http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-03

Maybe it could also become a standard for SIP? I hope, but currently
it's not, so by *mandating* DTLS-EKT-SRTP in WebRTC we are creating a
island.


Regards.


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

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Roman Shpount'" <roman@telurix.com>, "'jesse'" <chat2jesse@gmail.com>
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Hi,

I think that your point of view about IP PBX that need to support RTCweb is
interesting but the problem is that RTCweb is not the application, it is an
enabler, so you may see multiple applications servers that will offer
communication services (like Skype, MS Messenger, Google), so I am wondering
if the IP PBX will need to peer with all of them or will the application
servers peer with the IP PBXs using SIP as was proposed in RTCweb. 

Roni Even

 

 

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Roman Shpount
Sent: Thursday, April 05, 2012 5:14 AM
To: jesse
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need

 




On Wed, Apr 4, 2012 at 8:42 PM, jesse <chat2jesse@gmail.com> wrote:


On Apr 4, 2012 10:24 AM, "Roman Shpount" <roman@telurix.com> wrote:
>
> My assumption is that IP phones will migrate to WebRTC effectively
eliminating SIP in end user devices.

That is your academic assumption, why does IT department need to spend extra
money to either desert or upgrade its existing SIP devices without
substantial gain? After all, tons of users are still using window xp.

Backward capability is a key to the success of new technology.

Please note the <sarcasm> tags. 

On the more serious note, very few SIP end points offer working ICE support.
So, in a large sense, interop with them is not an option. Out of the ones
that do support ICE and SRTP, very few are actually connected directly to a
public internet. Most of them are connected to some sort of PBX or an IP PBX
type service. So, in reality you do not need to bridge every IP phone with
WebRTC. If a few PBX and hosted centrex vendors will add support for WebRTC
required features, we will get compatibility with existing end points. To
support the rest you will need to deploy some sort of gateway.

In my personal opinion, tons of users use XP due to the fact that never MS
OS versions do not offer a significant reason to upgrade, while introducing
enough usability and operational issues to create a significant upgrade
hurdle (I am trying to be politically correct here). WebRTC enabled end
points, on the other hand, will offer significant benefits to traditional
SIP phones, since they will allow development of higher quality integrated
real time communications services. I hope this will drive a much quicker
standard adoption.
_____________
Roman Shpount


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application, it is an enabler, so you may see multiple applications =
servers that will offer communication services (like Skype, MS =
Messenger, Google), so I am wondering if the IP PBX will need to peer =
with all of them or will the application servers peer with the IP PBXs =
using SIP as was proposed in RTCweb. <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 Even<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><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-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Roman Shpount<br><b>Sent:</b> Thursday, April 05, 2012 5:14 =
AM<br><b>To:</b> jesse<br><b>Cc:</b> rtcweb@ietf.org<br><b>Subject:</b> =
Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a =
need<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><br =
clear=3Dall><o:p></o:p></p><div><p class=3DMsoNormal>On Wed, Apr 4, 2012 =
at 8:42 PM, jesse &lt;<a =
href=3D"mailto:chat2jesse@gmail.com">chat2jesse@gmail.com</a>&gt; =
wrote:<o:p></o:p></p><div><p><br>On Apr 4, 2012 10:24 AM, &quot;Roman =
Shpount&quot; &lt;<a href=3D"mailto:roman@telurix.com" =
target=3D"_blank">roman@telurix.com</a>&gt; wrote:<br>&gt;<br>&gt; My =
assumption is that IP phones will migrate to WebRTC effectively =
eliminating SIP in end user devices.<o:p></o:p></p></div><p>That is your =
academic assumption, why does IT department need to spend extra money to =
either desert or upgrade its existing SIP devices without substantial =
gain? After all, tons of users are still using window =
xp.<o:p></o:p></p><p>Backward capability is a key to the success of new =
technology.<o:p></o:p></p></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>Please note the &lt;sarcasm&gt; tags. =
<br><br>On the more serious note, very few SIP end points offer working =
ICE support. So, in a large sense, interop with them is not an option. =
Out of the ones that do support ICE and SRTP, very few are actually =
connected directly to a public internet. Most of them are connected to =
some sort of PBX or an IP PBX type service. So, in reality you do not =
need to bridge every IP phone with WebRTC. If a few PBX and hosted =
centrex vendors will add support for WebRTC required features, we will =
get compatibility with existing end points. To support the rest you will =
need to deploy some sort of gateway.<br><br>In my personal opinion, tons =
of users use XP due to the fact that never MS OS versions do not offer a =
significant reason to upgrade, while introducing enough usability and =
operational issues to create a significant upgrade hurdle (I am trying =
to be politically correct here). WebRTC enabled end points, on the other =
hand, will offer significant benefits to traditional SIP phones, since =
they will allow development of higher quality integrated real time =
communications services. I hope this will drive a much quicker standard =
adoption.<br>_____________<br>Roman =
Shpount<o:p></o:p></p></div></div></body></html>
------=_NextPart_000_00D8_01CD1325.01F65150--


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Date: Thu, 5 Apr 2012 11:18:40 +0200
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2012/4/5 Roni Even <ron.even.tlv@gmail.com>:
> so I am wondering if the IP PBX will need to peer with all of them or wil=
l
> the application servers peer with the IP PBXs using SIP as was proposed i=
n
> RTCweb.

You don't need to make a web browser to talk SIP (UDP/TCP/TLS) in
order to interop with a SIP network.

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

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 12:02:53 +0200
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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2012/4/5 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> IMHO, there is no need to tweak WebRTC recommendation for the sake of SIP=
 proxy. I'm fine as long as there is a way in WebRTC to interop with SIP. F=
or example, SIP proxy is not suitable for IETF SIPREC recording implementat=
ion itself!!!

For the cases in which you need recording, ok, go ahead with a SIP
SBC/B2BUA/server or whatever you need, but please let using a pure SIP
proxy for those not interested in media recording ;)


> But I'm also surprised to see that there is no response for Fabio Pietros=
anti mail on DTLS-SRTP trust model mail thread.

Me too. I hope Fabio's nice explanations will be considered by the WG
rather than continuing with "DTLS (which has been NEVER tested with
SRTP) is the only magical solution for the multimedia communications".

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

From lists@infosecurity.ch  Thu Apr  5 03:16:41 2012
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Subject: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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Hi,

i've been discussing with several friends about the current discussion
on the security standard in this mailing lists, in particular regarding
the topic of DTLS-SRTP trust model posted there
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04007.html .

I found that there is no common definition for the concept of
"end-to-end encryption" and there are a lot of misunderstanding about it.

In particular, the fact that two peer can community by exchanging keys
directly among them, tend to typically to be defined as end-to-end
encryption.

However the term "end-to-end encryption" have also a general
misconception that's something that "no one can intercept" .

Unfortunately this is not true for DTLS-SRTP, while for example it's
true for OpenPGP/MIME or for ZRTP with SAS.

We need to separate the 4 concepts:

- end-to-end encryption
Ability for a key management system to exchange keys directly without
relying on intermediate server actively involved in encryption process.

- end-to-end authentication
Ability for end-user to authenticate the end-to-end encryption process
without the need to rely on a single or multiple set of trusted third
parties

- end-to-site encryption
Ability for a key management system to exchange keys with/trough the
server with which data are exchanged, involving it in the authentication
process.

- end-to-site authentication
Ability for end-user to authenticate the end-to-site encryption process
based on the same security mechanism used to establish trust with the
server with which data are exchanged.


What i would like to focus is that:

- DTLS does provide end-to-end encryption (in specific context of use)
- DTLS does provide end-to-site authentication (rely on third party trust)

- ZRTP does provide end-to-end encryption (in all context of use)
- ZRTP does provide end-to-end authentication (does not rely on third
party trust)

- SDES-SRTP does provide end-to-site encryption (encryption with/trough
your server)
- SDES-SRTP does provide end-to-site authentication (you trust your
server involved in key exchange)

So when we tend to think about the "how much security a technology
provide" we should probably compare in a scale:

- ZRTP
  - end-to-end encryption
  - end-to-end authentication
- DTLS-SRTP
  - end-to-end encryption
  - end-to-site authentication
- SDES-SRTP
  - end-to-site encryption
  - end-to-site authentication

So currently we can affirm that:

- ZRTP does not rely on third party trust for effective security
- DTLS-SRTP rely on third party trust for effective security
- SDES-SRTP rely on third party trust for effective security

This is the *MOST IMPORTANT* distinction for an encryption technology:
			WHO SHOULD I TRUST?

Well, basically it seems to me that DTLS-SRTP and SDES-SRTP both require

So, my point are to:

- Introduce SDES-SRTP for compatibility and simplicity
  - Specify that the Browser will need to provide indicate the security
level to the user (like the lock of HTTPS, same security model)

- Introduce end-to-end authentication support for DTLS-SRTP via SAS
  - Specify that the browser will need to to provide the end-user way to
use end-to-end authentication and indicate the security level to the user.

Then it will be up to the signaling server and/or to the browser
settings to specificy the required security model:
- end-to-end encryption + end-to-end authentication
or
- end-to-site encryption + end-to-site authentication

But please don't mix those two as it will be *Very difficult, near to
impossible* to explain to the user "WHO SHOULD HE TRUST" .

Fabio

From christer.holmberg@ericsson.com  Thu Apr  5 04:14:21 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, Roman Shpount <roman@telurix.com>
Date: Thu, 5 Apr 2012 13:14:17 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 13:20:05 +0200
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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2012/4/5 Christer Holmberg <christer.holmberg@ericsson.com>:
>> The problem arises when media encrypt/decrypt is required, and evenr mor=
e when a key update in RTP (like the DTLS EKT update) must be converted int=
o a signaling re-INVITE by a super Signaling+Media B2BUA:
>
> ...and, in general we should not specify procedures which require an inte=
rmediary to trigger and send re-INVITEs in the first place, because that it=
self can then cause lots of issues.

Agreed. So if WebRTC-SIP interop is important (or a valuable point for
WebRTC) then please DON'T rely on the existence of such a monstrous
super signaling+media B2BUA, and DON'T assume that WebRTC-SIP interop
is the communication between two separate islands. Please. It should
be much better, and allowing SDES-SRTP is the key for that.

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

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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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On Thu, Apr 5, 2012 at 5:04 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2012/4/5 Roman Shpount <roman@telurix.com>:
> > On the more serious note, very few SIP end points offer working ICE
> support.
> > So, in a large sense, interop with them is not an option.
>
> For me there is a BIG difference when a kind of B2BUA is required
> (which involves signaling "transaction"). That's the barrier IMHO.
>
> ICE support can be implemented in a ICE-Lite RTP/SRTP proxy, see:
>
>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.png


This picture is not entirely correct -- ICE Lite proxy should be on both
SIP and Media path.

The problem arises when media encrypt/decrypt is required, and evenr
> more when a key update in RTP (like the DTLS EKT update) must be
> converted into a signaling re-INVITE by a super Signaling+Media B2BUA:
>
>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png
>
>
Not sure why key update needs to be converted into signaling if media is
re-encrypted. Key updates on both sides of the proxy are independent. Such
proxy can also change SDES-SRTP key on each re-INVITE initiated by either
side, but apart from doing actual encryption/decryption, operation wise it
should not be any different then ICE-Lite proxy.


> > Out of the ones
> > that do support ICE and SRTP, very few are actually connected directly
> to a
> > public internet. Most of them are connected to some sort of PBX or an I=
P
> PBX
> > type service. So, in reality you do not need to bridge every IP phone
> with
> > WebRTC.
>
> That is a *very* limited scope of what WebRTC can provide. An IT
> department should be able to deploy its own WebRTC infrastructure (a
> Web+WebSocket server) within its "local" network, so browsers
> accessing to such a local website share the network with SIP/XMPP
> phones/devices/softphones.
>
> Please don't imagine WebRTC and SIP interop as the communication
> between two islands ;)
>

I actually do imagine WebRTC/SIP interop as two islands. If we need to
proxy media and convert Web+WebSocket into SIP this is a definition of
island bridge for me.

 > If a few PBX and hosted centrex vendors will add support for WebRTC
> > required features, we will get compatibility with existing end points.
>
> Pure SIP proxies do exist. A PBX is not always needed, so again, SIP
> world does not require to be "an island".
>

Once  again, if SIP proxy is combined with media proxy, you can bridge SIP
and WebRTC, even if DTLS-SRTP is used.


>  > To support the rest you will need to deploy some sort of gateway.
>
> Not in the case SDES+SRTP is allowed in WebRTC (see Fabius's recent
> mails about SDES and DTLS).
>
>
Once again, we still need a gateway to support ICE. If end point is adding
ICE to interop with WebRTC, why not add DTLS-SRTP at the same time.


> > hurdle (I am trying to be politically correct here). WebRTC enabled end
> > points, on the other hand, will offer significant benefits to tradition=
al
> > SIP phones, since they will allow development of higher quality
> integrated
> > real time communications services. I hope this will drive a much quicke=
r
> > standard adoption.
>
> The problem is that there is a *NEW* SRTP related spec for WebRTC:
>
>  http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-03
>
> Maybe it could also become a standard for SIP? I hope, but currently
> it's not, so by *mandating* DTLS-EKT-SRTP in WebRTC we are creating a
> island.
>
>
Same as above.

P.S. I actually do not like DTLS-SRTP, since it does not have a large
installed base, comes with too many features, slows down call setup, will
requirecomplex interop and such. At the same time I do not think that
interop with legacy SDES-SRTP is a reason not to use it. I believe we
should design a new key exchange method, that can be mapped to SDES-SRTP
via signaling alone, addresses SDES-SRTP issues (transmitting keys over
clear channel in signaling, replay of signaling) without introducing all
the problems related to DTLS-SRTP. Any number of methods using public key
based exchange over signaling and ICE should do the trick.
_____________
Roman Shpount

--047d7b33da10e30f4504bceedcbc
Content-Type: text/html; charset=ISO-8859-1
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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 5:04 AM=
, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.ne=
t" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">

2012/4/5 Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D"_=
blank">roman@telurix.com</a>&gt;:<br>
<div>&gt; On the more serious note, very few SIP end points offer working I=
CE support.<br>
&gt; So, in a large sense, interop with them is not an option.<br>
<br>
</div>For me there is a BIG difference when a kind of B2BUA is required<br>
(which involves signaling &quot;transaction&quot;). That&#39;s the barrier =
IMHO.<br>
<br>
ICE support can be implemented in a ICE-Lite RTP/SRTP proxy, see:<br>
<br>
 =A0<a href=3D"http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.=
png" target=3D"_blank">http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SD=
ES-SRTP.png</a></blockquote><div><br>This picture is not entirely correct -=
- ICE Lite proxy should be on both SIP and Media path. <br>

<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">

The problem arises when media encrypt/decrypt is required, and evenr<br>
more when a key update in RTP (like the DTLS EKT update) must be<br>
converted into a signaling re-INVITE by a super Signaling+Media B2BUA:<br>
<br>
 =A0<a href=3D"http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-S=
RTP.png" target=3D"_blank">http://public.aliax.net/WebRTC/WebRTC_SIP_Intero=
p_DTLS-EKT-SRTP.png</a><br>
<div><br></div></blockquote><div>=A0</div><div>Not sure why key update need=
s to be converted into signaling if media is re-encrypted. Key updates on b=
oth sides of the proxy are independent. Such proxy can also change SDES-SRT=
P key on each re-INVITE initiated by either side, but apart from doing actu=
al encryption/decryption, operation wise it should not be any different the=
n ICE-Lite proxy.<br>

=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div>
&gt; Out of the ones<br>
&gt; that do support ICE and SRTP, very few are actually connected directly=
 to a<br>
&gt; public internet. Most of them are connected to some sort of PBX or an =
IP PBX<br>
&gt; type service. So, in reality you do not need to bridge every IP phone =
with<br>
&gt; WebRTC.<br>
<br>
</div>That is a *very* limited scope of what WebRTC can provide. An IT<br>
department should be able to deploy its own WebRTC infrastructure (a<br>
Web+WebSocket server) within its &quot;local&quot; network, so browsers<br>
accessing to such a local website share the network with SIP/XMPP<br>
phones/devices/softphones.<br>
<br>
Please don&#39;t imagine WebRTC and SIP interop as the communication<br>
between two islands ;)<br></blockquote><div><br>I actually do imagine WebRT=
C/SIP interop as two islands. If we need to proxy media and convert Web+Web=
Socket into SIP this is a definition of island bridge for me. <br><br>

</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8ex;b=
order-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
&gt; If a few PBX and hosted centrex vendors will add support for WebRTC<br=
>
&gt; required features, we will get compatibility with existing end points.=
<br>
<br>
</div>Pure SIP proxies do exist. A PBX is not always needed, so again, SIP<=
br>
world does not require to be &quot;an island&quot;.<br>
<div></div></blockquote><div><br>Once=A0 again, if SIP proxy is combined wi=
th media proxy, you can bridge SIP and WebRTC, even if DTLS-SRTP is used.<b=
r></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt=
 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">

<div>

&gt; To support the rest you will need to deploy some sort of gateway.<br>
<br>
</div>Not in the case SDES+SRTP is allowed in WebRTC (see Fabius&#39;s rece=
nt<br>
mails about SDES and DTLS).<br><div><br></div></blockquote><div>=A0</div><d=
iv>Once again, we still need a gateway to support ICE. If end point is addi=
ng ICE to interop with WebRTC, why not add DTLS-SRTP at the same time.<br>

=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div>
&gt; hurdle (I am trying to be politically correct here). WebRTC enabled en=
d<br>
&gt; points, on the other hand, will offer significant benefits to traditio=
nal<br>
&gt; SIP phones, since they will allow development of higher quality integr=
ated<br>
&gt; real time communications services. I hope this will drive a much quick=
er<br>
&gt; standard adoption.<br>
<br>
</div>The problem is that there is a *NEW* SRTP related spec for WebRTC:<br=
>
<br>
 =A0<a href=3D"http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-03" targe=
t=3D"_blank">http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-03</a><br>
<br>
Maybe it could also become a standard for SIP? I hope, but currently<br>
it&#39;s not, so by *mandating* DTLS-EKT-SRTP in WebRTC we are creating a<b=
r>
island.<br>
<div><div></div><div><br></div></div></blockquote><div><br>Same as above.<b=
r><br>P.S. I actually do not like DTLS-SRTP, since it does not have a large=
 installed base, comes with too many features, slows down call setup, will =
requirecomplex interop and such. At the same time I do not think that inter=
op with legacy SDES-SRTP is a reason not to use it. I believe we should des=
ign a new key exchange method, that can be mapped to SDES-SRTP via signalin=
g alone, addresses SDES-SRTP issues (transmitting keys over clear channel i=
n signaling, replay of signaling) without introducing all the problems rela=
ted to DTLS-SRTP. Any number of methods using public key based exchange ove=
r signaling and ICE should do the trick.<br>

</div></div>_____________<br>Roman Shpount<br>
<br>

--047d7b33da10e30f4504bceedcbc--

From ibc@aliax.net  Thu Apr  5 07:40:01 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 16:39:39 +0200
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To: Roman Shpount <roman@telurix.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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2012/4/5 Roman Shpount <roman@telurix.com>:
>
> On Thu, Apr 5, 2012 at 5:04 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> w=
rote:
>> ICE support can be implemented in a ICE-Lite RTP/SRTP proxy, see:
>>
>> =C2=A0http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.png
>
>
> This picture is not entirely correct -- ICE Lite proxy should be on both =
SIP
> and Media path.

Incorrect. The proxy would communicate with the ICE-Lite server(s)
using some kind of proprietary protocol. This is already true in some
SIP proxies (SER, Kamailio, OpenSer) and media-proxies (rtpproxy,
mediaproxy, irtpproxy).


>> The problem arises when media encrypt/decrypt is required, and evenr
>> more when a key update in RTP (like the DTLS EKT update) must be
>> converted into a signaling re-INVITE by a super Signaling+Media B2BUA:
>>
>> =C2=A0http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.pn=
g
>>
>
> Not sure why key update needs to be converted into signaling if media is
> re-encrypted. Key updates on both sides of the proxy are independent. Suc=
h
> proxy can also change SDES-SRTP key on each re-INVITE initiated by either
> side, but apart from doing actual encryption/decryption, operation wise i=
t
> should not be any different then ICE-Lite proxy.

Maybe.




>> That is a *very* limited scope of what WebRTC can provide. An IT
>> department should be able to deploy its own WebRTC infrastructure (a
>> Web+WebSocket server) within its "local" network, so browsers
>> accessing to such a local website share the network with SIP/XMPP
>> phones/devices/softphones.
>>
>> Please don't imagine WebRTC and SIP interop as the communication
>> between two islands ;)
>
>
> I actually do imagine WebRTC/SIP interop as two islands. If we need to pr=
oxy
> media and convert Web+WebSocket into SIP this is a definition of island
> bridge for me.

If you say that we need to "convert Web+WebSocket into SIP" then you
are absolutely wrong, even more taking into account that
"draft-ibc-sipcore-sip-websocket-01" (The WebSocket Protocol as a
Transport for SIP) is being adopted by the IETF SIPCORE Working Group
as a new SIP transport.

The slide 8 in [2] shows a *pure* SIP proxy, and not a "WebSocket to
SIP gateway". So ***NO*** island at signaling level, not at all.


[1] http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-01
[2] http://sip-on-the-web.aliax.net/






>> Not in the case SDES+SRTP is allowed in WebRTC (see Fabius's recent
>> mails about SDES and DTLS).
>>
>
> Once again, we still need a gateway to support ICE. If end point is addin=
g
> ICE to interop with WebRTC, why not add DTLS-SRTP at the same time.

Building an ICE-Lite RTP/SRTP tunnel/proxy is "easy". After ICE
"handhake" the server can relay UDP packets as kernel space (conntrack
in Linux), but if you need to decrypt and encrypt you need to do that
at user space, which is much worse and requires much more CPU.



> P.S. I actually do not like DTLS-SRTP, since it does not have a large
> installed base, comes with too many features, slows down call setup, will
> requirecomplex interop and such.

> I believe we should
> design a new key exchange method, that can be mapped to SDES-SRTP via
> signaling alone,

And wouldn't that be a mechanism with "non a large installed base"?


> addresses SDES-SRTP issues (transmitting keys over clear
> channel in signaling, replay of signaling)

If HTTPS / WSS is used, there is no "clear channel". It would be easy
to mandate *all* the browser communication using TLS for the case in
which SDES-SRTP is used. That is a feasible and valid requirement
IMHO.



> without introducing all the
> problems related to DTLS-SRTP. Any number of methods using public key bas=
ed
> exchange over signaling and ICE should do the trick.

So a complete new protocol. Bye any kind of interop (except if you
have the magic super B2BUA).



PS: Please consider reading Fabios's mails and reply to them. The
discussion will be more interesting is such nice explanations are not
ignored ;)


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

From jmvalin@mozilla.com  Thu Apr  5 08:10:22 2012
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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On 05/04/12 02:47 AM, Paul E. Jones wrote:
> I know, but the reluctance to do so means that Google knows that there *may
> be* IPR on VP8.  This is just the facts of life.  I believe it's extremely
> important that people know and understand that nobody knows the IPR
> situation with VP8.  If they did, they would offer indemnity, because there
> is nothing to worry about, but nobody can.

Personally, I would be reluctant to indemnify anyone over IPR in a paper
clip. This is the sad state of the patent system and it applies to every
piece of technology.

> Indeed, but I have a significantly higher level of confidence that I can
> identify all of the legitimate companies with IPR on H.264, whereas I
> haven't a clue where to start (outside of Google) for VP8. 

Obviously, that makes H.264 so much safer, right?
http://www.theregister.co.uk/2012/03/29/microsoft_motorola_patent_cases/

> If I owned IPR on VP8 (which I don't personally; can't speak for my
> employer), I certainly would not tell you and I would not join a patent
> pool, either.  I would wait until you adopt VP8, build it into software and
> hardware products, have it massively deployed, and then I'd come along and
> collect my royalties.  There is absolutely no financial incentive for an IPR
> holder to join a patent pool.

I assume this is why we've seen all these lawsuits against Google,
Microsoft, Cisco, Apple... over their use of Vorbis and Speex, right?
Seriously, I can count at least 10 free AV codecs and none of them have
had any patent lawsuits that I'm aware of.

> H.264, on the other hand, is different.  H.264 was developed jointly by
> video coding experts in ISO and ITU.  As an part of that participation, they
> are requested to disclose IPR.  And, respectable companies will disclose
> their IPR.   Every company participating in that work has a vested interest
> in the success of H.264.  Some are looking for royalties and some are
> looking for the technology to enable their products.  Thus, the formation of
> the patent pool is in the best interest of all involved so that people will
> adopt the technology.  While there are those not in the pool, they have
> submitted IPR statements to the ITU.

The best interest of a H.264 patent owner is to avoid having to
disclose, wait for adoption, and then be free to ask for a lot more than
what they would get from the patent pool. This is why I consider H.264
to be a higher risk than VP8. The current Motorola lawsuit confirms this.

Cheers,

	Jean-Marc

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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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--e89a8ff1c650fe3ffa04bcf05b1b
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On Thu, Apr 5, 2012 at 10:39 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2012/4/5 Roman Shpount <roman@telurix.com>:
> >
> > On Thu, Apr 5, 2012 at 5:04 AM, I=F1aki Baz Castillo <ibc@aliax.net>
> wrote:
> >> ICE support can be implemented in a ICE-Lite RTP/SRTP proxy, see:
> >>
> >>  http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.png
> >
> >
> > This picture is not entirely correct -- ICE Lite proxy should be on bot=
h
> SIP
> > and Media path.
>
> Incorrect. The proxy would communicate with the ICE-Lite server(s)
> using some kind of proprietary protocol. This is already true in some
> SIP proxies (SER, Kamailio, OpenSer) and media-proxies (rtpproxy,
> mediaproxy, irtpproxy).
>
> Internal protocols do not matter. The "super B2BUA" can be using a custom
protocol between RTP forwarder and SIP processor. Does not change the fact
that from logical point of view it is on both SIP and media path. At
minimum you are missing a custom protocol connection between ICE-Lite
gateway and SIP proxy. So, your picture is still wrong.


> >> That is a *very* limited scope of what WebRTC can provide. An IT
> >> department should be able to deploy its own WebRTC infrastructure (a
> >> Web+WebSocket server) within its "local" network, so browsers
> >> accessing to such a local website share the network with SIP/XMPP
> >> phones/devices/softphones.
> >>
> >> Please don't imagine WebRTC and SIP interop as the communication
> >> between two islands ;)
> >
> >
> > I actually do imagine WebRTC/SIP interop as two islands. If we need to
> proxy
> > media and convert Web+WebSocket into SIP this is a definition of island
> > bridge for me.
>
> If you say that we need to "convert Web+WebSocket into SIP" then you
> are absolutely wrong, even more taking into account that
> "draft-ibc-sipcore-sip-websocket-01" (The WebSocket Protocol as a
> Transport for SIP) is being adopted by the IETF SIPCORE Working Group
> as a new SIP transport.
>
> The slide 8 in [2] shows a *pure* SIP proxy, and not a "WebSocket to
> SIP gateway". So ***NO*** island at signaling level, not at all.
>
>
> [1] http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-01
> [2] http://sip-on-the-web.aliax.net/
>
> This assumes ICE support in SIP end points. If they do not support ICE yo=
u
still need a media proxy driven by SIP proxy. Not that different from
encryption.



> >> Not in the case SDES+SRTP is allowed in WebRTC (see Fabius's recent
> >> mails about SDES and DTLS).
> >>
> >
> > Once again, we still need a gateway to support ICE. If end point is
> adding
> > ICE to interop with WebRTC, why not add DTLS-SRTP at the same time.
>
> Building an ICE-Lite RTP/SRTP tunnel/proxy is "easy". After ICE
> "handhake" the server can relay UDP packets as kernel space (conntrack
> in Linux), but if you need to decrypt and encrypt you need to do that
> at user space, which is much worse and requires much more CPU.
>
> You can still get STUN messages after initial handshake, so you need to
monitor/filter media. Not sure this is possible with conntrack. If you are
doing media proxy with re-encryption in the user space you can probably get
about 10GB/sec worth of throughput with the modern dual Xeon E5 server.


> > P.S. I actually do not like DTLS-SRTP, since it does not have a large
> > installed base, comes with too many features, slows down call setup, wi=
ll
> > requirecomplex interop and such.
>
> > I believe we should
> > design a new key exchange method, that can be mapped to SDES-SRTP via
> > signaling alone,
>
> And wouldn't that be a mechanism with "non a large installed base"?
>

I would prefer a simpler new protocol then a complex existing one with no
real user base. Something as simple as sending public key in the SDP and
sending session key in ICE handshake STUN message would be better the DTLS.
It is simple to implement, does not send keys over clear channel, and can
be mapped to SDES-SRTP using ICE-Lite proxy with no additional signaling
messages or doing re-encryption. This probably needs to be enhanced to
support public key algorithm/key length negotiation and replay protection,
but this should be trivial as well.


> > addresses SDES-SRTP issues (transmitting keys over clear
> > channel in signaling, replay of signaling)
>
> If HTTPS / WSS is used, there is no "clear channel". It would be easy
> to mandate *all* the browser communication using TLS for the case in
> which SDES-SRTP is used. That is a feasible and valid requirement
> IMHO.
>

HTTPS/WSS is used you provide better protection, but there are still issues
with cross-site scripting, web site security etc. You did prevent some of
the attacks, but you still nowhere near being secure. Once again, if we
have a key exchnage that can be mapped to SDES-SRTP, there is no need to
compromise.

> without introducing all the
> > problems related to DTLS-SRTP. Any number of methods using public key
> based
> > exchange over signaling and ICE should do the trick.
>
> So a complete new protocol. Bye any kind of interop (except if you
> have the magic super B2BUA).
>

I think idea of interop is gone for all practical purposes. We gave it up
when we made ICE a requirement.


> PS: Please consider reading Fabios's mails and reply to them. The
> discussion will be more interesting is such nice explanations are not
> ignored ;)
>
>
I've read it. Nice tutorial, but does not change anything I am saying here.

P.S. I have feeling that anything security related keeps being discussed
here without converging anywhere. My proposal (DTLS-SRTP by default,
SDES-SRTP from HTTPS sessions or RTP from HTTP session if enabled) got
ignored as well. This at least inline with current HTTP(S) security model.
______________
Roman Shpount

--e89a8ff1c650fe3ffa04bcf05b1b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 10:39 A=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</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 class=3D"im">2012/4/5 Roman Shpount &lt;<a href=3D"mailto:roman@teluri=
x.com">roman@telurix.com</a>&gt;:<br>
&gt;<br>
&gt; On Thu, Apr 5, 2012 at 5:04 AM, I=F1aki Baz Castillo &lt;<a href=3D"ma=
ilto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
</div><div class=3D"im">&gt;&gt; ICE support can be implemented in a ICE-Li=
te RTP/SRTP proxy, see:<br>
&gt;&gt;<br>
&gt;&gt; =A0<a href=3D"http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SD=
ES-SRTP.png" target=3D"_blank">http://public.aliax.net/WebRTC/WebRTC_SIP_In=
terop_SDES-SRTP.png</a><br>
&gt;<br>
&gt;<br>
&gt; This picture is not entirely correct -- ICE Lite proxy should be on bo=
th SIP<br>
&gt; and Media path.<br>
<br>
</div>Incorrect. The proxy would communicate with the ICE-Lite server(s)<br=
>
using some kind of proprietary protocol. This is already true in some<br>
SIP proxies (SER, Kamailio, OpenSer) and media-proxies (rtpproxy,<br>
mediaproxy, irtpproxy).<br>
<div class=3D"im"><br></div></blockquote><div>Internal protocols do not mat=
ter. The &quot;super B2BUA&quot; can be using a custom protocol between RTP=
 forwarder and SIP processor. Does not change the fact that from logical po=
int of view it is on both SIP and media path. At minimum you are missing a =
custom protocol connection between ICE-Lite gateway and SIP proxy. So, your=
 picture is still wrong.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im=
">
&gt;&gt; That is a *very* limited scope of what WebRTC can provide. An IT<b=
r>
&gt;&gt; department should be able to deploy its own WebRTC infrastructure =
(a<br>
&gt;&gt; Web+WebSocket server) within its &quot;local&quot; network, so bro=
wsers<br>
&gt;&gt; accessing to such a local website share the network with SIP/XMPP<=
br>
&gt;&gt; phones/devices/softphones.<br>
&gt;&gt;<br>
&gt;&gt; Please don&#39;t imagine WebRTC and SIP interop as the communicati=
on<br>
&gt;&gt; between two islands ;)<br>
&gt;<br>
&gt;<br>
&gt; I actually do imagine WebRTC/SIP interop as two islands. If we need to=
 proxy<br>
&gt; media and convert Web+WebSocket into SIP this is a definition of islan=
d<br>
&gt; bridge for me.<br>
<br>
</div>If you say that we need to &quot;convert Web+WebSocket into SIP&quot;=
 then you<br>
are absolutely wrong, even more taking into account that<br>
&quot;draft-ibc-sipcore-sip-websocket-01&quot; (The WebSocket Protocol as a=
<br>
Transport for SIP) is being adopted by the IETF SIPCORE Working Group<br>
as a new SIP transport.<br>
<br>
The slide 8 in [2] shows a *pure* SIP proxy, and not a &quot;WebSocket to<b=
r>
SIP gateway&quot;. So ***NO*** island at signaling level, not at all.<br>
<br>
<br>
[1] <a href=3D"http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-0=
1" target=3D"_blank">http://tools.ietf.org/html/draft-ibc-sipcore-sip-webso=
cket-01</a><br>
[2] <a href=3D"http://sip-on-the-web.aliax.net/" target=3D"_blank">http://s=
ip-on-the-web.aliax.net/</a><br>
<div class=3D"im">
<br></div></blockquote><div>This assumes ICE support in SIP end points. If =
they do not support ICE you still need a media proxy driven by SIP proxy. N=
ot that different from encryption.<br><br>=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex">
<div class=3D"im">
&gt;&gt; Not in the case SDES+SRTP is allowed in WebRTC (see Fabius&#39;s r=
ecent<br>
&gt;&gt; mails about SDES and DTLS).<br>
&gt;&gt;<br>
&gt;<br>
&gt; Once again, we still need a gateway to support ICE. If end point is ad=
ding<br>
&gt; ICE to interop with WebRTC, why not add DTLS-SRTP at the same time.<br=
>
<br>
</div>Building an ICE-Lite RTP/SRTP tunnel/proxy is &quot;easy&quot;. After=
 ICE<br>
&quot;handhake&quot; the server can relay UDP packets as kernel space (conn=
track<br>
in Linux), but if you need to decrypt and encrypt you need to do that<br>
at user space, which is much worse and requires much more CPU.<br>
<div class=3D"im"><br></div></blockquote><div>You can still get STUN messag=
es after initial handshake, so you need to monitor/filter media. Not sure t=
his is possible with conntrack. If you are doing media proxy with re-encryp=
tion in the user space you can probably get about 10GB/sec worth of through=
put with the modern dual Xeon E5 server.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im=
">
&gt; P.S. I actually do not like DTLS-SRTP, since it does not have a large<=
br>
&gt; installed base, comes with too many features, slows down call setup, w=
ill<br>
&gt; requirecomplex interop and such.<br>
<br>
</div><div class=3D"im">&gt; I believe we should<br>
&gt; design a new key exchange method, that can be mapped to SDES-SRTP via<=
br>
&gt; signaling alone,<br>
<br>
</div>And wouldn&#39;t that be a mechanism with &quot;non a large installed=
 base&quot;?<br></blockquote><div><br>I would prefer a simpler new protocol=
 then a complex existing one with no real user base. Something as simple as=
 sending public key in the SDP and sending session key in ICE handshake STU=
N message would be better the DTLS. It is simple to implement, does not sen=
d keys over clear channel, and can be mapped to SDES-SRTP using ICE-Lite pr=
oxy with no additional signaling messages or doing re-encryption. This prob=
ably needs to be enhanced to support public key algorithm/key length negoti=
ation and replay protection, but this should be trivial as well.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div class=3D"im">&gt; addresses SDES-SRTP issues (transmitting keys over c=
lear<br>
&gt; channel in signaling, replay of signaling)<br>
<br>
</div>If HTTPS / WSS is used, there is no &quot;clear channel&quot;. It wou=
ld be easy<br>
to mandate *all* the browser communication using TLS for the case in<br>
which SDES-SRTP is used. That is a feasible and valid requirement<br>
IMHO.<br>
<div class=3D"im"></div></blockquote><div><br>HTTPS/WSS is used you provide=
 better protection, but there are still issues with cross-site scripting, w=
eb site security etc. You did prevent some of the attacks, but you still no=
where near being secure. Once again, if we have a key exchnage that can be =
mapped to SDES-SRTP, there is no need to compromise. <br>
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"i=
m">

&gt; without introducing all the<br>
&gt; problems related to DTLS-SRTP. Any number of methods using public key =
based<br>
&gt; exchange over signaling and ICE should do the trick.<br>
<br>
</div>So a complete new protocol. Bye any kind of interop (except if you<br=
>
have the magic super B2BUA).<br></blockquote><div><br>I think idea of inter=
op is gone for all practical purposes. We gave it up when we made ICE a req=
uirement.<br>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt=
 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">


PS: Please consider reading Fabios&#39;s mails and reply to them. The<br>
discussion will be more interesting is such nice explanations are not<br>
ignored ;)<br>
<div><div></div><br></div></blockquote></div><br>I&#39;ve read it. Nice tut=
orial, but does not change anything I am saying here.<br><br>P.S. I have fe=
eling that anything security related keeps being discussed here without con=
verging anywhere. My proposal (DTLS-SRTP by default, SDES-SRTP from HTTPS s=
essions  or RTP from HTTP session if enabled) got ignored as well. This at =
least inline with current HTTP(S) security model.<br>
______________<br>Roman Shpount<br>

--e89a8ff1c650fe3ffa04bcf05b1b--

From igor.faynberg@alcatel-lucent.com  Thu Apr  5 08:49:24 2012
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Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication	(DTLS-SRTP / SDES-SRTP)
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Fabio,

I am finding these definitions hard to understand.  I am afraid that 
somehow the activity (encryption) gets mixed with the services 
(confidentiality and authentication).

  I think your major point is that effective encryption must rely on 
symmetric keys, and that the key distribution is crucial to ensuring 
confidentiality provably.  I agree with that.

I also agree with sepating the case when session keys are handed out by 
another entity (such as KDC) and the case where the keys are known only 
to the end-points.  The latter can be achieved with signed 
Diffie-Hellman (using PKI, or PAK, or EKE).

Confidentiality, of course, relies on authentication: If I encrypt 
something with the key I share only with you, I'd better know that I 
share this key ONLY with you.

But I don't get the point of the taxonomy you propose...

Igor

On 4/5/2012 6:16 AM, Fabio Pietrosanti (naif) wrote:
> Hi,
>
> i've been discussing with several friends about the current discussion
> on the security standard in this mailing lists, in particular regarding
> the topic of DTLS-SRTP trust model posted there
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04007.html .
>
> I found that there is no common definition for the concept of
> "end-to-end encryption" and there are a lot of misunderstanding about it.
>
> In particular, the fact that two peer can community by exchanging keys
> directly among them, tend to typically to be defined as end-to-end
> encryption.
>
> However the term "end-to-end encryption" have also a general
> misconception that's something that "no one can intercept" .
>
> Unfortunately this is not true for DTLS-SRTP, while for example it's
> true for OpenPGP/MIME or for ZRTP with SAS.
>
> We need to separate the 4 concepts:
>
> - end-to-end encryption
> Ability for a key management system to exchange keys directly without
> relying on intermediate server actively involved in encryption process.
>
> - end-to-end authentication
> Ability for end-user to authenticate the end-to-end encryption process
> without the need to rely on a single or multiple set of trusted third
> parties
>
> - end-to-site encryption
> Ability for a key management system to exchange keys with/trough the
> server with which data are exchanged, involving it in the authentication
> process.
>
> - end-to-site authentication
> Ability for end-user to authenticate the end-to-site encryption process
> based on the same security mechanism used to establish trust with the
> server with which data are exchanged.
>
>
> What i would like to focus is that:
>
> - DTLS does provide end-to-end encryption (in specific context of use)
> - DTLS does provide end-to-site authentication (rely on third party trust)
>
> - ZRTP does provide end-to-end encryption (in all context of use)
> - ZRTP does provide end-to-end authentication (does not rely on third
> party trust)
>
> - SDES-SRTP does provide end-to-site encryption (encryption with/trough
> your server)
> - SDES-SRTP does provide end-to-site authentication (you trust your
> server involved in key exchange)
>
> So when we tend to think about the "how much security a technology
> provide" we should probably compare in a scale:
>
> - ZRTP
>    - end-to-end encryption
>    - end-to-end authentication
> - DTLS-SRTP
>    - end-to-end encryption
>    - end-to-site authentication
> - SDES-SRTP
>    - end-to-site encryption
>    - end-to-site authentication
>
> So currently we can affirm that:
>
> - ZRTP does not rely on third party trust for effective security
> - DTLS-SRTP rely on third party trust for effective security
> - SDES-SRTP rely on third party trust for effective security
>
> This is the *MOST IMPORTANT* distinction for an encryption technology:
> 			WHO SHOULD I TRUST?
>
> Well, basically it seems to me that DTLS-SRTP and SDES-SRTP both require
>
> So, my point are to:
>
> - Introduce SDES-SRTP for compatibility and simplicity
>    - Specify that the Browser will need to provide indicate the security
> level to the user (like the lock of HTTPS, same security model)
>
> - Introduce end-to-end authentication support for DTLS-SRTP via SAS
>    - Specify that the browser will need to to provide the end-user way to
> use end-to-end authentication and indicate the security level to the user.
>
> Then it will be up to the signaling server and/or to the browser
> settings to specificy the required security model:
> - end-to-end encryption + end-to-end authentication
> or
> - end-to-site encryption + end-to-site authentication
>
> But please don't mix those two as it will be *Very difficult, near to
> impossible* to explain to the user "WHO SHOULD HE TRUST" .
>
> Fabio
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 17:54:18 +0200
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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2012/4/5 Roman Shpount <roman@telurix.com>:
>
> On Thu, Apr 5, 2012 at 10:39 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:

>> Incorrect. The proxy would communicate with the ICE-Lite server(s)
>> using some kind of proprietary protocol. This is already true in some
>> SIP proxies (SER, Kamailio, OpenSer) and media-proxies (rtpproxy,
>> mediaproxy, irtpproxy).

> Internal protocols do not matter. The "super B2BUA" can be using a custom
> protocol between RTP forwarder and SIP processor. Does not change the fac=
t
> that from logical point of view it is on both SIP and media path. At mini=
mum
> you are missing a custom protocol connection between ICE-Lite gateway and
> SIP proxy. So, your picture is still wrong.

Too much lines if I draw such a custom protocol ;)


> This assumes ICE support in SIP end points. If they do not support ICE yo=
u
> still need a media proxy driven by SIP proxy. Not that different from
> encryption.

Lot's of SIP providers force the media throught a RTP proxy, which
indeed is a protocol violation since such proxies do alter the SDP
which is not allowed (theorically). But such a RTP proxy is just a UDP
tunnel so it can be done at kernel space (and it is).





>> Building an ICE-Lite RTP/SRTP tunnel/proxy is "easy". After ICE
>> "handhake" the server can relay UDP packets as kernel space (conntrack
>> in Linux), but if you need to decrypt and encrypt you need to do that
>> at user space, which is much worse and requires much more CPU.
>>
> You can still get STUN messages after initial handshake, so you need to
> monitor/filter media. Not sure this is possible with conntrack.

It's possible, sure. First packets are handled in user space, and
later go to kernel space with conntrack rules. It's real and there are
two RTP proxies doing that: MediaProxy and irtpProxy.



> If you are
> doing media proxy with re-encryption in the user space you can probably g=
et
> about 10GB/sec worth of throughput with the modern dual Xeon E5 server.

If the "EKT update to SIP re-INVITE" is not required that's a good new
(at least). What I would HATE is having to deal with a signaling B2BUA
for interop with SIP.




>> If HTTPS / WSS is used, there is no "clear channel". It would be easy
>> to mandate *all* the browser communication using TLS for the case in
>> which SDES-SRTP is used. That is a feasible and valid requirement
>> IMHO.
>
>
> HTTPS/WSS is used you provide better protection, but there are still issu=
es
> with cross-site scripting, web site security etc. You did prevent some of
> the attacks, but you still nowhere near being secure. Once again, if we h=
ave
> a key exchnage that can be mapped to SDES-SRTP, there is no need to
> compromise.

I'm sure that DTLS does not solve *all* those problems.




> I think idea of interop is gone for all practical purposes. We gave it up
> when we made ICE a requirement.

I don't agree. At least ICE is specified for SIP and XMPP-Jingle. SIP
vendors are lazy since they rely of annoying SBC's that break
everything but gives them money, so they "don't need" ICE. But if
WebRTC becomes real then SIP vendors will read the ICE RFC and can
implement it.

The same is not true if a new DTLS-XXXX spec is born for WebRTC since
that would require a new standard also for SIP, XMPP and so.



> P.S. I have feeling that anything security related keeps being discussed
> here without converging anywhere.

Well, I'm learning a lot about SRTP, SDES and DTLS XD


> My proposal (DTLS-SRTP by default,
> SDES-SRTP from HTTPS sessions or RTP from HTTP session if enabled) got
> ignored as well. This at least inline with current HTTP(S) security model=
.

So you proposed allowing plain RTP and now just a new DTLS-XXXX-SRTP
is valid for you? :)


Well, ok, IMHO there is enough debate for now. Let's wait a bit :)


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

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--e89a8fb208dcbcce4204bcf10d81
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On Thu, Apr 5, 2012 at 11:54 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2012/4/5 Roman Shpount <roman@telurix.com>:
> It's possible, sure. First packets are handled in user space, and
> later go to kernel space with conntrack rules. It's real and there are
> two RTP proxies doing that: MediaProxy and irtpProxy.
>
> I do not believe either one of them does ICE-Lite to RTP. As far as I
know, they are both RTP-to-RTP only.


> > My proposal (DTLS-SRTP by default,
> > SDES-SRTP from HTTPS sessions or RTP from HTTP session if enabled) got
> > ignored as well. This at least inline with current HTTP(S) security
> model.
>
> So you proposed allowing plain RTP and now just a new DTLS-XXXX-SRTP
> is valid for you? :)
>

I am still proposing the same thing -- some new protocol instead of
DTLS-SRTP, which does public key key exchange, and SDES-SRTP from HTTPS
with RTP from HTTP if enabled. This should allow the widest possible
interop without forcing a security model on them. I do not belive this
compromises security since it allows the application developer pick the
security model they want (new and most secure, old but interoperable, or
none at all).
______________
Roman Shpount

--e89a8fb208dcbcce4204bcf10d81
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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 11:54 A=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</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 class=3D"im">2012/4/5 Roman Shpount &lt;<a href=3D"mailto:roman@teluri=
x.com">roman@telurix.com</a>&gt;:<br>
</div>It&#39;s possible, sure. First packets are handled in user space, and=
<br>
later go to kernel space with conntrack rules. It&#39;s real and there are<=
br>
two RTP proxies doing that: MediaProxy and irtpProxy.<br>
<div class=3D"im"><br></div></blockquote><div>I do not believe either one o=
f them does ICE-Lite to RTP. As far as I know, they are both RTP-to-RTP onl=
y.<br>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0p=
t 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div class=3D"im">
&gt; My proposal (DTLS-SRTP by default,<br>
&gt; SDES-SRTP from HTTPS sessions or RTP from HTTP session if enabled) got=
<br>
&gt; ignored as well. This at least inline with current HTTP(S) security mo=
del.<br>
<br>
</div>So you proposed allowing plain RTP and now just a new DTLS-XXXX-SRTP<=
br>
is valid for you? :)<br>
</blockquote><div><br>I am still proposing the same thing -- some new proto=
col instead of DTLS-SRTP, which does public key key exchange, and SDES-SRTP=
 from HTTPS with RTP from HTTP if enabled. This should allow the widest pos=
sible interop without forcing a security model on them. I do not belive thi=
s compromises security since it allows the application developer pick the s=
ecurity model they want (new and most secure, old but interoperable, or non=
e at all). <br>
</div></div>______________<br>Roman Shpount<br>

--e89a8fb208dcbcce4204bcf10d81--

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2012/4/5 Roman Shpount <roman@telurix.com>:
>> 2012/4/5 Roman Shpount <roman@telurix.com>:
>> It's possible, sure. First packets are handled in user space, and
>> later go to kernel space with conntrack rules. It's real and there are
>> two RTP proxies doing that: MediaProxy and irtpProxy.
>>
> I do not believe either one of them does ICE-Lite to RTP. As far as I kno=
w,
> they are both RTP-to-RTP only.

Sure. I just say that "it's feasible" to do.



> I am still proposing the same thing -- some new protocol instead of
> DTLS-SRTP, which does public key key exchange, and SDES-SRTP from HTTPS w=
ith
> RTP from HTTP if enabled. This should allow the widest possible interop
> without forcing a security model on them. I do not belive this compromise=
s
> security since it allows the application developer pick the security mode=
l
> they want (new and most secure, old but interoperable, or none at all).

draft required :)


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

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Subject: [rtcweb] DTLS-SRTP with end-to-end security: Short Authentication String
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Hi,

i've been reviewing the WebRTC security architecture and it seems to me
that the authors did not considered the value of end-to-end security.

DTLS-SRTP does not provide end-to-end security because it rely on third
party of key authentication.

In the RTCWeb security there is a proposed solution, to use SAS:
http://tools.ietf.org/html/draft-ietf-rtcweb-security-02#section-4.3.2.2

SAS, Short authentication string, allow to achieve end-to-end security
within the DTLS-SRTP end-to-end encryption protocol.

Unfortunately the author of rtcweb-security described the SAS as a bad
way to verify fingerprints.

I would like to argue that SAS are instead a valuable, secure mechanism
to provide end-to-end authentication to an end-to-end encryption system,
thus realizing end-to-end security.

I would like to bring the attention of the WG to provide a MANDATORY
requirement to enable SAS verification.

RFC 6189 (ZRTP) explain the reason why SAS are secure enough and the
comments on the rtcweb-security draft should be reconsidered:
		
		http://tools.ietf.org/html/rfc6189#section-15

  "Some questions have been raised about voice spoofing during the short
   authentication string (SAS) comparison.  But it is a mistake to think
   this is simply an exercise in voice impersonation (perhaps this could
   be called the "Rich Little" attack).  Although there are digital
   signal processing techniques for changing a person's voice, that does
   not mean a MiTM attacker can safely break into a phone conversation
   and inject his own SAS at just the right moment.  He doesn't know
   exactly when or in what manner the users will choose to read aloud
   the SAS, or in what context they will bring it up or say it, or even
   which of the two speakers will say it, or if indeed they both will
   say it.  In addition, some methods of rendering the SAS involve using
   a list of words such as the PGP word list[Juola2], in a manner
   analogous to how pilots use the NATO phonetic alphabet to convey
   information.  This can make it even more complicated for the
   attacker, because these words can be worked into the conversation in
   unpredictable ways.  If the session also includes video (an
   increasingly common usage scenario), the MiTM may be further deterred
   by the difficulty of making the lips sync with the voice-spoofed SAS.
   The PGP word list is designed to make each word phonetically
   distinct, which also tends to create distinctive lip movements.
   Remember that the attacker places a very high value on not being
   detected, and if he makes a mistake, he doesn't get to do it over.

   A question has been raised regarding the safety of the SAS procedure
   for people who don't know each other's voices, because it may allow
   an attack from a MiTM even if he lacks voice impersonation
   capabilities.  This is not as much of a problem as it seems, because
   it isn't necessary that users recognize each other by their voice.
   It is only necessary that they detect that the voice used for the SAS
   procedure doesn't match the voice in the rest of the phone
   conversation."

In any crypto-system if the user is not able to INDEPENDENTLY verify the
integrity of the key exchange we refer it to END-TO-SITE SECURITY.

In any crypto-system if the user is able to INDEPENDENTLY verify the
integrity of the key exchange, we refer it to END-TO-END SECURITY.

So basically WebRTC currently does NOT guarantee end-to-end security.
But making SAS mandatory will allow independent way to verify the keys.

Fabio

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Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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On 4/5/12 5:49 PM, Igor Faynberg wrote:
> Fabio,
> 
> I am finding these definitions hard to understand.  I am afraid that
> somehow the activity (encryption) gets mixed with the services
> (confidentiality and authentication).
> 
>  I think your major point is that effective encryption must rely on
> symmetric keys, and that the key distribution is crucial to ensuring
> confidentiality provably.  I agree with that.
> 
> I also agree with sepating the case when session keys are handed out by
> another entity (such as KDC) and the case where the keys are known only
> to the end-points.  The latter can be achieved with signed
> Diffie-Hellman (using PKI, or PAK, or EKE).
> 
> Confidentiality, of course, relies on authentication: If I encrypt
> something with the key I share only with you, I'd better know that I
> share this key ONLY with you.
> 
> But I don't get the point of the taxonomy you propose...

Well, the main point is that DTLS-SRTP claim to provide end-to-end
encryption, however the user is not able to deliver "end-to-end security" .

end-to-end encryption != end-to-end security .

In the encryption technologies landscape, when we refer to end-to-end
encryption we always refer to cryptographic architecture where the users
communicating are guaranteed respect to the underlying transport and
communication architecture.

This bring a misleading concept on DTLS-SRTP in WebRTC because it's
referred as end-to-end encryption, but it doesn't provide end-to-end
security like other encryption technologies does (ZRTP, OpenPGP, OTR, etc).

DTLS-SRTP rely on third party for security verification.

DTLS-SRTP = end-to-site security
SRTP-SDES = end-to-site security
ZRTP = end-to-end security

This means that DTLS-SRTP, from a trust-model point of view, does not
provide end-to-end security because there will always be a trusted third
party able to authorize Man in the Middle to do eavesdropping.

>From that point of view SDES-SRTP provide very similar warranty, except
that you have to trust only 1 third party rather than 2 third party.

Fabio

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On Thu, Apr 5, 2012 at 1:07 PM, Fabio Pietrosanti (naif) <
lists@infosecurity.ch> wrote:

> This means that DTLS-SRTP, from a trust-model point of view, does not
> provide end-to-end security because there will always be a trusted third
> party able to authorize Man in the Middle to do eavesdropping.
>

Incorrect. If fingerprint is exposed and can be verified, DTLS-SRTP does
provide end-to-end security. No third parties involved.
______________
Roman Shpount

--047d7b33da10ab4be704bcf20f8e
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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 1:07 PM=
, Fabio Pietrosanti (naif) <span dir=3D"ltr">&lt;<a href=3D"mailto:lists@in=
fosecurity.ch">lists@infosecurity.ch</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">
This means that DTLS-SRTP, from a trust-model point of view, does not<br>
provide end-to-end security because there will always be a trusted third<br=
>
party able to authorize Man in the Middle to do eavesdropping.<br></blockqu=
ote></div><br>Incorrect. If fingerprint is exposed and can be verified, DTL=
S-SRTP does provide end-to-end security. No third parties involved.<br>
______________<br>Roman Shpount<br>

--047d7b33da10ab4be704bcf20f8e--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 19:43:52 +0200
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Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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2012/4/5 Roman Shpount <roman@telurix.com>:
> On Thu, Apr 5, 2012 at 1:07 PM, Fabio Pietrosanti (naif)
> <lists@infosecurity.ch> wrote:
>>
>> This means that DTLS-SRTP, from a trust-model point of view, does not
>> provide end-to-end security because there will always be a trusted third
>> party able to authorize Man in the Middle to do eavesdropping.
>
>
> Incorrect. If fingerprint is exposed and can be verified, DTLS-SRTP does
> provide end-to-end security. No third parties involved.

The fingerprint is included in the SDP which goes throught the Web
server, right? You know what I mean ;)

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

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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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On 4/5/2012 9:53 AM, Roman Shpount wrote:
>
> On Thu, Apr 5, 2012 at 5:04 AM, Iñaki Baz Castillo <ibc@aliax.net
> <mailto:ibc@aliax.net>> wrote:
>
>     2012/4/5 Roman Shpount <roman@telurix.com <mailto:roman@telurix.com>>:
>      > On the more serious note, very few SIP end points offer working
>     ICE support.
>      > So, in a large sense, interop with them is not an option.
>
>     For me there is a BIG difference when a kind of B2BUA is required
>     (which involves signaling "transaction"). That's the barrier IMHO.
>
>     ICE support can be implemented in a ICE-Lite RTP/SRTP proxy, see:
>
>     http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_SDES-SRTP.png
>
>
> This picture is not entirely correct -- ICE Lite proxy should be on both
> SIP and Media path.

I agree.

>     The problem arises when media encrypt/decrypt is required, and evenr
>     more when a key update in RTP (like the DTLS EKT update) must be
>     converted into a signaling re-INVITE by a super Signaling+Media B2BUA:
>
>     http://public.aliax.net/WebRTC/WebRTC_SIP_Interop_DTLS-EKT-SRTP.png
>
> Not sure why key update needs to be converted into signaling if media is
> re-encrypted. Key updates on both sides of the proxy are independent.
> Such proxy can also change SDES-SRTP key on each re-INVITE initiated by
> either side, but apart from doing actual encryption/decryption,
> operation wise it should not be any different then ICE-Lite proxy.

I think he's assuming the gateway/B2BUA would not decrypt/re-encrypt, 
but would proxy keys from one protocol to the other.  That's obviously 
lower overhead, but a more complex protocol.  I have never assumed that 
proxying of keys would be the default.

>
>      > Out of the ones
>      > that do support ICE and SRTP, very few are actually connected
>     directly to a
>      > public internet. Most of them are connected to some sort of PBX
>     or an IP PBX
>      > type service. So, in reality you do not need to bridge every IP
>     phone with
>      > WebRTC.
>
>     That is a *very* limited scope of what WebRTC can provide. An IT
>     department should be able to deploy its own WebRTC infrastructure (a
>     Web+WebSocket server) within its "local" network, so browsers
>     accessing to such a local website share the network with SIP/XMPP
>     phones/devices/softphones.
>
>     Please don't imagine WebRTC and SIP interop as the communication
>     between two islands ;)
>
>
> I actually do imagine WebRTC/SIP interop as two islands. If we need to
> proxy media and convert Web+WebSocket into SIP this is a definition of
> island bridge for me.
>
>      > If a few PBX and hosted centrex vendors will add support for WebRTC
>      > required features, we will get compatibility with existing end
>     points.
>
>     Pure SIP proxies do exist. A PBX is not always needed, so again, SIP
>     world does not require to be "an island".
>
>
> Once  again, if SIP proxy is combined with media proxy, you can bridge
> SIP and WebRTC, even if DTLS-SRTP is used.
>
>      > To support the rest you will need to deploy some sort of gateway.
>
>     Not in the case SDES+SRTP is allowed in WebRTC (see Fabius's recent
>     mails about SDES and DTLS).
>
> Once again, we still need a gateway to support ICE. If end point is
> adding ICE to interop with WebRTC, why not add DTLS-SRTP at the same time.
>
>      > hurdle (I am trying to be politically correct here). WebRTC
>     enabled end
>      > points, on the other hand, will offer significant benefits to
>     traditional
>      > SIP phones, since they will allow development of higher quality
>     integrated
>      > real time communications services. I hope this will drive a much
>     quicker
>      > standard adoption.
>
>     The problem is that there is a *NEW* SRTP related spec for WebRTC:
>
>     http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-03
>
>     Maybe it could also become a standard for SIP? I hope, but currently
>     it's not, so by *mandating* DTLS-EKT-SRTP in WebRTC we are creating a
>     island.
>
>
> Same as above.
>
> P.S. I actually do not like DTLS-SRTP, since it does not have a large
> installed base, comes with too many features, slows down call setup,
> will requirecomplex interop and such. At the same time I do not think
> that interop with legacy SDES-SRTP is a reason not to use it. I believe
> we should design a new key exchange method, that can be mapped to
> SDES-SRTP via signaling alone, addresses SDES-SRTP issues (transmitting
> keys over clear channel in signaling, replay of signaling) without
> introducing all the problems related to DTLS-SRTP. Any number of methods
> using public key based exchange over signaling and ICE should do the trick.

I find this argument more compelling than the previous one.  Call setup 
is certainly an issue (though I believe there are answers to it).  If 
there's a good alternative (and of course I remember the IETF 
presentation from a number of years ago that showed all 14 or so extant 
key-exchange methods, none of which was an obvious choice (witness all 
the problems actually ever agreeing on one, which is why SDES has become 
the lowest-common-denominator exchange, even though most don't actually 
use it).

However, the bar for introducing Yet Another Key Exchange Protocol at 
this late date is pretty darn high, unless it's amazingly simple (and 
likely insecure or with other unwelcome properties) or a modification of 
an existing protocol.

I strongly dislike SDES for a number of reasons - not solely that the 
key is exposed to the server (as in the server/app may be evil or 
hacked), but exposure of the keys will likely engender requirements from 
authorities to record keys to allow for later decryption of media. 
That's a direct threat to privacy (especially given many organizations 
willingness to provide access without legal orders).  Also, once those 
keys are recorded, the database of such keys becomes a tempting target 
in of itself (though it's only of use to someone tapping the raw 
packets, which limits the exposure - governments generally have no 
problem getting that; individuals have more problems (but can)).  And as 
mentioned earlier, it makes the servers much more of a target for 
hacking and requests/orders from authorities.


-- 
Randell Jesup
randell-ietf@jesup.org

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Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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On 4/5/2012 1:07 PM, Fabio Pietrosanti (naif) wrote:
> ...
> Well, the main point is that DTLS-SRTP claim to provide end-to-end
> encryption, however the user is not able to deliver "end-to-end security" .
>
> end-to-end encryption != end-to-end security .

Yes, so far this is how I read the present state of affairs.  We had a 
very interesting meeting with BrowserID people, but it was too short--at 
least for me--to figure everything out.

I find the issue of self-signed certificates problematic, too.

DTLS-SRTP would provide end-to-end security though if 1) the end points 
had a common PKI chain of trust or 2) if each of them could be 
authenticated to another by an assertion from a trusted IdP.  In neither 
case there would be a key escrow, right?

A level below is the KDC (Kerberos-like) arrangement.  Here both end 
points would need to trust the KDC, and  there are serious business 
cases for that.




> ...
> This bring a misleading concept on DTLS-SRTP in WebRTC because it's
> referred as end-to-end encryption, but it doesn't provide end-to-end
> security like other encryption technologies does (ZRTP, OpenPGP, OTR, etc).
>
> DTLS-SRTP rely on third party for security verification.
>
> DTLS-SRTP = end-to-site security
> SRTP-SDES = end-to-site security
> ZRTP = end-to-end security
>
> This means that DTLS-SRTP, from a trust-model point of view, does not
> provide end-to-end security because there will always be a trusted third
> party able to authorize Man in the Middle to do eavesdropping.
>
> > From that point of view SDES-SRTP provide very similar warranty, except
> that you have to trust only 1 third party rather than 2 third party.

Yes, now I understand what you were saying. I agree.  SRTP-SDES is, in 
fact, what should work with KDC.

And, to throw oil to the fire, did take a look at IBAKE RFCs?  This 
provides an interesting example of what you call end-to-site.

Igor

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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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On 4/5/12 7:42 PM, Roman Shpount wrote:
> 
> On Thu, Apr 5, 2012 at 1:07 PM, Fabio Pietrosanti (naif)
> <lists@infosecurity.ch <mailto:lists@infosecurity.ch>> wrote:
> 
>     This means that DTLS-SRTP, from a trust-model point of view, does not
>     provide end-to-end security because there will always be a trusted third
>     party able to authorize Man in the Middle to do eavesdropping.
> 
> 
> Incorrect. If fingerprint is exposed and can be verified, DTLS-SRTP does
> provide end-to-end security. No third parties involved.

No, you are wrong in the understanding.

The fingerprint is always delivered from the signaling services, so by
the HTTPS website providing the JS calling application.

>From :
                      RTCWEB Security Architecture
http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-01#section-4.1

4.1.  Initial Signaling
[...]

In either case, it will contain:

   o  Media channel information
   o  ICE candidates
   o  A fingerprint attribute binding the communication to Alice's
      public key [RFC5763]

[...]
   This message is sent to the signaling server, e.g., by XMLHttpRequest
   [XmlHttpRequest] or by WebSockets [RFC6455] The signaling server
   processes the message from Alice's browser


So basically the "signaling service" provide the fingerprint information.

Additionally, in case you want to trust the fingerprint verification via
an identity provider, please read below:
From:
		RTCWEB Generic Identity Provider Interface
http://tools.ietf.org/html/draft-rescorla-rtcweb-generic-idp-00#section-5.4.1

5.4.3.1.  Authenticating Party
[...]
   o  If a IdP is provided by the calling application use that.


The IdP is provide *by the signaling service*.


So basically:

- The user get the calling application from signaling service
- The user get the fingerprint from signaling service
- The user get the IdP from the signaling service


So basically the signaling service can ALWAYS do MITM and provide
eavesdropping services to authorized and non-authorized users.

>From a trust model point of view:

		DTLS-SRTP = SDES-RTP

Because in both case the "signaling service" is a trusted third party
able to intercept phone calls.

It's important to define clearly somewhere that "WebRTC does not provide
end-to-end security"

Fabio

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Date: Thu, 5 Apr 2012 14:14:06 -0400
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From: Roman Shpount <roman@telurix.com>
To: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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On Thu, Apr 5, 2012 at 1:42 PM, Randell Jesup <randell-ietf@jesup.org>wrote:

> However, the bar for introducing Yet Another Key Exchange Protocol at this
> late date is pretty darn high, unless it's amazingly simple (and likely
> insecure or with other unwelcome properties) or a modification of an
> existing protocol.
>
> My issue is that patching DTLS-SRTP also comes with a fairly high cost now
and for the foreseeable future, since each of the patches will require an
independent interop. We either need to define and make all of those patches
a requirement, or each end point communicating with WebRTC will need to
verify if public-key exchange is supported, if ICE key exchange is
supported , and such. Each option provides a different key exchange path,
which would be a challenge for interop and provide a potential attack
vector (from my experience, protocols with a lot of complex scenarios
serving the same purpose are typically easier to break into).


> I strongly dislike SDES for a number of reasons - not solely that the key
> is exposed to the server (as in the server/app may be evil or hacked), but
> exposure of the keys will likely engender requirements from authorities to
> record keys to allow for later decryption of media. That's a direct threat
> to privacy (especially given many organizations willingness to provide
> access without legal orders).  Also, once those keys are recorded, the
> database of such keys becomes a tempting target in of itself (though it's
> only of use to someone tapping the raw packets, which limits the exposure -
> governments generally have no problem getting that; individuals have more
> problems (but can)).  And as mentioned earlier, it makes the servers much
> more of a target for hacking and requests/orders from authorities.
>

I dislike SDES for being a "somewhat secure" protocol. This is a very
dangerous territory where end user is under impression of being protected,
but reality is different. IMHO, in such situations it is better to have no
security at all, so that users know not to do anything they will regret.
______________
Roman Shpount

--047d7b10d0676e679904bcf2812b
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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 1:42 PM=
, Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf@jesup.=
org">randell-ietf@jesup.org</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">

However, the bar for introducing Yet Another Key Exchange Protocol at this =
late date is pretty darn high, unless it&#39;s amazingly simple (and likely=
 insecure or with other unwelcome properties) or a modification of an exist=
ing protocol.<br>

<br></blockquote><div>My issue is that patching DTLS-SRTP also comes with a=
 fairly high cost now and for the foreseeable future, since each of the pat=
ches will require an independent interop. We either need to define and make=
 all of those patches a requirement, or each end point communicating with W=
ebRTC will need to verify if public-key exchange is supported, if ICE key e=
xchange is supported , and such. Each option provides a different key excha=
nge path, which would be a challenge for interop and provide a potential at=
tack vector (from my experience, protocols with a lot of complex scenarios =
serving the same purpose are typically easier to break into).<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex">
I strongly dislike SDES for a number of reasons - not solely that the key i=
s exposed to the server (as in the server/app may be evil or hacked), but e=
xposure of the keys will likely engender requirements from authorities to r=
ecord keys to allow for later decryption of media. That&#39;s a direct thre=
at to privacy (especially given many organizations willingness to provide a=
ccess without legal orders). =A0Also, once those keys are recorded, the dat=
abase of such keys becomes a tempting target in of itself (though it&#39;s =
only of use to someone tapping the raw packets, which limits the exposure -=
 governments generally have no problem getting that; individuals have more =
problems (but can)). =A0And as mentioned earlier, it makes the servers much=
 more of a target for hacking and requests/orders from authorities.<font co=
lor=3D"#888888"></font><br>
</blockquote></div><br>I dislike SDES for being a &quot;somewhat secure&quo=
t; protocol. This is a very dangerous territory where end user is under imp=
ression of being protected, but reality is different. IMHO, in such situati=
ons it is better to have no security at all, so that users know not to do a=
nything they will regret. <br>
______________<br>Roman Shpount<br>

--047d7b10d0676e679904bcf2812b--

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References: <4F7D7103.6040102@infosecurity.ch> <4F7DBEFC.6040302@alcatel-lucent.com> <4F7DD13F.2010006@infosecurity.ch> <CAD5OKxv_e9Ncw7xt3eh9jNM9HWX1snDN1wVynkFT2GPoA+y1_w@mail.gmail.com> <4F7DE01C.4040800@infosecurity.ch>
Date: Thu, 5 Apr 2012 14:17:21 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>
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Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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On Thu, Apr 5, 2012 at 2:10 PM, Fabio Pietrosanti (naif) <
lists@infosecurity.ch> wrote:

> On 4/5/12 7:42 PM, Roman Shpount wrote:
> >
> > On Thu, Apr 5, 2012 at 1:07 PM, Fabio Pietrosanti (naif)
> > <lists@infosecurity.ch <mailto:lists@infosecurity.ch>> wrote:
> >
> >     This means that DTLS-SRTP, from a trust-model point of view, does not
> >     provide end-to-end security because there will always be a trusted
> third
> >     party able to authorize Man in the Middle to do eavesdropping.
> >
> >
> > Incorrect. If fingerprint is exposed and can be verified, DTLS-SRTP does
> > provide end-to-end security. No third parties involved.
>
> No, you are wrong in the understanding.
>
> The fingerprint is always delivered from the signaling services, so by
> the HTTPS website providing the JS calling application.
>
> If fingerprint is exposed to the user and be compared through some
alternative communications channel, the it can provide an independent
security validation similar to the one used in ZRTP. If signaling server
replaces the fingerprint for some sort of attack this can be detected, even
though it can be argued that very few people will do so.
______________
Roman Shpount

--047d7b33da1014a87804bcf28d42
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<br clear=3D"all"><br><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 2:1=
0 PM, Fabio Pietrosanti (naif) <span dir=3D"ltr">&lt;<a href=3D"mailto:list=
s@infosecurity.ch">lists@infosecurity.ch</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex">
<div class=3D"im">On 4/5/12 7:42 PM, Roman Shpount wrote:<br>
&gt;<br>
&gt; On Thu, Apr 5, 2012 at 1:07 PM, Fabio Pietrosanti (naif)<br>
</div><div class=3D"im">&gt; &lt;<a href=3D"mailto:lists@infosecurity.ch">l=
ists@infosecurity.ch</a> &lt;mailto:<a href=3D"mailto:lists@infosecurity.ch=
">lists@infosecurity.ch</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =A0 =A0 This means that DTLS-SRTP, from a trust-model point of view, d=
oes not<br>
&gt; =A0 =A0 provide end-to-end security because there will always be a tru=
sted third<br>
&gt; =A0 =A0 party able to authorize Man in the Middle to do eavesdropping.=
<br>
&gt;<br>
&gt;<br>
&gt; Incorrect. If fingerprint is exposed and can be verified, DTLS-SRTP do=
es<br>
&gt; provide end-to-end security. No third parties involved.<br>
<br>
</div>No, you are wrong in the understanding.<br>
<br>
The fingerprint is always delivered from the signaling services, so by<br>
the HTTPS website providing the JS calling application.<br>
<br></blockquote><div>If fingerprint is exposed to the user and be compared=
 through some alternative communications channel, the it can provide an ind=
ependent security validation similar to the one used in ZRTP. If signaling =
server replaces the fingerprint for some sort of attack this can be detecte=
d, even though it can be argued that very few people will do so.<br>
</div></div>______________<br>Roman Shpount<br>

--047d7b33da1014a87804bcf28d42--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 20:19:39 +0200
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Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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2012/4/5 Roman Shpount <roman@telurix.com>:
>> The fingerprint is always delivered from the signaling services, so by
>> the HTTPS website providing the JS calling application.
>>
> If fingerprint is exposed to the user and be compared through some
> alternative communications channel, the it can provide an independent
> security validation similar to the one used in ZRTP.

Define such an "alternative communications channel" and explain me how
the signaling server cannot alter that channel.



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

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 5 Apr 2012 20:21:04 +0200
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Subject: Re: [rtcweb] WebRTC-SIP interop: and why SDES-SRTP is a need
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2012/4/5 Roman Shpount <roman@telurix.com>:
> I dislike SDES for being a "somewhat secure" protocol. This is a very
> dangerous territory where end user is under impression of being protected=
,
> but reality is different. IMHO, in such situations it is better to have n=
o
> security at all, so that users know not to do anything they will regret.

And the same is true for DTLS, so?

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

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--e89a8ff2521c0675f904bcf2a6c6
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On Thu, Apr 5, 2012 at 2:19 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> Define such an "alternative communications channel" and explain me how
> the signaling server cannot alter that channel.
>
>
I will send it to you using Frogo, my trusty carrier pidgin ;).

I can read them to you (possible to modify but much harder then simple
signaling), I can email them to you using some sort of trusted service, or
I can call you using old fashioned telephone. I remember times when such
keys were exchanged via fax.
______________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 2:19 PM=
, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.ne=
t">ibc@aliax.net</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">
Define such an &quot;alternative communications channel&quot; and explain m=
e how<br>
the signaling server cannot alter that channel.<br>
<div><div></div><br></div></blockquote><div><br>I will send it to you using=
 Frogo, my trusty carrier pidgin ;).<br><br>I can read them to you (possibl=
e to modify but much harder then simple signaling), I can email them to you=
 using some sort of trusted service, or I can call you using old fashioned =
telephone. I remember times when such keys were exchanged via fax. <br>
</div></div>______________<br>Roman Shpount<br>

--e89a8ff2521c0675f904bcf2a6c6--

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On Thu, Apr 5, 2012 at 2:21 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2012/4/5 Roman Shpount <roman@telurix.com>:
> > I dislike SDES for being a "somewhat secure" protocol. This is a very
> > dangerous territory where end user is under impression of being
> protected,
> > but reality is different. IMHO, in such situations it is better to have
> no
> > security at all, so that users know not to do anything they will regret=
.
>
> And the same is true for DTLS, so?
>
> This can be mitigated with either identity validation or via fingerprint
validation via an independent channel. For SDES-SRTP the only way is secure
is that everything on its path is secure, which is much harder to validate.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 2:21 PM=
, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.ne=
t">ibc@aliax.net</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">
2012/4/5 Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telur=
ix.com</a>&gt;:<br>
<div class=3D"im">&gt; I dislike SDES for being a &quot;somewhat secure&quo=
t; protocol. This is a very<br>
&gt; dangerous territory where end user is under impression of being protec=
ted,<br>
&gt; but reality is different. IMHO, in such situations it is better to hav=
e no<br>
&gt; security at all, so that users know not to do anything they will regre=
t.<br>
<br>
</div>And the same is true for DTLS, so?<br>
<div><div></div><div class=3D"h5"><br></div></div></blockquote><div>This ca=
n be mitigated with either identity validation or via fingerprint validatio=
n via an independent channel. For SDES-SRTP the only way is secure is that =
everything on its path is secure, which is much harder to validate.<br>
</div></div>_____________<br>Roman Shpount<br>

--f46d040f9baebd677104bcf2d41c--

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2012/4/5 Roman Shpount <roman@telurix.com>:
> This can be mitigated with either identity validation

This argument has been already refuted by Fabio, am I right? The
identity is provided by the signaling server so nothing new here.


> or via fingerprint
> validation via an independent channel.

Again, define such an "independent channel" and let me know how the
signaling server CANNOT alter it.


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

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2012/4/5 Roman Shpount <roman@telurix.com>:
> On Thu, Apr 5, 2012 at 2:19 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> w=
rote:
>>
>> Define such an "alternative communications channel" and explain me how
>> the signaling server cannot alter that channel.
>>
>
> I will send it to you using Frogo, my trusty carrier pidgin ;).
>
> I can read them to you (possible to modify but much harder then simple
> signaling), I can email them to you using some sort of trusted service, o=
r I
> can call you using old fashioned telephone. I remember times when such ke=
ys
> were exchanged via fax.

So the success of DTLS (the "security end-to-end panacea") depends on
two users of any web sharing their "keys" via email, phone, or fax...

Did you forget the <sarcasm> tag? XD

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

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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Jean-Marc,

> Personally, I would be reluctant to indemnify anyone over IPR in a paper
> clip. This is the sad state of the patent system and it applies to every
> piece of technology.

Oh, I agree with you. :-)
 
> > Indeed, but I have a significantly higher level of confidence that I
> > can identify all of the legitimate companies with IPR on H.264,
> > whereas I haven't a clue where to start (outside of Google) for VP8.
> 
> Obviously, that makes H.264 so much safer, right?
> http://www.theregister.co.uk/2012/03/29/microsoft_motorola_patent_cases/

Motorola is a known patent holder.  So if Microsoft.  I assume Microsoft
elected to ignore Motorola's patent claims, and I assume Motorola ignored
Microsoft's patents, too, as I'm sure there would have been a reciprocity
agreement in place otherwise.  I don't know, but if you want to use H.264,
then you would have known to talk to Motorola and MS.
 
> > If I owned IPR on VP8 (which I don't personally; can't speak for my
> > employer), I certainly would not tell you and I would not join a
> > patent pool, either.  I would wait until you adopt VP8, build it into
> > software and hardware products, have it massively deployed, and then
> > I'd come along and collect my royalties.  There is absolutely no
> > financial incentive for an IPR holder to join a patent pool.
> 
> I assume this is why we've seen all these lawsuits against Google,
> Microsoft, Cisco, Apple... over their use of Vorbis and Speex, right?
> Seriously, I can count at least 10 free AV codecs and none of them have
> had any patent lawsuits that I'm aware of.

The "why" is unknown. It might be that a license was acquired. It might be
that patent holders do not care since the same companies are already
licensing their patents.  Perhaps the usage is not as high as other codecs,
including MP3 and G.729?  The fact there isn't a lawsuit does not mean these
are free from patents, unless the technology is so old patents are expired.
(I think that's one argument for using H.261, in spite of its small video
sizes.)
 
> The best interest of a H.264 patent owner is to avoid having to disclose,
> wait for adoption, and then be free to ask for a lot more than what they
> would get from the patent pool. This is why I consider H.264 to be a
> higher risk than VP8. The current Motorola lawsuit confirms this.

The Motorola lawsuit does not confirm this at all.  This is nothing but a
fight between those two companies, both of which were likely aware of the
others' IPR here.

You're ignoring one point I made.  Those involved in the creation of H.264
have quite likely filed IPR statements with ISO or ITU.  If some other
company were to come out of the closet on this with something that
blindsides the industry, the industry would fight it.  At the very least,
whatever bit of IPR that is claimed would have to be put into perspective
with respect to the other hundreds of patents on H.264.

H.264 has already become widely deployed.  Work is well underway on H.265
now.  So, there are patent trolls out there, right now is the time to start
filing lawsuits.  And as I said, the reputable companies would have
disclosed IPR and you'd know to talk to them.

VP8 is a technology where nobody other than Google has to answer about IPR.
BT, Oki, France Telecom, Qualcom, Nokia, Microsoft, Sony, NTT, Motorola, TI,
etc. are not obligated to make a statement, yet it would not surprise me in
the least if these companies do not have IPR.  Whether the enforce it is
another matter.  They will only if it is in their interest.

Paul



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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Fri, 6 Apr 2012 07:33:41 +1000
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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On Fri, Apr 6, 2012 at 4:50 AM, Paul E. Jones <paulej@packetizer.com> wrote=
:
>> > Indeed, but I have a significantly higher level of confidence that I
>> > can identify all of the legitimate companies with IPR on H.264,
>> > whereas I haven't a clue where to start (outside of Google) for VP8.
>>
>> Obviously, that makes H.264 so much safer, right?
>> http://www.theregister.co.uk/2012/03/29/microsoft_motorola_patent_cases/
>
> Motorola is a known patent holder. =A0So if Microsoft. =A0I assume Micros=
oft
> elected to ignore Motorola's patent claims, and I assume Motorola ignored
> Microsoft's patents, too, as I'm sure there would have been a reciprocity
> agreement in place otherwise. =A0I don't know, but if you want to use H.2=
64,
> then you would have known to talk to Motorola and MS.

So what's the point in having the patent pool if you have to talk to
all the individual patent owners anyway?


>> > If I owned IPR on VP8 (which I don't personally; can't speak for my
>> > employer), I certainly would not tell you and I would not join a
>> > patent pool, either. =A0I would wait until you adopt VP8, build it int=
o
>> > software and hardware products, have it massively deployed, and then
>> > I'd come along and collect my royalties. =A0There is absolutely no
>> > financial incentive for an IPR holder to join a patent pool.
>>
>> I assume this is why we've seen all these lawsuits against Google,
>> Microsoft, Cisco, Apple... over their use of Vorbis and Speex, right?
>> Seriously, I can count at least 10 free AV codecs and none of them have
>> had any patent lawsuits that I'm aware of.
>
> The "why" is unknown. It might be that a license was acquired. It might b=
e
> that patent holders do not care since the same companies are already
> licensing their patents. =A0Perhaps the usage is not as high as other cod=
ecs,
> including MP3 and G.729?

You might be surprised about the usage - a substantial amount of games
use Vorbis because they don't have to pay for a license. Indeed, it is
well known that Microsoft's Halo used Vorbis. For that very reason, I
doubt that any of those companies acquired a patent license from any
company claiming to own a patent on Vorbis or Speex. In fact, since
nobody has claimed to have such patents, how would you even know who
to license it from?

> =A0The fact there isn't a lawsuit does not mean these
> are free from patents, unless the technology is so old patents are expire=
d.
> (I think that's one argument for using H.261, in spite of its small video
> sizes.)

Existing lawsuits are the only quantitative measure that is available
to find out about whether somebody is prepared to fight for their
patents on a technology. Even patent pools don't work for this,
apparently.


>> The best interest of a H.264 patent owner is to avoid having to disclose=
,
>> wait for adoption, and then be free to ask for a lot more than what they
>> would get from the patent pool. This is why I consider H.264 to be a
>> higher risk than VP8. The current Motorola lawsuit confirms this.
>
> The Motorola lawsuit does not confirm this at all. =A0This is nothing but=
 a
> fight between those two companies, both of which were likely aware of the
> others' IPR here.
>
> You're ignoring one point I made. =A0Those involved in the creation of H.=
264
> have quite likely filed IPR statements with ISO or ITU. =A0If some other
> company were to come out of the closet on this with something that
> blindsides the industry, the industry would fight it. =A0At the very leas=
t,
> whatever bit of IPR that is claimed would have to be put into perspective
> with respect to the other hundreds of patents on H.264.

If somebody was to come out of the woodworks over VP8, I'd expect
Google to fight them. And all the other companies that have now
invested into VP8. I don't think patents mean anything when it comes
to the decision of whether to fight a patent claim or not: money
talks.

> H.264 has already become widely deployed. =A0Work is well underway on H.2=
65
> now. =A0So, there are patent trolls out there, right now is the time to s=
tart
> filing lawsuits. =A0And as I said, the reputable companies would have
> disclosed IPR and you'd know to talk to them.
>
> VP8 is a technology where nobody other than Google has to answer about IP=
R.
> BT, Oki, France Telecom, Qualcom, Nokia, Microsoft, Sony, NTT, Motorola, =
TI,
> etc. are not obligated to make a statement, yet it would not surprise me =
in
> the least if these companies do not have IPR. =A0Whether the enforce it i=
s
> another matter. =A0They will only if it is in their interest.

If they've decided to use VP8, they license makes them take choose
sides. There's no need for a patent pool to force them to take sides
(in fact, that doesn't seem to have worked for Motorola and Microsoft
anyway.

Cheers,
Silvia.

From stewe@stewe.org  Thu Apr  5 15:33:37 2012
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From: Stephan Wenger <stewe@stewe.org>
To: Silvia Pfeiffer <silviapfeiffer1@gmail.com>, "Paul E. Jones" <paulej@packetizer.com>
Thread-Topic: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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Hi,
Joining the fun again, so that Paul does not feel lonely...
Albeit at a slower pace, due to day job requirements plus splendid (and
plentiful) Sierra snow :-)
Stephan

On 4.5.2012 14:33 , "Silvia Pfeiffer" <silviapfeiffer1@gmail.com> wrote:

>On Fri, Apr 6, 2012 at 4:50 AM, Paul E. Jones <paulej@packetizer.com>
>wrote:
>>> > Indeed, but I have a significantly higher level of confidence that I
>>> > can identify all of the legitimate companies with IPR on H.264,
>>> > whereas I haven't a clue where to start (outside of Google) for VP8.
>>>
>>> Obviously, that makes H.264 so much safer, right?
>>>=20
>>>http://www.theregister.co.uk/2012/03/29/microsoft_motorola_patent_cases/
>>
>> Motorola is a known patent holder.  So if Microsoft.  I assume Microsoft
>> elected to ignore Motorola's patent claims, and I assume Motorola
>>ignored
>> Microsoft's patents, too, as I'm sure there would have been a
>>reciprocity
>> agreement in place otherwise.  I don't know, but if you want to use
>>H.264,
>> then you would have known to talk to Motorola and MS.
>
>So what's the point in having the patent pool if you have to talk to
>all the individual patent owners anyway?

The answer to this rhetoric question is twofold.  The obvious part is that
it's usually cheaper and easier to pick one license from a pool at known
and published terms than to negotiate a license with all those
rightholders outside of a pool arrangement (which, by the way, is an
option some companies have chosen).  The less obvious part is that the
percentage of patents in a pool reading on a standard is normally
substantial.  For H.264, one estimate I have heard (and that I believe
could well be realistic) is 80%.  As the terms of the pool and the number
of patents therein are known, one can calculate a "market price" for a
patent license reading on H.264.  It's not that easy for a rightholder
outside the pool to ask for royalties substantially higher than what the
pool charges (and get away with it); at least not without raising concerns
about the fairness and reasonableness of said rightholder.  Which one of
the more important aspects of what currently happens in the antitrust
activity going on in Europe in the Moto/MS case.  As I have said in the
meeting: FRAND has recently gotten teeth.  I firmly believe so.

>
>
>>> > If I owned IPR on VP8 (which I don't personally; can't speak for my
>>> > employer), I certainly would not tell you and I would not join a
>>> > patent pool, either.  I would wait until you adopt VP8, build it into
>>> > software and hardware products, have it massively deployed, and then
>>> > I'd come along and collect my royalties.  There is absolutely no
>>> > financial incentive for an IPR holder to join a patent pool.
>>>
>>> I assume this is why we've seen all these lawsuits against Google,
>>> Microsoft, Cisco, Apple... over their use of Vorbis and Speex, right?
>>> Seriously, I can count at least 10 free AV codecs and none of them have
>>> had any patent lawsuits that I'm aware of.
>>
>> The "why" is unknown. It might be that a license was acquired. It might
>>be
>> that patent holders do not care since the same companies are already
>> licensing their patents.  Perhaps the usage is not as high as other
>>codecs,
>> including MP3 and G.729?
>
>You might be surprised about the usage - a substantial amount of games
>use Vorbis because they don't have to pay for a license. Indeed, it is
>well known that Microsoft's Halo used Vorbis. For that very reason, I
>doubt that any of those companies acquired a patent license from any
>company claiming to own a patent on Vorbis or Speex. In fact, since
>nobody has claimed to have such patents, how would you even know who
>to license it from?

The vast majority of patent disputes never see a courtroom and never get
published.  A rightholder sends a letter alleging infringement, the
parties meet, a license is being negotiated (the terms may or may not
include a monetary component), and that's the end of it.  Outside of
certain ideologically restricted circles, pragmatic solutions can be found
rather easily in many cases.

>
>>  The fact there isn't a lawsuit does not mean these
>> are free from patents, unless the technology is so old patents are
>>expired.
>> (I think that's one argument for using H.261, in spite of its small
>>video
>> sizes.)
>
>Existing lawsuits are the only quantitative measure that is available
>to find out about whether somebody is prepared to fight for their
>patents on a technology. Even patent pools don't work for this,
>apparently.
>
>
>>> The best interest of a H.264 patent owner is to avoid having to
>>>disclose,
>>> wait for adoption, and then be free to ask for a lot more than what
>>>they
>>> would get from the patent pool. This is why I consider H.264 to be a
>>> higher risk than VP8. The current Motorola lawsuit confirms this.
>>
>> The Motorola lawsuit does not confirm this at all.  This is nothing but
>>a
>> fight between those two companies, both of which were likely aware of
>>the
>> others' IPR here.

Paul is correct here.  Motorola did not come out of the woods.  The
patents asserted are well known, and at least one of them has been
asserted and litigated before (in an MPEG-2 context) if I recall
correctly.  They were also declared as being available under RAND terms in
both ISO/IEC an the ITU.

>>
>> You're ignoring one point I made.  Those involved in the creation of
>>H.264
>> have quite likely filed IPR statements with ISO or ITU.  If some other
>> company were to come out of the closet on this with something that
>> blindsides the industry, the industry would fight it.  At the very
>>least,
>> whatever bit of IPR that is claimed would have to be put into
>>perspective
>> with respect to the other hundreds of patents on H.264.
>
>If somebody was to come out of the woodworks over VP8, I'd expect
>Google to fight them. And all the other companies that have now
>invested into VP8. I don't think patents mean anything when it comes
>to the decision of whether to fight a patent claim or not: money
>talks.
>
>> H.264 has already become widely deployed.  Work is well underway on
>>H.265
>> now.  So, there are patent trolls out there, right now is the time to
>>start
>> filing lawsuits.  And as I said, the reputable companies would have
>> disclosed IPR and you'd know to talk to them.
>>
>> VP8 is a technology where nobody other than Google has to answer about
>>IPR.
>> BT, Oki, France Telecom, Qualcom, Nokia, Microsoft, Sony, NTT,
>>Motorola, TI,
>> etc. are not obligated to make a statement, yet it would not surprise
>>me in
>> the least if these companies do not have IPR.  Whether the enforce it is
>> another matter.  They will only if it is in their interest.
>
>If they've decided to use VP8, they license makes them take choose
>sides. There's no need for a patent pool to force them to take sides
>(in fact, that doesn't seem to have worked for Motorola and Microsoft
>anyway.



>
>Cheers,
>Silvia.
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



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Silvia,

> > Motorola is a known patent holder. =A0So if Microsoft. =A0I assume
> > Microsoft elected to ignore Motorola's patent claims, and I assume
> > Motorola ignored Microsoft's patents, too, as I'm sure there would
> > have been a reciprocity agreement in place otherwise. =A0I don't =
know,
> > but if you want to use H.264, then you would have known to talk to
> Motorola and MS.
>=20
> So what's the point in having the patent pool if you have to talk to =
all
> the individual patent owners anyway?

Motorola is not part of the pool, I do not believe.  I would assume one
would not have to talk to those that do participate in the pool, but =
that
may not be true, either.  In any case, the pool certainly makes it =
easier to
license the collective set of patents that are listed in the pool.

> > The "why" is unknown. It might be that a license was acquired. It
> > might be that patent holders do not care since the same companies =
are
> > already licensing their patents. =A0Perhaps the usage is not as high =
as
> > other codecs, including MP3 and G.729?
>=20
> You might be surprised about the usage - a substantial amount of games =
use
> Vorbis because they don't have to pay for a license. Indeed, it is =
well
> known that Microsoft's Halo used Vorbis. For that very reason, I doubt
> that any of those companies acquired a patent license from any company
> claiming to own a patent on Vorbis or Speex. In fact, since nobody has
> claimed to have such patents, how would you even know who to license =
it
> from?

One would not.  And, one would still have to pay through the nose,
potentially, for patent infringement.  That's the problem with patents =
and
technology where the patent landscape is entirely unknown.
=20
> > =A0The fact there isn't a lawsuit does not mean these are free from
> > patents, unless the technology is so old patents are expired.
> > (I think that's one argument for using H.261, in spite of its small
> > video
> > sizes.)
>=20
> Existing lawsuits are the only quantitative measure that is available =
to
> find out about whether somebody is prepared to fight for their patents =
on
> a technology. Even patent pools don't work for this, apparently.

Can't agree with that one.  Recall the GIF file format?  Well after it
became widely used, Unisys went around forcing people to pay royalties.  =
How
many years did people believe that was free?
=20
> > You're ignoring one point I made. =A0Those involved in the creation =
of
> > H.264 have quite likely filed IPR statements with ISO or ITU. =A0If =
some
> > other company were to come out of the closet on this with something
> > that blindsides the industry, the industry would fight it. =A0At the
> > very least, whatever bit of IPR that is claimed would have to be put
> > into perspective with respect to the other hundreds of patents on =
H.264.
>=20
> If somebody was to come out of the woodworks over VP8, I'd expect =
Google
> to fight them. And all the other companies that have now invested into
> VP8. I don't think patents mean anything when it comes to the decision =
of
> whether to fight a patent claim or not: money talks.

By "fighting", I mean coming into court with evidence that would =
contradict
the infringed patent.  Google might be able to show that the particular =
way
that the bits of video are manipulated were already public or covered by =
one
of their patents, maybe not.

My feeling -- and that's all it is -- likely has a bunch more patent =
holders
who could demonstrate that their patents covered whatever it is that a
patent troll might try to claim.
=20
> > VP8 is a technology where nobody other than Google has to answer =
about
> IPR.
> > BT, Oki, France Telecom, Qualcom, Nokia, Microsoft, Sony, NTT,
> > Motorola, TI, etc. are not obligated to make a statement, yet it =
would
> > not surprise me in the least if these companies do not have IPR.
> > Whether the enforce it is another matter. =A0They will only if it is =
in
> their interest.
>=20
> If they've decided to use VP8, they license makes them take choose =
sides.
> There's no need for a patent pool to force them to take sides (in =
fact,
> that doesn't seem to have worked for Motorola and Microsoft anyway.

I don't understand your point here.  Perhaps I was not clear.  My point =
was
that these companies have no obligation to report IPR they hold on VP8,
whereas they do have some obligation to report on H.264.

It just comes down to the fact that I can *probably* identify all the =
patent
holders of H.264.  I might miss one, but it's doubtful that a legitimate
company would have not reported IPR by now.  That only leaves trolls to =
deal
with on H.264, and I suspect they would have a hard time making =
successful
claims.

VP8, though, is a wide-open opportunity for patent trolls and legit
companies alike to lay claim after the technology is widely deployed.  =
It
does not mean they will, but it's your gamble. :-)

Paul



From marshall.eubanks@gmail.com  Thu Apr  5 20:37:31 2012
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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On Thu, Apr 5, 2012 at 6:33 PM, Stephan Wenger <stewe@stewe.org> wrote:
> Hi,
> Joining the fun again, so that Paul does not feel lonely...
> Albeit at a slower pace, due to day job requirements plus splendid (and
> plentiful) Sierra snow :-)
> Stephan
>
> On 4.5.2012 14:33 , "Silvia Pfeiffer" <silviapfeiffer1@gmail.com> wrote:
>
>>On Fri, Apr 6, 2012 at 4:50 AM, Paul E. Jones <paulej@packetizer.com>
>>wrote:
>>>> > Indeed, but I have a significantly higher level of confidence that I
>>>> > can identify all of the legitimate companies with IPR on H.264,
>>>> > whereas I haven't a clue where to start (outside of Google) for VP8.
>>>>
>>>> Obviously, that makes H.264 so much safer, right?
>>>>
>>>>http://www.theregister.co.uk/2012/03/29/microsoft_motorola_patent_cases=
/
>>>
>>> Motorola is a known patent holder. =A0So if Microsoft. =A0I assume Micr=
osoft
>>> elected to ignore Motorola's patent claims, and I assume Motorola
>>>ignored
>>> Microsoft's patents, too, as I'm sure there would have been a
>>>reciprocity
>>> agreement in place otherwise. =A0I don't know, but if you want to use
>>>H.264,
>>> then you would have known to talk to Motorola and MS.
>>
>>So what's the point in having the patent pool if you have to talk to
>>all the individual patent owners anyway?
>
> The answer to this rhetoric question is twofold. =A0The obvious part is t=
hat
> it's usually cheaper and easier to pick one license from a pool at known
> and published terms than to negotiate a license with all those
> rightholders outside of a pool arrangement (which, by the way, is an
> option some companies have chosen). =A0The less obvious part is that the
> percentage of patents in a pool reading on a standard is normally
> substantial. =A0For H.264, one estimate I have heard (and that I believe
> could well be realistic) is 80%. =A0As the terms of the pool and the numb=
er
> of patents therein are known, one can calculate a "market price" for a
> patent license reading on H.264. =A0It's not that easy for a rightholder
> outside the pool to ask for royalties substantially higher than what the
> pool charges (and get away with it); at least not without raising concern=
s
> about the fairness and reasonableness of said rightholder. =A0Which one o=
f
> the more important aspects of what currently happens in the antitrust
> activity going on in Europe in the Moto/MS case. =A0As I have said in the
> meeting: FRAND has recently gotten teeth. =A0I firmly believe so.
>

I think that the EC may agree with you :

http://europa.eu/rapid/pressReleasesAction.do?reference=3DIP/12/345&format=
=3DHTML&aged=3D0&language=3DEN&guiLanguage=3Den

"Following complaints by Apple and Microsoft, the Commission will
investigate, in particular, whether by seeking and enforcing
injunctions against Apple's and Microsoft's flagship products such as
iPhone, iPad, Windows and Xbox on the basis of patents it had declared
essential to produce standard-compliant products, Motorola has failed
to honour its irrevocable commitments made to standard setting
organisations. In these commitments, Motorola engaged to license those
standard-essential patents on fair, reasonable and non-discriminatory
(FRAND) terms. The Commission will examine whether Motorola's
behaviour amounts to an abuse of a dominant market position prohibited
by Article 102 of the Treaty on the Functioning of the EU (TFEU)."

I believe that the primary SSO of issue here is ETSI.

Regards
Marshall

>>
>>
>>>> > If I owned IPR on VP8 (which I don't personally; can't speak for my
>>>> > employer), I certainly would not tell you and I would not join a
>>>> > patent pool, either. =A0I would wait until you adopt VP8, build it i=
nto
>>>> > software and hardware products, have it massively deployed, and then
>>>> > I'd come along and collect my royalties. =A0There is absolutely no
>>>> > financial incentive for an IPR holder to join a patent pool.
>>>>
>>>> I assume this is why we've seen all these lawsuits against Google,
>>>> Microsoft, Cisco, Apple... over their use of Vorbis and Speex, right?
>>>> Seriously, I can count at least 10 free AV codecs and none of them hav=
e
>>>> had any patent lawsuits that I'm aware of.
>>>
>>> The "why" is unknown. It might be that a license was acquired. It might
>>>be
>>> that patent holders do not care since the same companies are already
>>> licensing their patents. =A0Perhaps the usage is not as high as other
>>>codecs,
>>> including MP3 and G.729?
>>
>>You might be surprised about the usage - a substantial amount of games
>>use Vorbis because they don't have to pay for a license. Indeed, it is
>>well known that Microsoft's Halo used Vorbis. For that very reason, I
>>doubt that any of those companies acquired a patent license from any
>>company claiming to own a patent on Vorbis or Speex. In fact, since
>>nobody has claimed to have such patents, how would you even know who
>>to license it from?
>
> The vast majority of patent disputes never see a courtroom and never get
> published. =A0A rightholder sends a letter alleging infringement, the
> parties meet, a license is being negotiated (the terms may or may not
> include a monetary component), and that's the end of it. =A0Outside of
> certain ideologically restricted circles, pragmatic solutions can be foun=
d
> rather easily in many cases.
>
>>
>>> =A0The fact there isn't a lawsuit does not mean these
>>> are free from patents, unless the technology is so old patents are
>>>expired.
>>> (I think that's one argument for using H.261, in spite of its small
>>>video
>>> sizes.)
>>
>>Existing lawsuits are the only quantitative measure that is available
>>to find out about whether somebody is prepared to fight for their
>>patents on a technology. Even patent pools don't work for this,
>>apparently.
>>
>>
>>>> The best interest of a H.264 patent owner is to avoid having to
>>>>disclose,
>>>> wait for adoption, and then be free to ask for a lot more than what
>>>>they
>>>> would get from the patent pool. This is why I consider H.264 to be a
>>>> higher risk than VP8. The current Motorola lawsuit confirms this.
>>>
>>> The Motorola lawsuit does not confirm this at all. =A0This is nothing b=
ut
>>>a
>>> fight between those two companies, both of which were likely aware of
>>>the
>>> others' IPR here.
>
> Paul is correct here. =A0Motorola did not come out of the woods. =A0The
> patents asserted are well known, and at least one of them has been
> asserted and litigated before (in an MPEG-2 context) if I recall
> correctly. =A0They were also declared as being available under RAND terms=
 in
> both ISO/IEC an the ITU.
>
>>>
>>> You're ignoring one point I made. =A0Those involved in the creation of
>>>H.264
>>> have quite likely filed IPR statements with ISO or ITU. =A0If some othe=
r
>>> company were to come out of the closet on this with something that
>>> blindsides the industry, the industry would fight it. =A0At the very
>>>least,
>>> whatever bit of IPR that is claimed would have to be put into
>>>perspective
>>> with respect to the other hundreds of patents on H.264.
>>
>>If somebody was to come out of the woodworks over VP8, I'd expect
>>Google to fight them. And all the other companies that have now
>>invested into VP8. I don't think patents mean anything when it comes
>>to the decision of whether to fight a patent claim or not: money
>>talks.
>>
>>> H.264 has already become widely deployed. =A0Work is well underway on
>>>H.265
>>> now. =A0So, there are patent trolls out there, right now is the time to
>>>start
>>> filing lawsuits. =A0And as I said, the reputable companies would have
>>> disclosed IPR and you'd know to talk to them.
>>>
>>> VP8 is a technology where nobody other than Google has to answer about
>>>IPR.
>>> BT, Oki, France Telecom, Qualcom, Nokia, Microsoft, Sony, NTT,
>>>Motorola, TI,
>>> etc. are not obligated to make a statement, yet it would not surprise
>>>me in
>>> the least if these companies do not have IPR. =A0Whether the enforce it=
 is
>>> another matter. =A0They will only if it is in their interest.
>>
>>If they've decided to use VP8, they license makes them take choose
>>sides. There's no need for a patent pool to force them to take sides
>>(in fact, that doesn't seem to have worked for Motorola and Microsoft
>>anyway.
>
>
>
>>
>>Cheers,
>>Silvia.
>>_______________________________________________
>>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

From randell-ietf@jesup.org  Thu Apr  5 22:01:09 2012
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Subject: Re: [rtcweb] DTLS-SRTP with end-to-end security: Short Authentication String
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tl;dr:  I believe SAS should be supported by the browser (or non-browser 
app/device) UI, and I believe that's the current plan though perhaps 
it's not written as a MUST.  I am much more hesitant to endorse 
key-chaining due to usability issues.

On 4/5/2012 1:04 PM, Fabio Pietrosanti (naif) wrote:
> In the RTCWeb security there is a proposed solution, to use SAS:
> http://tools.ietf.org/html/draft-ietf-rtcweb-security-02#section-4.3.2.2
>
> SAS, Short authentication string, allow to achieve end-to-end security
> within the DTLS-SRTP end-to-end encryption protocol.
>
> Unfortunately the author of rtcweb-security described the SAS as a bad
> way to verify fingerprints.
>
> I would like to argue that SAS are instead a valuable, secure mechanism
> to provide end-to-end authentication to an end-to-end encryption system,
> thus realizing end-to-end security.

The reason the document described it that way (and it was a subject of 
discussion at IETF 81 or an interim phone call since then) is that users 
generally don't use or understand SAS.  Yes, extremely 
security-conscious individuals can use it to verify a given connection. 
  The ZRTP key-chaining is somewhat problematic in a modern world where 
destinations are people who may use one of N devices, not hard-coded 
numbers tied to a specific piece of hardware.

ZRTP was designed to lower the bar for encryption, by making it possible 
to ignore SAS initially under the assumption that the first connection 
is safe.  That assumption might not hold.  And the occasional 
key-chaining failure caused by either loss of key-chain info (device 
reset, etc) or by substitution of devices or forwarding of a call will 
cause an ongoing level of false-positives that will weaken any response 
to a real MITM.  If in response to a failure they don't do SAS, but 
instead just accept it, then they are now silently MITM'd until they do SAS.

Fundamentally, though, as stated in the meetings, the ease-of-use bar 
for ZRTP-like SAS+chaining is just too high for non-security-experts (or 
the non-paranoid (whether rightfully non-paranoid or not)).  We could 
run chaining with the current design, and that has been discussed, but 
when I call you from my tablet instead of my PC, the chain won't match 
and the MITM flag would fire.

This is why we explored and suggested to allow for an Idp to allow 
authentication of the other party to a call.  Does this provide the same 
level of security by default as a working PKI infrastructure, or of 
ZRTP+SAS?  No.  But SAS is still supported (I believe this question was 
asked at IETF 83), so the truly security knowledgeable can verify 
end-to-end security anytime they want, while general users can get 
considerably more security (if the Idp is not also their service/app 
provider) than they would with SDES.

Note that if the cert used by an attacker is the result of a hack like 
Diginotar (or government-sponsored spoofing via a captive CA), then the 
user is severely at risk in any scenario due to injection of compromised 
JS code from a supposedly trusted source, doubly so if the app has been 
granted the right to turn on cameras and mics.  SDES is totally broken 
at that point.

> I would like to bring the attention of the WG to provide a MANDATORY
> requirement to enable SAS verification.

I think the question was asked at IETF (paraphrased): will SAS be 
available through the UI in some manner?  Yes.

So I don't think there's any problem with this request.  Chaining may be 
more problematical.

-- 
Randell Jesup
randell-ietf@jesup.org

From pravindran@sonusnet.com  Thu Apr  5 23:23:43 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
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Subject: Re: [rtcweb] End-to-end encryption vs end-to-end authentication	(DTLS-SRTP / SDES-SRTP)
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Fabio,

End-to-end looks misnomer to me as the maximum media security/authenticatio=
n provided shall be between two media hops. The media hops in the RTCWeb ar=
e web-browser. So, the aim of the media security/authentication mechanism h=
as to between two web-browsers. The intermediate elements like http proxy s=
hould not be in a position to perform MITM attack or atleast, web-browser s=
hould have the mechanism to perform post-mortem analysis that man-in-the-mi=
ddle exist as HTTP proxy. I wish to see the discussion for media security/a=
uthentication between web-browser to web-browser first.

Please note that web-browser is not required to standard browser like Chrom=
e, IE, Mozilla but web-browser shall be custom-made which is RTCWeb complia=
nt.

AFAIK, SDES-SRTP does not provide web-browser to web-browser authentication=
 and it shall mimic web-browser to web-browser security. I agree with Hadri=
el that SDES-SRTP mechanism provides marketing advance as most of the user =
atleast believes that the session is secured :-) but it should not be crite=
ria for selecting key mechanism.  We shall discuss in detail about DTLS-SRT=
P or zRTP or any other mechanism if exists for web-browser to web-browser m=
edia security.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Fabio Pietrosanti (naif)
>Sent: Thursday, April 05, 2012 3:47 PM
>To: rtcweb@ietf.org
>Subject: [rtcweb] End-to-end encryption vs end-to-end authentication
>(DTLS-SRTP / SDES-SRTP)
>
>Hi,
>
>i've been discussing with several friends about the current discussion
>on the security standard in this mailing lists, in particular regarding
>the topic of DTLS-SRTP trust model posted there
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg04007.html .
>
>I found that there is no common definition for the concept of "end-to-
>end encryption" and there are a lot of misunderstanding about it.
>
>In particular, the fact that two peer can community by exchanging keys
>directly among them, tend to typically to be defined as end-to-end
>encryption.
>
>However the term "end-to-end encryption" have also a general
>misconception that's something that "no one can intercept" .
>
>Unfortunately this is not true for DTLS-SRTP, while for example it's
>true for OpenPGP/MIME or for ZRTP with SAS.
>
>We need to separate the 4 concepts:
>
>- end-to-end encryption
>Ability for a key management system to exchange keys directly without
>relying on intermediate server actively involved in encryption process.
>
>- end-to-end authentication
>Ability for end-user to authenticate the end-to-end encryption process
>without the need to rely on a single or multiple set of trusted third
>parties
>
>- end-to-site encryption
>Ability for a key management system to exchange keys with/trough the
>server with which data are exchanged, involving it in the authentication
>process.
>
>- end-to-site authentication
>Ability for end-user to authenticate the end-to-site encryption process
>based on the same security mechanism used to establish trust with the
>server with which data are exchanged.
>
>
>What i would like to focus is that:
>
>- DTLS does provide end-to-end encryption (in specific context of use)
>- DTLS does provide end-to-site authentication (rely on third party
>trust)
>
>- ZRTP does provide end-to-end encryption (in all context of use)
>- ZRTP does provide end-to-end authentication (does not rely on third
>party trust)
>
>- SDES-SRTP does provide end-to-site encryption (encryption with/trough
>your server)
>- SDES-SRTP does provide end-to-site authentication (you trust your
>server involved in key exchange)
>
>So when we tend to think about the "how much security a technology
>provide" we should probably compare in a scale:
>
>- ZRTP
>  - end-to-end encryption
>  - end-to-end authentication
>- DTLS-SRTP
>  - end-to-end encryption
>  - end-to-site authentication
>- SDES-SRTP
>  - end-to-site encryption
>  - end-to-site authentication
>
>So currently we can affirm that:
>
>- ZRTP does not rely on third party trust for effective security
>- DTLS-SRTP rely on third party trust for effective security
>- SDES-SRTP rely on third party trust for effective security
>
>This is the *MOST IMPORTANT* distinction for an encryption technology:
>			WHO SHOULD I TRUST?
>
>Well, basically it seems to me that DTLS-SRTP and SDES-SRTP both require
>
>So, my point are to:
>
>- Introduce SDES-SRTP for compatibility and simplicity
>  - Specify that the Browser will need to provide indicate the security
>level to the user (like the lock of HTTPS, same security model)
>
>- Introduce end-to-end authentication support for DTLS-SRTP via SAS
>  - Specify that the browser will need to to provide the end-user way to
>use end-to-end authentication and indicate the security level to the
>user.
>
>Then it will be up to the signaling server and/or to the browser
>settings to specificy the required security model:
>- end-to-end encryption + end-to-end authentication or
>- end-to-site encryption + end-to-site authentication
>
>But please don't mix those two as it will be *Very difficult, near to
>impossible* to explain to the user "WHO SHOULD HE TRUST" .
>
>Fabio
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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From: Justin Uberti <juberti@google.com>
Date: Fri, 6 Apr 2012 12:12:54 -0400
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Subject: Re: [rtcweb] DTLS-SRTP with end-to-end security: Short Authentication String
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--20cf306f72ee077b4d04bd04efe2
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On Fri, Apr 6, 2012 at 12:57 AM, Randell Jesup <randell-ietf@jesup.org>wrote:

> tl;dr:  I believe SAS should be supported by the browser (or non-browser
> app/device) UI, and I believe that's the current plan though perhaps it's
> not written as a MUST.  I am much more hesitant to endorse key-chaining due
> to usability issues.


+1 for browsers presenting the digest information to the user on demand, so
that they can perform a manual verification step, if desired.


>
>
> On 4/5/2012 1:04 PM, Fabio Pietrosanti (naif) wrote:
>
>> In the RTCWeb security there is a proposed solution, to use SAS:
>> http://tools.ietf.org/html/**draft-ietf-rtcweb-security-02#**
>> section-4.3.2.2<http://tools.ietf.org/html/draft-ietf-rtcweb-security-02#section-4.3.2.2>
>>
>> SAS, Short authentication string, allow to achieve end-to-end security
>> within the DTLS-SRTP end-to-end encryption protocol.
>>
>> Unfortunately the author of rtcweb-security described the SAS as a bad
>> way to verify fingerprints.
>>
>> I would like to argue that SAS are instead a valuable, secure mechanism
>> to provide end-to-end authentication to an end-to-end encryption system,
>> thus realizing end-to-end security.
>>
>
> The reason the document described it that way (and it was a subject of
> discussion at IETF 81 or an interim phone call since then) is that users
> generally don't use or understand SAS.  Yes, extremely security-conscious
> individuals can use it to verify a given connection.  The ZRTP key-chaining
> is somewhat problematic in a modern world where destinations are people who
> may use one of N devices, not hard-coded numbers tied to a specific piece
> of hardware.
>
> ZRTP was designed to lower the bar for encryption, by making it possible
> to ignore SAS initially under the assumption that the first connection is
> safe.  That assumption might not hold.  And the occasional key-chaining
> failure caused by either loss of key-chain info (device reset, etc) or by
> substitution of devices or forwarding of a call will cause an ongoing level
> of false-positives that will weaken any response to a real MITM.  If in
> response to a failure they don't do SAS, but instead just accept it, then
> they are now silently MITM'd until they do SAS.
>
> Fundamentally, though, as stated in the meetings, the ease-of-use bar for
> ZRTP-like SAS+chaining is just too high for non-security-experts (or the
> non-paranoid (whether rightfully non-paranoid or not)).  We could run
> chaining with the current design, and that has been discussed, but when I
> call you from my tablet instead of my PC, the chain won't match and the
> MITM flag would fire.
>
> This is why we explored and suggested to allow for an Idp to allow
> authentication of the other party to a call.  Does this provide the same
> level of security by default as a working PKI infrastructure, or of
> ZRTP+SAS?  No.  But SAS is still supported (I believe this question was
> asked at IETF 83), so the truly security knowledgeable can verify
> end-to-end security anytime they want, while general users can get
> considerably more security (if the Idp is not also their service/app
> provider) than they would with SDES.
>
> Note that if the cert used by an attacker is the result of a hack like
> Diginotar (or government-sponsored spoofing via a captive CA), then the
> user is severely at risk in any scenario due to injection of compromised JS
> code from a supposedly trusted source, doubly so if the app has been
> granted the right to turn on cameras and mics.  SDES is totally broken at
> that point.
>
>
>  I would like to bring the attention of the WG to provide a MANDATORY
>> requirement to enable SAS verification.
>>
>
> I think the question was asked at IETF (paraphrased): will SAS be
> available through the UI in some manner?  Yes.
>
> So I don't think there's any problem with this request.  Chaining may be
> more problematical.
>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--20cf306f72ee077b4d04bd04efe2
Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Fri, Apr 6, 2012 at 12:57 AM, Randell=
 Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf@jesup.org">rand=
ell-ietf@jesup.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

tl;dr: =C2=A0I believe SAS should be supported by the browser (or non-brows=
er app/device) UI, and I believe that&#39;s the current plan though perhaps=
 it&#39;s not written as a MUST. =C2=A0I am much more hesitant to endorse k=
ey-chaining due to usability issues.</blockquote>

<div><br></div><div>+1 for browsers presenting the digest information to th=
e user on demand, so that they can perform a manual verification step, if d=
esired.=C2=A0</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div class=3D"im"><br>
<br>
On 4/5/2012 1:04 PM, Fabio Pietrosanti (naif) wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
In the RTCWeb security there is a proposed solution, to use SAS:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-security-02#section=
-4.3.2.2" target=3D"_blank">http://tools.ietf.org/html/<u></u>draft-ietf-rt=
cweb-security-02#<u></u>section-4.3.2.2</a><br>
<br>
SAS, Short authentication string, allow to achieve end-to-end security<br>
within the DTLS-SRTP end-to-end encryption protocol.<br>
<br>
Unfortunately the author of rtcweb-security described the SAS as a bad<br>
way to verify fingerprints.<br>
<br>
I would like to argue that SAS are instead a valuable, secure mechanism<br>
to provide end-to-end authentication to an end-to-end encryption system,<br=
>
thus realizing end-to-end security.<br>
</blockquote>
<br></div>
The reason the document described it that way (and it was a subject of disc=
ussion at IETF 81 or an interim phone call since then) is that users genera=
lly don&#39;t use or understand SAS. =C2=A0Yes, extremely security-consciou=
s individuals can use it to verify a given connection. =C2=A0The ZRTP key-c=
haining is somewhat problematic in a modern world where destinations are pe=
ople who may use one of N devices, not hard-coded numbers tied to a specifi=
c piece of hardware.<br>


<br>
ZRTP was designed to lower the bar for encryption, by making it possible to=
 ignore SAS initially under the assumption that the first connection is saf=
e. =C2=A0That assumption might not hold. =C2=A0And the occasional key-chain=
ing failure caused by either loss of key-chain info (device reset, etc) or =
by substitution of devices or forwarding of a call will cause an ongoing le=
vel of false-positives that will weaken any response to a real MITM. =C2=A0=
If in response to a failure they don&#39;t do SAS, but instead just accept =
it, then they are now silently MITM&#39;d until they do SAS.<br>


<br>
Fundamentally, though, as stated in the meetings, the ease-of-use bar for Z=
RTP-like SAS+chaining is just too high for non-security-experts (or the non=
-paranoid (whether rightfully non-paranoid or not)). =C2=A0We could run cha=
ining with the current design, and that has been discussed, but when I call=
 you from my tablet instead of my PC, the chain won&#39;t match and the MIT=
M flag would fire.<br>


<br>
This is why we explored and suggested to allow for an Idp to allow authenti=
cation of the other party to a call. =C2=A0Does this provide the same level=
 of security by default as a working PKI infrastructure, or of ZRTP+SAS? =
=C2=A0No. =C2=A0But SAS is still supported (I believe this question was ask=
ed at IETF 83), so the truly security knowledgeable can verify end-to-end s=
ecurity anytime they want, while general users can get considerably more se=
curity (if the Idp is not also their service/app provider) than they would =
with SDES.<br>


<br>
Note that if the cert used by an attacker is the result of a hack like Digi=
notar (or government-sponsored spoofing via a captive CA), then the user is=
 severely at risk in any scenario due to injection of compromised JS code f=
rom a supposedly trusted source, doubly so if the app has been granted the =
right to turn on cameras and mics. =C2=A0SDES is totally broken at that poi=
nt.<div class=3D"im">

<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I would like to bring the attention of the WG to provide a MANDATORY<br>
requirement to enable SAS verification.<br>
</blockquote>
<br></div>
I think the question was asked at IETF (paraphrased): will SAS be available=
 through the UI in some manner? =C2=A0Yes.<br>
<br>
So I don&#39;t think there&#39;s any problem with this request. =C2=A0Chain=
ing may be more problematical.<span class=3D"HOEnZb"><font color=3D"#888888=
"><br>
<br>
-- <br>
Randell Jesup<br>
<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@je=
sup.org</a></font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
______________________________<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>

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From harald@alvestrand.no  Mon Apr  9 04:02:41 2012
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Date: Mon, 09 Apr 2012 13:02:24 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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On 04/02/2012 07:22 PM, Randell Jesup wrote:
> So the question for me is "what scenarios/use-cases does this enable?"
>
> 1) Mesh Conferencing
>
> While there are many ways to handle conferencing, if you have a mesh 
> (full or partial), you need to create new PeerConnections for existing 
> members of the mesh when someone joins.  There are two ways to do this:
> a) somehow application-specific tell the new member who to send 
> invites to, and then start N PeerConnection calls with some "I'm in 
> the conference" token;
> b)have them send N separate generic invites to the service provider 
> which either forwards it to an existing member, or returns it with 
> "it's ok, I've already forwarded invites for it);
> c) somehow tell all the existing members to invite the new member; or
> d) have the service provider fork the OFFER from the new member to all 
> the existing members, and have all of them reply with ANSWERs.  This 
> requires PeerConnection cloning to work well.
>
> I strongly prefer d), and it also reduces conference join time (less 
> round-trips, at least in theory - perhaps noticeably less by sharing 
> the ICE candidate generation).

Hmm.... for one new member, I don't think d) is actually different from 
the case where the client sends an "extra" OFFER when he joins, and the 
server uses the "hanging OFFER" to signal the new member.
This is a windowing protocol, though - the client would need to send a 
new "hanging OFFER" after the current "hanging OFFER" has been "used 
up". Cloning allows you to connect multiple new conference mebers as 
fast as the Javascript can accept them. (Which may have issues with 
congestion, but that might be a price worth paying).

It's not clear to me whether hanging OFFERs work with NAT traversal; 
would the ICE machine send keepalives for an association that hasn't 
been associated yet?

                    Harald


From ekr@rtfm.com  Mon Apr  9 06:58:23 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 9 Apr 2012 06:57:39 -0700
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Subject: [rtcweb] Data on travel times
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Hi folks,

Since it seems like we're going to be having a large number of
interims, I thought it might be instructive to try to analyze a bunch
of different locations to figure out the best strategy. My first cut
analysis is below.

Note that I'm not trying to make any claims about what the best set of
venues is. It's obviously easy to figure out any statistic we want
about each proposed venue, but how you map that data to "best" is up
to you. In particular, there's some tradeoff between minimal total
travel time and a "fair" distribution of travel times (not that I
claim to know what that means).


METHODOLOGY
The data below is derived by treating both people and venues as
airport locations and using travel time as our primary instrument.

1. For each responder for the current Doodle poll, assign a home
   airport based on their draft publication history.  We're missing a
   few people but basically it should be pretty complete. Since
   these people responded before the venue is known, it's at
   least somewhat unbiased.

2. Compute the shortest advertised flight between each home airport
   and the locations for each venue by looking at the shortest
   advertised Kayak flights around one of the proposed interim
   dates (6/10 - 6/13), ignoring price, but excluding "Hacker fares".
   [Thanks to Martin Thomson or helping me gather these.]

This lets us compute statistics for any venue and/or combination
of venues, based on the candidate attendee list.

The three proposed venues:

- San Francisco (SFO)
- Boston (BOS)
- Stockholm (ARN)

Three hubs not too distant from the proposed venues:

- London (LHR)
- Frankfurt (FRA)
- New York (NYC) [0]

Also, Calgary (YYC), since the other two chair locations (BOS and SFO)
were already proposed as venues, and I didn't want Cullen to feel
left out.


RESULTS
Here are the results for each of the above venues, measured in total
hours of travel (i.e., round trip).

Venue         Mean         Median           SD
----------------------------------------------
SFO           13.5             11         12.2
BOS           12.3             11          7.5
ARN           17.0             21         10.7
FRA           14.8             17          7.3
LHR           13.3             14          7.5
NYC           11.5             11          5.8
YYC           14.9             13         10.2
SFO/BOS/ARN   14.3             13          3.6
SFO/NYC/LHR   12.7             11.3        3.7

XXX/YYY/ZZZ a three-way rotation of XXX, YYY, and ZZZ. Obviously, mean
and median are intended to be some sort of aggregate measure of travel
time. I don't have any way to measure "fairness", but SD is intended
as some metric of the variation in travel time between attendees.

The raw data and software are attached. The files are:

  home-airports     -- the list of people's home airports
  durations.txt     -- the list of airport-airport durations
  doodle.txt        -- the attendees list
  pairings.py       -- the software to compute travel times
  doodle-out.txt -- the computed travel times for each attendee

Obviously, there could be an error in the raw data or the software.
Please feel free to send corrections, especially if you find
something material.


OBSERVATIONS
Obviously, it's hard to know what the optimal solution is without
some model for optimality, but we can still make some observations
based on this data:

1. If we're just concerned with minimizing total travel time, then we
would always in New York, since it has both the shortest mean travel
time and the shortest median travel time, but as I said above, this
arguably isn't fair to people who live either in Europe or California,
since they always have to travel.

2. Combining West Coast, East Coast, and European venues has
comparable (or at least not too much worse) mean/median values than
NYC with much lower SDs. So, arguably that kind of mix is more fair.

3. There's a pretty substantial difference between hub and non-hub
venues. In particular, LHR has a median travel time 7 hours less than
ARN, and the SFO/NYC/LHR combination has a median/mean travel time
about 2 hours less than SFO/BOS/ARN (primarily accounted for by the
LHR/ARN difference). [Full disclosure, I've favored Star Alliance hubs
here, but you'd probably get similar results if, for instance, you
used AMS instead of LHR.]


Obviously, your mileage may vary based on your location and feelings
about what's fair, but based on this data, it looks to me like a
three-way rotation between West Coast, East Coast, and European hubs
offers a good compromise between minimum cost and a flat distribution
of travel times.

Personally, whatever we decide to do I'd ask that the WG settle now on
a pattern going forward so that we can predictably budget our travel
time and dollars.


[0] Treating all three NYC airports as a single location.

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From randell-ietf@jesup.org  Mon Apr  9 07:23:16 2012
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Subject: Re: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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On 4/9/2012 7:02 AM, Harald Alvestrand wrote:
> On 04/02/2012 07:22 PM, Randell Jesup wrote:
>> So the question for me is "what scenarios/use-cases does this enable?"
>>
>> 1) Mesh Conferencing
>>
>> While there are many ways to handle conferencing, if you have a mesh
>> (full or partial), you need to create new PeerConnections for existing
>> members of the mesh when someone joins. There are two ways to do this:
>> a) somehow application-specific tell the new member who to send
>> invites to, and then start N PeerConnection calls with some "I'm in
>> the conference" token;
>> b)have them send N separate generic invites to the service provider
>> which either forwards it to an existing member, or returns it with
>> "it's ok, I've already forwarded invites for it);
>> c) somehow tell all the existing members to invite the new member; or
>> d) have the service provider fork the OFFER from the new member to all
>> the existing members, and have all of them reply with ANSWERs. This
>> requires PeerConnection cloning to work well.
>>
>> I strongly prefer d), and it also reduces conference join time (less
>> round-trips, at least in theory - perhaps noticeably less by sharing
>> the ICE candidate generation).
>
> Hmm.... for one new member, I don't think d) is actually different from
> the case where the client sends an "extra" OFFER when he joins, and the
> server uses the "hanging OFFER" to signal the new member.
> This is a windowing protocol, though - the client would need to send a
> new "hanging OFFER" after the current "hanging OFFER" has been "used
> up".

Right, and there would be complication when a new "hanging offer" hadn't 
yet been sent when another member joins (and it would be very easy for 
bugs to creep in there, as it's evil race-condition testing).  "hanging 
offers" is not a protocol that I would like to implement.

> Cloning allows you to connect multiple new conference mebers as
> fast as the Javascript can accept them. (Which may have issues with
> congestion, but that might be a price worth paying).

And removes a whole passel of error paths and handling.

> It's not clear to me whether hanging OFFERs work with NAT traversal;
> would the ICE machine send keepalives for an association that hasn't
> been associated yet?

This certainly needs investigation; I could see it needing some type of 
occasional "still holding it" notification from the server anyways to 
keep the state active in the client.


-- 
Randell Jesup
randell-ietf@jesup.org

From marshall.eubanks@gmail.com  Mon Apr  9 08:36:01 2012
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Date: Mon, 9 Apr 2012 11:35:59 -0400
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From: Marshall Eubanks <marshall.eubanks@gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Data on travel times
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I really like this analysis. Some questions.

2012/4/9 Eric Rescorla <ekr@rtfm.com>:
> Hi folks,
>
> Since it seems like we're going to be having a large number of
> interims, I thought it might be instructive to try to analyze a bunch
> of different locations to figure out the best strategy. My first cut
> analysis is below.
>
> Note that I'm not trying to make any claims about what the best set of
> venues is. It's obviously easy to figure out any statistic we want
> about each proposed venue, but how you map that data to "best" is up
> to you. In particular, there's some tradeoff between minimal total
> travel time and a "fair" distribution of travel times (not that I
> claim to know what that means).
>
>
> METHODOLOGY
> The data below is derived by treating both people and venues as
> airport locations and using travel time as our primary instrument.
>
> 1. For each responder for the current Doodle poll, assign a home
> =A0 airport based on their draft publication history. =A0We're missing a
> =A0 few people but basically it should be pretty complete. Since
> =A0 these people responded before the venue is known, it's at
> =A0 least somewhat unbiased.
>
> 2. Compute the shortest advertised flight between each home airport
> =A0 and the locations for each venue by looking at the shortest
> =A0 advertised Kayak flights around one of the proposed interim
> =A0 dates (6/10 - 6/13), ignoring price, but excluding "Hacker fares".
> =A0 [Thanks to Martin Thomson or helping me gather these.]
>

1.) Why are some fields doubled ? I.e.,

ARN SFO 14 13

Are these counted twice ? That would, of course, give more weight to
those records.

2.) At any rate, I couldn't quite match your numbers. For SFO, for
example, I got

# SFO

 Records            29  |
 Mean            12.52  |
 RMS             15.34  |
 Std Dev          8.55  |
 Minimum          1.00  |
 Maximum         34.00  |

This assumes that each doubled entry counts as 2 separate entries. If
the second entries are ignored, I get

# SFO

 Records            21  |
 Mean            14.05  |
 RMS             17.05  |
 Std Dev          9.14  |
 Minimum          1.00  |
 Maximum         34.00  |

If two entries are averaged together (when present)

# SFO
 Records            21  |
 Mean            13.93  |
 RMS             16.97  |
 Std Dev          9.18  |
 Minimum          1.00  |
 Maximum         34.00  |

None of these 3 options match your

Venue         Mean         Median           SD
----------------------------------------------
SFO           13.5             11         12.2

In particular, your SD value seems high.

(Note, I use the SD =3D root mean square /(n-1) not / n convention, but
that won't explain the difference. )

Regards
Marshall


> This lets us compute statistics for any venue and/or combination
> of venues, based on the candidate attendee list.
>
> The three proposed venues:
>
> - San Francisco (SFO)
> - Boston (BOS)
> - Stockholm (ARN)
>
> Three hubs not too distant from the proposed venues:
>
> - London (LHR)
> - Frankfurt (FRA)
> - New York (NYC) [0]
>
> Also, Calgary (YYC), since the other two chair locations (BOS and SFO)
> were already proposed as venues, and I didn't want Cullen to feel
> left out.
>
>
> RESULTS
> Here are the results for each of the above venues, measured in total
> hours of travel (i.e., round trip).
>
> Venue =A0 =A0 =A0 =A0 Mean =A0 =A0 =A0 =A0 Median =A0 =A0 =A0 =A0 =A0 SD
> ----------------------------------------------
> SFO =A0 =A0 =A0 =A0 =A0 13.5 =A0 =A0 =A0 =A0 =A0 =A0 11 =A0 =A0 =A0 =A0 1=
2.2
> BOS =A0 =A0 =A0 =A0 =A0 12.3 =A0 =A0 =A0 =A0 =A0 =A0 11 =A0 =A0 =A0 =A0 =
=A07.5
> ARN =A0 =A0 =A0 =A0 =A0 17.0 =A0 =A0 =A0 =A0 =A0 =A0 21 =A0 =A0 =A0 =A0 1=
0.7
> FRA =A0 =A0 =A0 =A0 =A0 14.8 =A0 =A0 =A0 =A0 =A0 =A0 17 =A0 =A0 =A0 =A0 =
=A07.3
> LHR =A0 =A0 =A0 =A0 =A0 13.3 =A0 =A0 =A0 =A0 =A0 =A0 14 =A0 =A0 =A0 =A0 =
=A07.5
> NYC =A0 =A0 =A0 =A0 =A0 11.5 =A0 =A0 =A0 =A0 =A0 =A0 11 =A0 =A0 =A0 =A0 =
=A05.8
> YYC =A0 =A0 =A0 =A0 =A0 14.9 =A0 =A0 =A0 =A0 =A0 =A0 13 =A0 =A0 =A0 =A0 1=
0.2
> SFO/BOS/ARN =A0 14.3 =A0 =A0 =A0 =A0 =A0 =A0 13 =A0 =A0 =A0 =A0 =A03.6
> SFO/NYC/LHR =A0 12.7 =A0 =A0 =A0 =A0 =A0 =A0 11.3 =A0 =A0 =A0 =A03.7
>
> XXX/YYY/ZZZ a three-way rotation of XXX, YYY, and ZZZ. Obviously, mean
> and median are intended to be some sort of aggregate measure of travel
> time. I don't have any way to measure "fairness", but SD is intended
> as some metric of the variation in travel time between attendees.
>
> The raw data and software are attached. The files are:
>
> =A0home-airports =A0 =A0 -- the list of people's home airports
> =A0durations.txt =A0 =A0 -- the list of airport-airport durations
> =A0doodle.txt =A0 =A0 =A0 =A0-- the attendees list
> =A0pairings.py =A0 =A0 =A0 -- the software to compute travel times
> =A0doodle-out.txt -- the computed travel times for each attendee
>
> Obviously, there could be an error in the raw data or the software.
> Please feel free to send corrections, especially if you find
> something material.
>
>
> OBSERVATIONS
> Obviously, it's hard to know what the optimal solution is without
> some model for optimality, but we can still make some observations
> based on this data:
>
> 1. If we're just concerned with minimizing total travel time, then we
> would always in New York, since it has both the shortest mean travel
> time and the shortest median travel time, but as I said above, this
> arguably isn't fair to people who live either in Europe or California,
> since they always have to travel.
>
> 2. Combining West Coast, East Coast, and European venues has
> comparable (or at least not too much worse) mean/median values than
> NYC with much lower SDs. So, arguably that kind of mix is more fair.
>
> 3. There's a pretty substantial difference between hub and non-hub
> venues. In particular, LHR has a median travel time 7 hours less than
> ARN, and the SFO/NYC/LHR combination has a median/mean travel time
> about 2 hours less than SFO/BOS/ARN (primarily accounted for by the
> LHR/ARN difference). [Full disclosure, I've favored Star Alliance hubs
> here, but you'd probably get similar results if, for instance, you
> used AMS instead of LHR.]
>
>
> Obviously, your mileage may vary based on your location and feelings
> about what's fair, but based on this data, it looks to me like a
> three-way rotation between West Coast, East Coast, and European hubs
> offers a good compromise between minimum cost and a flat distribution
> of travel times.
>
> Personally, whatever we decide to do I'd ask that the WG settle now on
> a pattern going forward so that we can predictably budget our travel
> time and dollars.
>
>
> [0] Treating all three NYC airports as a single location.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From ekr@rtfm.com  Mon Apr  9 09:03:22 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 9 Apr 2012 09:02:41 -0700
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Subject: Re: [rtcweb] Data on travel times
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On Mon, Apr 9, 2012 at 8:35 AM, Marshall Eubanks
<marshall.eubanks@gmail.com> wrote:
> I really like this analysis. Some questions.
>
> 2012/4/9 Eric Rescorla <ekr@rtfm.com>:
>> Hi folks,
>>
>> Since it seems like we're going to be having a large number of
>> interims, I thought it might be instructive to try to analyze a bunch
>> of different locations to figure out the best strategy. My first cut
>> analysis is below.
>>
>> Note that I'm not trying to make any claims about what the best set of
>> venues is. It's obviously easy to figure out any statistic we want
>> about each proposed venue, but how you map that data to "best" is up
>> to you. In particular, there's some tradeoff between minimal total
>> travel time and a "fair" distribution of travel times (not that I
>> claim to know what that means).
>>
>>
>> METHODOLOGY
>> The data below is derived by treating both people and venues as
>> airport locations and using travel time as our primary instrument.
>>
>> 1. For each responder for the current Doodle poll, assign a home
>> =A0 airport based on their draft publication history. =A0We're missing a
>> =A0 few people but basically it should be pretty complete. Since
>> =A0 these people responded before the venue is known, it's at
>> =A0 least somewhat unbiased.
>>
>> 2. Compute the shortest advertised flight between each home airport
>> =A0 and the locations for each venue by looking at the shortest
>> =A0 advertised Kayak flights around one of the proposed interim
>> =A0 dates (6/10 - 6/13), ignoring price, but excluding "Hacker fares".
>> =A0 [Thanks to Martin Thomson or helping me gather these.]
>>
>
> 1.) Why are some fields doubled ? I.e.,
>
> ARN SFO 14 13
>
> Are these counted twice ? That would, of course, give more weight to
> those records.

Laziness. When I started recording flight times, I used the total time
and then later realized that what I wanted was to break them out by
out and back, but I was too lazy to go back and fix the earlier ones.


> 2.) At any rate, I couldn't quite match your numbers. For SFO, for
> example, I got
>
> # SFO
>
> =A0Records =A0 =A0 =A0 =A0 =A0 =A029 =A0|
> =A0Mean =A0 =A0 =A0 =A0 =A0 =A012.52 =A0|
> =A0RMS =A0 =A0 =A0 =A0 =A0 =A0 15.34 =A0|
> =A0Std Dev =A0 =A0 =A0 =A0 =A08.55 =A0|
> =A0Minimum =A0 =A0 =A0 =A0 =A01.00 =A0|
> =A0Maximum =A0 =A0 =A0 =A0 34.00 =A0|
>
> This assumes that each doubled entry counts as 2 separate entries. If
> the second entries are ignored, I get

I'm not sure what procedure you are following here, but if it's taking the
SD of the data in durations.txt, that's not what I did. That's just
the input data.

The summary data that I am showing is produced by weighting by
participant from each home airport. The script to generate that is
pairings.py and the results are found in doodle-out.txt. Of course,
it could still all be wrong.

FWIW, I'm using R's sd() which uses n-1.

-Ekr



> # SFO
>
> =A0Records =A0 =A0 =A0 =A0 =A0 =A021 =A0|
> =A0Mean =A0 =A0 =A0 =A0 =A0 =A014.05 =A0|
> =A0RMS =A0 =A0 =A0 =A0 =A0 =A0 17.05 =A0|
> =A0Std Dev =A0 =A0 =A0 =A0 =A09.14 =A0|
> =A0Minimum =A0 =A0 =A0 =A0 =A01.00 =A0|
> =A0Maximum =A0 =A0 =A0 =A0 34.00 =A0|
>
> If two entries are averaged together (when present)
>
> # SFO
> =A0Records =A0 =A0 =A0 =A0 =A0 =A021 =A0|
> =A0Mean =A0 =A0 =A0 =A0 =A0 =A013.93 =A0|
> =A0RMS =A0 =A0 =A0 =A0 =A0 =A0 16.97 =A0|
> =A0Std Dev =A0 =A0 =A0 =A0 =A09.18 =A0|
> =A0Minimum =A0 =A0 =A0 =A0 =A01.00 =A0|
> =A0Maximum =A0 =A0 =A0 =A0 34.00 =A0|
>
> None of these 3 options match your
>
> Venue =A0 =A0 =A0 =A0 Mean =A0 =A0 =A0 =A0 Median =A0 =A0 =A0 =A0 =A0 SD
> ----------------------------------------------
> SFO =A0 =A0 =A0 =A0 =A0 13.5 =A0 =A0 =A0 =A0 =A0 =A0 11 =A0 =A0 =A0 =A0 1=
2.2
>
> In particular, your SD value seems high.
>
> (Note, I use the SD =3D root mean square /(n-1) not / n convention, but
> that won't explain the difference. )
>
> Regards
> Marshall
>
>
>> This lets us compute statistics for any venue and/or combination
>> of venues, based on the candidate attendee list.
>>
>> The three proposed venues:
>>
>> - San Francisco (SFO)
>> - Boston (BOS)
>> - Stockholm (ARN)
>>
>> Three hubs not too distant from the proposed venues:
>>
>> - London (LHR)
>> - Frankfurt (FRA)
>> - New York (NYC) [0]
>>
>> Also, Calgary (YYC), since the other two chair locations (BOS and SFO)
>> were already proposed as venues, and I didn't want Cullen to feel
>> left out.
>>
>>
>> RESULTS
>> Here are the results for each of the above venues, measured in total
>> hours of travel (i.e., round trip).
>>
>> Venue =A0 =A0 =A0 =A0 Mean =A0 =A0 =A0 =A0 Median =A0 =A0 =A0 =A0 =A0 SD
>> ----------------------------------------------
>> SFO =A0 =A0 =A0 =A0 =A0 13.5 =A0 =A0 =A0 =A0 =A0 =A0 11 =A0 =A0 =A0 =A0 =
12.2
>> BOS =A0 =A0 =A0 =A0 =A0 12.3 =A0 =A0 =A0 =A0 =A0 =A0 11 =A0 =A0 =A0 =A0 =
=A07.5
>> ARN =A0 =A0 =A0 =A0 =A0 17.0 =A0 =A0 =A0 =A0 =A0 =A0 21 =A0 =A0 =A0 =A0 =
10.7
>> FRA =A0 =A0 =A0 =A0 =A0 14.8 =A0 =A0 =A0 =A0 =A0 =A0 17 =A0 =A0 =A0 =A0 =
=A07.3
>> LHR =A0 =A0 =A0 =A0 =A0 13.3 =A0 =A0 =A0 =A0 =A0 =A0 14 =A0 =A0 =A0 =A0 =
=A07.5
>> NYC =A0 =A0 =A0 =A0 =A0 11.5 =A0 =A0 =A0 =A0 =A0 =A0 11 =A0 =A0 =A0 =A0 =
=A05.8
>> YYC =A0 =A0 =A0 =A0 =A0 14.9 =A0 =A0 =A0 =A0 =A0 =A0 13 =A0 =A0 =A0 =A0 =
10.2
>> SFO/BOS/ARN =A0 14.3 =A0 =A0 =A0 =A0 =A0 =A0 13 =A0 =A0 =A0 =A0 =A03.6
>> SFO/NYC/LHR =A0 12.7 =A0 =A0 =A0 =A0 =A0 =A0 11.3 =A0 =A0 =A0 =A03.7
>>
>> XXX/YYY/ZZZ a three-way rotation of XXX, YYY, and ZZZ. Obviously, mean
>> and median are intended to be some sort of aggregate measure of travel
>> time. I don't have any way to measure "fairness", but SD is intended
>> as some metric of the variation in travel time between attendees.
>>
>> The raw data and software are attached. The files are:
>>
>> =A0home-airports =A0 =A0 -- the list of people's home airports
>> =A0durations.txt =A0 =A0 -- the list of airport-airport durations
>> =A0doodle.txt =A0 =A0 =A0 =A0-- the attendees list
>> =A0pairings.py =A0 =A0 =A0 -- the software to compute travel times
>> =A0doodle-out.txt -- the computed travel times for each attendee
>>
>> Obviously, there could be an error in the raw data or the software.
>> Please feel free to send corrections, especially if you find
>> something material.
>>
>>
>> OBSERVATIONS
>> Obviously, it's hard to know what the optimal solution is without
>> some model for optimality, but we can still make some observations
>> based on this data:
>>
>> 1. If we're just concerned with minimizing total travel time, then we
>> would always in New York, since it has both the shortest mean travel
>> time and the shortest median travel time, but as I said above, this
>> arguably isn't fair to people who live either in Europe or California,
>> since they always have to travel.
>>
>> 2. Combining West Coast, East Coast, and European venues has
>> comparable (or at least not too much worse) mean/median values than
>> NYC with much lower SDs. So, arguably that kind of mix is more fair.
>>
>> 3. There's a pretty substantial difference between hub and non-hub
>> venues. In particular, LHR has a median travel time 7 hours less than
>> ARN, and the SFO/NYC/LHR combination has a median/mean travel time
>> about 2 hours less than SFO/BOS/ARN (primarily accounted for by the
>> LHR/ARN difference). [Full disclosure, I've favored Star Alliance hubs
>> here, but you'd probably get similar results if, for instance, you
>> used AMS instead of LHR.]
>>
>>
>> Obviously, your mileage may vary based on your location and feelings
>> about what's fair, but based on this data, it looks to me like a
>> three-way rotation between West Coast, East Coast, and European hubs
>> offers a good compromise between minimum cost and a flat distribution
>> of travel times.
>>
>> Personally, whatever we decide to do I'd ask that the WG settle now on
>> a pattern going forward so that we can predictably budget our travel
>> time and dollars.
>>
>>
>> [0] Treating all three NYC airports as a single location.
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>

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This post is the first in the history of standards--for the past 200 or 
so years-- that I have been following where a scientific approach for a 
decision on the meeting venue has been planned and actually carried out.

Bravo, Eric!

Igor


On 4/9/2012 9:57 AM, Eric Rescorla wrote:
> Hi folks,
>
> Since it seems like we're going to be having a large number of
> interims, I thought it might be instructive to try to analyze a bunch
> of different locations to figure out the best strategy. My first cut
> analysis is below.
>
> Note that I'm not trying to make any claims about what the best set of
> venues is. It's obviously easy to figure out any statistic we want
> about each proposed venue, but how you map that data to "best" is up
> to you. In particular, there's some tradeoff between minimal total
> travel time and a "fair" distribution of travel times (not that I
> claim to know what that means).
>
>
> METHODOLOGY
> The data below is derived by treating both people and venues as
> airport locations and using travel time as our primary instrument.
>
> 1. For each responder for the current Doodle poll, assign a home
>     airport based on their draft publication history.  We're missing a
>     few people but basically it should be pretty complete. Since
>     these people responded before the venue is known, it's at
>     least somewhat unbiased.
>
> 2. Compute the shortest advertised flight between each home airport
>     and the locations for each venue by looking at the shortest
>     advertised Kayak flights around one of the proposed interim
>     dates (6/10 - 6/13), ignoring price, but excluding "Hacker fares".
>     [Thanks to Martin Thomson or helping me gather these.]
>
> This lets us compute statistics for any venue and/or combination
> of venues, based on the candidate attendee list.
>
> The three proposed venues:
>
> - San Francisco (SFO)
> - Boston (BOS)
> - Stockholm (ARN)
>
> Three hubs not too distant from the proposed venues:
>
> - London (LHR)
> - Frankfurt (FRA)
> - New York (NYC) [0]
>
> Also, Calgary (YYC), since the other two chair locations (BOS and SFO)
> were already proposed as venues, and I didn't want Cullen to feel
> left out.
>
>
> RESULTS
> Here are the results for each of the above venues, measured in total
> hours of travel (i.e., round trip).
>
> Venue         Mean         Median           SD
> ----------------------------------------------
> SFO           13.5             11         12.2
> BOS           12.3             11          7.5
> ARN           17.0             21         10.7
> FRA           14.8             17          7.3
> LHR           13.3             14          7.5
> NYC           11.5             11          5.8
> YYC           14.9             13         10.2
> SFO/BOS/ARN   14.3             13          3.6
> SFO/NYC/LHR   12.7             11.3        3.7
>
> XXX/YYY/ZZZ a three-way rotation of XXX, YYY, and ZZZ. Obviously, mean
> and median are intended to be some sort of aggregate measure of travel
> time. I don't have any way to measure "fairness", but SD is intended
> as some metric of the variation in travel time between attendees.
>
> The raw data and software are attached. The files are:
>
>    home-airports     -- the list of people's home airports
>    durations.txt     -- the list of airport-airport durations
>    doodle.txt        -- the attendees list
>    pairings.py       -- the software to compute travel times
>    doodle-out.txt -- the computed travel times for each attendee
>
> Obviously, there could be an error in the raw data or the software.
> Please feel free to send corrections, especially if you find
> something material.
>
>
> OBSERVATIONS
> Obviously, it's hard to know what the optimal solution is without
> some model for optimality, but we can still make some observations
> based on this data:
>
> 1. If we're just concerned with minimizing total travel time, then we
> would always in New York, since it has both the shortest mean travel
> time and the shortest median travel time, but as I said above, this
> arguably isn't fair to people who live either in Europe or California,
> since they always have to travel.
>
> 2. Combining West Coast, East Coast, and European venues has
> comparable (or at least not too much worse) mean/median values than
> NYC with much lower SDs. So, arguably that kind of mix is more fair.
>
> 3. There's a pretty substantial difference between hub and non-hub
> venues. In particular, LHR has a median travel time 7 hours less than
> ARN, and the SFO/NYC/LHR combination has a median/mean travel time
> about 2 hours less than SFO/BOS/ARN (primarily accounted for by the
> LHR/ARN difference). [Full disclosure, I've favored Star Alliance hubs
> here, but you'd probably get similar results if, for instance, you
> used AMS instead of LHR.]
>
>
> Obviously, your mileage may vary based on your location and feelings
> about what's fair, but based on this data, it looks to me like a
> three-way rotation between West Coast, East Coast, and European hubs
> offers a good compromise between minimum cost and a flat distribution
> of travel times.
>
> Personally, whatever we decide to do I'd ask that the WG settle now on
> a pattern going forward so that we can predictably budget our travel
> time and dollars.
>
>
> [0] Treating all three NYC airports as a single location.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--------------020703020000000006030103
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    This post is the first in the history of standards--for the past 200
    or so years-- that I have been following where a scientific approach
    for a decision on the meeting venue has been planned and actually
    carried out.<br>
    <br>
    Bravo, Eric!<br>
    <br>
    Igor<br>
    <br>
    <br>
    On 4/9/2012 9:57 AM, Eric Rescorla wrote:
    <blockquote
cite="mid:CABcZeBPDpguge1zT5JyDk+tohMn1_av4jgdgDhNLnXMFKNzcbg@mail.gmail.com"
      type="cite">
      <pre wrap="">Hi folks,

Since it seems like we're going to be having a large number of
interims, I thought it might be instructive to try to analyze a bunch
of different locations to figure out the best strategy. My first cut
analysis is below.

Note that I'm not trying to make any claims about what the best set of
venues is. It's obviously easy to figure out any statistic we want
about each proposed venue, but how you map that data to "best" is up
to you. In particular, there's some tradeoff between minimal total
travel time and a "fair" distribution of travel times (not that I
claim to know what that means).


METHODOLOGY
The data below is derived by treating both people and venues as
airport locations and using travel time as our primary instrument.

1. For each responder for the current Doodle poll, assign a home
   airport based on their draft publication history.  We're missing a
   few people but basically it should be pretty complete. Since
   these people responded before the venue is known, it's at
   least somewhat unbiased.

2. Compute the shortest advertised flight between each home airport
   and the locations for each venue by looking at the shortest
   advertised Kayak flights around one of the proposed interim
   dates (6/10 - 6/13), ignoring price, but excluding "Hacker fares".
   [Thanks to Martin Thomson or helping me gather these.]

This lets us compute statistics for any venue and/or combination
of venues, based on the candidate attendee list.

The three proposed venues:

- San Francisco (SFO)
- Boston (BOS)
- Stockholm (ARN)

Three hubs not too distant from the proposed venues:

- London (LHR)
- Frankfurt (FRA)
- New York (NYC) [0]

Also, Calgary (YYC), since the other two chair locations (BOS and SFO)
were already proposed as venues, and I didn't want Cullen to feel
left out.


RESULTS
Here are the results for each of the above venues, measured in total
hours of travel (i.e., round trip).

Venue         Mean         Median           SD
----------------------------------------------
SFO           13.5             11         12.2
BOS           12.3             11          7.5
ARN           17.0             21         10.7
FRA           14.8             17          7.3
LHR           13.3             14          7.5
NYC           11.5             11          5.8
YYC           14.9             13         10.2
SFO/BOS/ARN   14.3             13          3.6
SFO/NYC/LHR   12.7             11.3        3.7

XXX/YYY/ZZZ a three-way rotation of XXX, YYY, and ZZZ. Obviously, mean
and median are intended to be some sort of aggregate measure of travel
time. I don't have any way to measure "fairness", but SD is intended
as some metric of the variation in travel time between attendees.

The raw data and software are attached. The files are:

  home-airports     -- the list of people's home airports
  durations.txt     -- the list of airport-airport durations
  doodle.txt        -- the attendees list
  pairings.py       -- the software to compute travel times
  doodle-out.txt -- the computed travel times for each attendee

Obviously, there could be an error in the raw data or the software.
Please feel free to send corrections, especially if you find
something material.


OBSERVATIONS
Obviously, it's hard to know what the optimal solution is without
some model for optimality, but we can still make some observations
based on this data:

1. If we're just concerned with minimizing total travel time, then we
would always in New York, since it has both the shortest mean travel
time and the shortest median travel time, but as I said above, this
arguably isn't fair to people who live either in Europe or California,
since they always have to travel.

2. Combining West Coast, East Coast, and European venues has
comparable (or at least not too much worse) mean/median values than
NYC with much lower SDs. So, arguably that kind of mix is more fair.

3. There's a pretty substantial difference between hub and non-hub
venues. In particular, LHR has a median travel time 7 hours less than
ARN, and the SFO/NYC/LHR combination has a median/mean travel time
about 2 hours less than SFO/BOS/ARN (primarily accounted for by the
LHR/ARN difference). [Full disclosure, I've favored Star Alliance hubs
here, but you'd probably get similar results if, for instance, you
used AMS instead of LHR.]


Obviously, your mileage may vary based on your location and feelings
about what's fair, but based on this data, it looks to me like a
three-way rotation between West Coast, East Coast, and European hubs
offers a good compromise between minimum cost and a flat distribution
of travel times.

Personally, whatever we decide to do I'd ask that the WG settle now on
a pattern going forward so that we can predictably budget our travel
time and dollars.


[0] Treating all three NYC airports as a single location.
</pre>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
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>
  </body>
</html>

--------------020703020000000006030103--

From magnus.westerlund@ericsson.com  Tue Apr 10 01:18:35 2012
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References: <4F7415D5.80604@ericsson.com> <0E6A0E0D-BFDD-4974-87BE-B0DCD1ECE9D5@acmepacket.com> <CAD5OKxu77MTLmq-zFouQ2sHfqoQcBfVDPdvhuKSdRnxrOY1-vQ@mail.gmail.com>
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Subject: Re: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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On 2012-04-02 18:09, Roman Shpount wrote:
> 
> On Mon, Apr 2, 2012 at 9:02 AM, Hadriel Kaplan <HKaplan@acmepacket.com
> <mailto:HKaplan@acmepacket.com>> wrote:
> 
> 
>     Hi,
>     I was thinking the tricky part is more about the local transport
>     stuff getting cloned - i.e., imagine if you're using a TCP-based
>     TURN connection.
>     But maybe it's a non-issue.
> 
>  
> This really depends on how the TURN support is implemented, but
> typically using the same TCP connection or any other TURN allocation for
> multiple PC should be no more of an issue then using same connection for
> multiple streams within the PC.

Actually TURN forces one to setup at least new permissions for each peer
that you want to accept flows from. Thus, even cloning an existing local
state will require additional actions for any TURN candidates. I haven't
looked into the detail for TURN TCP.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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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References: <4F4759DC.7060303@ericsson.com> <CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com> <4F6AECC6.8020004@alvestrand.no> <CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com> <CABkgnnW3AyREj9T2zDzJf64Bhjdfc1K8-ebZe-V0a-tbiTYEpw@mail.gmail.com> <CAD5OKxswXun5nVhKcN2oXkTdr-wKuOB7PhbXgdGSE4RexhTd3A@mail.gmail.com> <CABkgnnUZgtgEWZvYiyxDVNsDRDNsdr30Pd38cZeR18UB1Z04mw@mail.gmail.com> <CAD5OKxuLY2hZVqmna_GmpcBdFbZiP6ybKTJ19DN4f25mabLTrg@mail.gmail.com> <CA+9kkMDFEmJVuZu_BCBvU34YbWB-0CCR7SbKt-3PF=XV0vs3JQ@mail.gmail.com> <CAD5OKxu=JPJyWWPii=t8JGKAKTqVvB5JFQJ-jBBnXmiK4xpPEQ@mail.gmail.com> <CA+9kkMALJW2H5bM9yiykfBUqWoZifkZWpve=ih+ors1u9mz=7A@mail.gmail.com> <CAD5OKxva=zm0NUwUE0mhaEOmrDYCvtn3kP701NeRbiBRKhPFJQ@mail.gmail.com> <13C620B3-1297-4212-B61C-5643E08E9748@acmepacket.com> <CAD5OKxt_i2eJ1hK6tR7aKUt4T+6sitfxdMaBfN-ASR6-UPRE2A@mail.gmail.com> <4F6CC346.9000703@alvestrand.no> <CAD5OKxtNLawChET2AJmokJBkGQN-MCVfecqQrf+SgYPum+OuCg@mail.gmail.com >
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 2012-03-29 08:20, Roman Shpount wrote:
> 
> On Fri, Mar 23, 2012 at 2:39 PM, Harald Alvestrand <harald@alvestrand.no
> <mailto:harald@alvestrand.no>> wrote:
> 
>     __
>     It seems to me that you are arguing that the scenarios in section
>     4.1 of the use cases document do not cover that specific case, and I
>     think you are right in that; the list is:
> 
>        The following considerations are applicable to all use cases:
>        o  Clients can be on IPv4-only
>        o  Clients can be on IPv6-only
>        o  Clients can be on dual-stack
>        o  Clients can be on wideband (10s of Mbits/sec)
>        o  Clients can be on narrowband (10s to 100s of Kbits/sec)
>        o  Clients can be on variable-media-quality networks (wireless)
>        o  Clients can be on congested networks
>        o  Clients can be on firewalled networks with no UDP allowed
>        o  Clients can be on networks with cone NAT
>        o  Clients can be on networks with symmetric NAT
> 
>     Now, there are two ways to interpret this omission:
> 
>     - The WG did not think of that use case when the list was created
>     - The WG does not want that use case on the list because it
>     constrains the solution space too much
> 
>     If (re)opening this issue, I think I'd find myself in the "do not
>     want that use case" camp.
> 
> 
> I do not recall this use case ever being discussed on the working group,
> so I would assume the current situation is due to WG not thinking about
> this case when the list was created.

Hi,

If I followed this thread I do believe that we do have a use case
description that puts in requirements for controlling how the media
traffic flows in and out of an enterprise when using WebRTC:

4.2.4.  Simple Video Communication Service, enterprise aspects

4.2.4.1.  Description

   This use-case is similar to the Simple Video Communication Service
   use-case (Section 4.2.1).

   What is added is aspects when using the service in enterprises.  ICE
   is assumed in the further description of this use-case.

   An enterprise that uses a RTCWEB based web application for
   communication desires to audit all RTCWEB based application session
   used from inside the company towards any external peer.  To be able
   to do this they deploy a TURN server that straddle the boundary
   between the internal network and the external.

   The firewall will block all attempts to use STUN with an external
   destination unless they go to the enterprise auditing TURN server.
   In cases where employees are using RTCWEB applications provided by an
   external service provider they still want to have the traffic to stay
   inside their internal network and in addition not load the straddling
   TURN server, thus they deploy a STUN server allowing the RTCWEB
   client to determine its server reflexive address on the internal
   side.  Thus enabling cases where peers are both on the internal side
   to connect without the traffic leaving the internal network.  It must
   be possibele to configure the browsers used in the enterprise with
   network specific STUN and TURN servers.  This should be possible to
   achieve by autoconfiguration methods.  The RTCWEB functionality will
   need to utilize both network specific STUN and TURN resources and
   STUN and TURN servers provisioned by the web application.

My interpretation of this and the discussion I can remember is that an
enterprise would configure the browsers for their internal computers
with a TURN server sitting on the border between the inside and outside.
That way one can at least log and audit which communication that occurs
using WebRTC from the inside to the outside. The enterprise can also
select to record media flows in the TURN server.

This still leaves the question if one can get to the keys. That will
depend on the mechanism used for keying and its transport and what
methods have been put in place to capture such traffic.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From roman@telurix.com  Tue Apr 10 11:22:29 2012
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References: <4F4759DC.7060303@ericsson.com> <CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com> <4F6AECC6.8020004@alvestrand.no> <CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com> <CABkgnnW3AyREj9T2zDzJf64Bhjdfc1K8-ebZe-V0a-tbiTYEpw@mail.gmail.com> <CAD5OKxswXun5nVhKcN2oXkTdr-wKuOB7PhbXgdGSE4RexhTd3A@mail.gmail.com> <CABkgnnUZgtgEWZvYiyxDVNsDRDNsdr30Pd38cZeR18UB1Z04mw@mail.gmail.com> <CAD5OKxuLY2hZVqmna_GmpcBdFbZiP6ybKTJ19DN4f25mabLTrg@mail.gmail.com> <CA+9kkMDFEmJVuZu_BCBvU34YbWB-0CCR7SbKt-3PF=XV0vs3JQ@mail.gmail.com> <CAD5OKxu=JPJyWWPii=t8JGKAKTqVvB5JFQJ-jBBnXmiK4xpPEQ@mail.gmail.com> <CA+9kkMALJW2H5bM9yiykfBUqWoZifkZWpve=ih+ors1u9mz=7A@mail.gmail.com> <CAD5OKxva=zm0NUwUE0mhaEOmrDYCvtn3kP701NeRbiBRKhPFJQ@mail.gmail.com> <13C620B3-1297-4212-B61C-5643E08E9748@acmepacket.com> <CAD5OKxt_i2eJ1hK6tR7aKUt4T+6sitfxdMaBfN-ASR6-UPRE2A@mail.gmail.com> <4F6CC346.9000703@alvestrand.no> <CAD5OKxtNLawChET2AJmokJBkGQN-MCVfecqQrf+SgYPum+OuCg@mail.gmail.com> <4F8400E8.6010102@ericsson.com>
Date: Tue, 10 Apr 2012 14:22:22 -0400
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From: Roman Shpount <roman@telurix.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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--e89a8ff1c34032140a04bd57340e
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On Tue, Apr 10, 2012 at 5:44 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2012-03-29 08:20, Roman Shpount wrote:
> >
> > On Fri, Mar 23, 2012 at 2:39 PM, Harald Alvestrand <harald@alvestrand.no
> > <mailto:harald@alvestrand.no>> wrote:
> >
> >     __
> >     It seems to me that you are arguing that the scenarios in section
> >     4.1 of the use cases document do not cover that specific case, and I
> >     think you are right in that; the list is:
> >
> >        The following considerations are applicable to all use cases:
> >        o  Clients can be on IPv4-only
> >        o  Clients can be on IPv6-only
> >        o  Clients can be on dual-stack
> >        o  Clients can be on wideband (10s of Mbits/sec)
> >        o  Clients can be on narrowband (10s to 100s of Kbits/sec)
> >        o  Clients can be on variable-media-quality networks (wireless)
> >        o  Clients can be on congested networks
> >        o  Clients can be on firewalled networks with no UDP allowed
> >        o  Clients can be on networks with cone NAT
> >        o  Clients can be on networks with symmetric NAT
> >
> >     Now, there are two ways to interpret this omission:
> >
> >     - The WG did not think of that use case when the list was created
> >     - The WG does not want that use case on the list because it
> >     constrains the solution space too much
> >
> >     If (re)opening this issue, I think I'd find myself in the "do not
> >     want that use case" camp.
> >
> >
> > I do not recall this use case ever being discussed on the working group,
> > so I would assume the current situation is due to WG not thinking about
> > this case when the list was created.
>
> Hi,
>
> If I followed this thread I do believe that we do have a use case
> description that puts in requirements for controlling how the media
> traffic flows in and out of an enterprise when using WebRTC:
>
> 4.2.4.  Simple Video Communication Service, enterprise aspects
>
> 4.2.4.1.  Description
>
>   This use-case is similar to the Simple Video Communication Service
>   use-case (Section 4.2.1).
>
>   What is added is aspects when using the service in enterprises.  ICE
>   is assumed in the further description of this use-case.
>
>   An enterprise that uses a RTCWEB based web application for
>   communication desires to audit all RTCWEB based application session
>   used from inside the company towards any external peer.  To be able
>   to do this they deploy a TURN server that straddle the boundary
>   between the internal network and the external.
>
>   The firewall will block all attempts to use STUN with an external
>   destination unless they go to the enterprise auditing TURN server.
>   In cases where employees are using RTCWEB applications provided by an
>   external service provider they still want to have the traffic to stay
>   inside their internal network and in addition not load the straddling
>   TURN server, thus they deploy a STUN server allowing the RTCWEB
>   client to determine its server reflexive address on the internal
>   side.  Thus enabling cases where peers are both on the internal side
>   to connect without the traffic leaving the internal network.  It must
>   be possibele to configure the browsers used in the enterprise with
>   network specific STUN and TURN servers.  This should be possible to
>   achieve by autoconfiguration methods.  The RTCWEB functionality will
>   need to utilize both network specific STUN and TURN resources and
>   STUN and TURN servers provisioned by the web application.
>
> My interpretation of this and the discussion I can remember is that an
> enterprise would configure the browsers for their internal computers
> with a TURN server sitting on the border between the inside and outside.
> That way one can at least log and audit which communication that occurs
> using WebRTC from the inside to the outside. The enterprise can also
> select to record media flows in the TURN server.
>
> This still leaves the question if one can get to the keys. That will
> depend on the mechanism used for keying and its transport and what
> methods have been put in place to capture such traffic.


I thought about customer provided TURN servers and I do not think they will
be sufficient, due to the fact that no information describing the media
(URL of the application that initiated this media call, codec,any codec
related parameters, keys). I think some sort of network based hook that
will allow browser to send all the signaling information to some sort of
WebRTC signaling proxy server would be required to enable any type of
managed corporate use. Since we gave up the standard signaling protocol we
gave up a lot of functionality enterprise customers expect from real time
communications. In case of SIP enterprise can deploy some sort of proxy
server and enforce any type of enterprise specific policy. The only way to
fill this gap is to provide an ability to modify signaling coming from and
being sent via WebRTC API in a manner independent of the application code,
by configuring a policy enforcement server in web browser. The protocol
used to communicate with the policy enforcement server can be something as
simple as HTTP post with SDP data with response being policy server
modified SDP.  Any additional requirements such as authentication and
encryption of communications between WebRTC client and WebRTC policy server
will be provided via standard HTTP means (HTTPS, and Digest authentication).
_____________
Roman Shpount

--e89a8ff1c34032140a04bd57340e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Apr 10, 2012 at 5:44 A=
M, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerl=
und@ericsson.com">magnus.westerlund@ericsson.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 class=3D"im">On 2012-03-29 08:20, Roman Shpount wrote:<br>
&gt;<br>
&gt; On Fri, Mar 23, 2012 at 2:39 PM, Harald Alvestrand &lt;<a href=3D"mail=
to:harald@alvestrand.no">harald@alvestrand.no</a><br>
</div>&gt; &lt;mailto:<a href=3D"mailto:harald@alvestrand.no">harald@alvest=
rand.no</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =A0 =A0 __<br>
<div><div></div><div class=3D"h5">&gt; =A0 =A0 It seems to me that you are =
arguing that the scenarios in section<br>
&gt; =A0 =A0 4.1 of the use cases document do not cover that specific case,=
 and I<br>
&gt; =A0 =A0 think you are right in that; the list is:<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0The following considerations are applicable to all use =
cases:<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on IPv4-only<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on IPv6-only<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on dual-stack<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on wideband (10s of Mbits/sec)<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on narrowband (10s to 100s of Kbits=
/sec)<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on variable-media-quality networks =
(wireless)<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on congested networks<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on firewalled networks with no UDP =
allowed<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on networks with cone NAT<br>
&gt; =A0 =A0 =A0 =A0o =A0Clients can be on networks with symmetric NAT<br>
&gt;<br>
&gt; =A0 =A0 Now, there are two ways to interpret this omission:<br>
&gt;<br>
&gt; =A0 =A0 - The WG did not think of that use case when the list was crea=
ted<br>
&gt; =A0 =A0 - The WG does not want that use case on the list because it<br=
>
&gt; =A0 =A0 constrains the solution space too much<br>
&gt;<br>
&gt; =A0 =A0 If (re)opening this issue, I think I&#39;d find myself in the =
&quot;do not<br>
&gt; =A0 =A0 want that use case&quot; camp.<br>
&gt;<br>
&gt;<br>
&gt; I do not recall this use case ever being discussed on the working grou=
p,<br>
&gt; so I would assume the current situation is due to WG not thinking abou=
t<br>
&gt; this case when the list was created.<br>
<br>
</div></div>Hi,<br>
<br>
If I followed this thread I do believe that we do have a use case<br>
description that puts in requirements for controlling how the media<br>
traffic flows in and out of an enterprise when using WebRTC:<br>
<br>
4.2.4. =A0Simple Video Communication Service, enterprise aspects<br>
<br>
4.2.4.1. =A0Description<br>
<br>
 =A0 This use-case is similar to the Simple Video Communication Service<br>
 =A0 use-case (Section 4.2.1).<br>
<br>
 =A0 What is added is aspects when using the service in enterprises. =A0ICE=
<br>
 =A0 is assumed in the further description of this use-case.<br>
<br>
 =A0 An enterprise that uses a RTCWEB based web application for<br>
 =A0 communication desires to audit all RTCWEB based application session<br=
>
 =A0 used from inside the company towards any external peer. =A0To be able<=
br>
 =A0 to do this they deploy a TURN server that straddle the boundary<br>
 =A0 between the internal network and the external.<br>
<br>
 =A0 The firewall will block all attempts to use STUN with an external<br>
 =A0 destination unless they go to the enterprise auditing TURN server.<br>
 =A0 In cases where employees are using RTCWEB applications provided by an<=
br>
 =A0 external service provider they still want to have the traffic to stay<=
br>
 =A0 inside their internal network and in addition not load the straddling<=
br>
 =A0 TURN server, thus they deploy a STUN server allowing the RTCWEB<br>
 =A0 client to determine its server reflexive address on the internal<br>
 =A0 side. =A0Thus enabling cases where peers are both on the internal side=
<br>
 =A0 to connect without the traffic leaving the internal network. =A0It mus=
t<br>
 =A0 be possibele to configure the browsers used in the enterprise with<br>
 =A0 network specific STUN and TURN servers. =A0This should be possible to<=
br>
 =A0 achieve by autoconfiguration methods. =A0The RTCWEB functionality will=
<br>
 =A0 need to utilize both network specific STUN and TURN resources and<br>
 =A0 STUN and TURN servers provisioned by the web application.<br>
<br>
My interpretation of this and the discussion I can remember is that an<br>
enterprise would configure the browsers for their internal computers<br>
with a TURN server sitting on the border between the inside and outside.<br=
>
That way one can at least log and audit which communication that occurs<br>
using WebRTC from the inside to the outside. The enterprise can also<br>
select to record media flows in the TURN server.<br>
<br>
This still leaves the question if one can get to the keys. That will<br>
depend on the mechanism used for keying and its transport and what<br>
methods have been put in place to capture such traffic.</blockquote><div><b=
r>I thought about customer provided TURN servers and I do not think they wi=
ll be sufficient, due to the fact that no information describing the media =
(URL of the application that initiated this media call, codec,any codec rel=
ated parameters, keys). I think some sort of network based hook that will a=
llow browser to send all the signaling information to some sort of WebRTC s=
ignaling proxy server would be required to enable any type of managed corpo=
rate use. Since we gave up the standard signaling protocol we gave up a lot=
 of functionality enterprise customers expect from real time communications=
. In case of SIP enterprise can deploy some sort of proxy server and enforc=
e any type of enterprise specific policy. The only way to fill this gap is =
to provide an ability to modify signaling coming from and being sent via We=
bRTC API in a manner independent of the application code, by configuring a =
policy enforcement server in web browser. The protocol used to communicate =
with the policy enforcement server can be something as simple as HTTP post =
with SDP data with response being policy server modified SDP.=A0 Any additi=
onal requirements such as authentication and encryption of communications b=
etween WebRTC client and WebRTC policy server will be provided via standard=
 HTTP means (HTTPS, and Digest authentication).<br>
</div></div>_____________<br>Roman Shpount<br>
<br>

--e89a8ff1c34032140a04bd57340e--

From harald@alvestrand.no  Tue Apr 10 11:39:40 2012
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To: Roman Shpount <roman@telurix.com>
References: <4F4759DC.7060303@ericsson.com>	<CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com>	<CABkgnnW3AyREj9T2zDzJf64Bhjdfc1K8-ebZe-V0a-tbiTYEpw@mail.gmail.com>	<CAD5OKxswXun5nVhKcN2oXkTdr-wKuOB7PhbXgdGSE4RexhTd3A@mail.gmail.com>	<CABkgnnUZgtgEWZvYiyxDVNsDRDNsdr30Pd38cZeR18UB1Z04mw@mail.gmail.com>	<CAD5OKxuLY2hZVqmna_GmpcBdFbZiP6ybKTJ19DN4f25mabLTrg@mail.gmail.com>	<CA+9kkMDFEmJVuZu_BCBvU34YbWB-0CCR7SbKt-3PF=XV0vs3JQ@mail.gmail.com>	<CAD5OKxu=JPJyWWPii=t8JGKAKTqVvB5JFQJ-jBBnXmiK4xpPEQ@mail.gmail.com>	<CA+9kkMALJW2H5bM9yiykfBUqWoZifkZWpve=ih+ors1u9mz=7A@mail.gmail.com>	<CAD5OKxva=zm0NUwUE0mhaEOmrDYCvtn3kP701NeRbiBRKhPFJQ@mail.gmail.com>	<13C620B3-1297-4212-B61C-5643E08E9748@acmepacket.com>	<CAD5OKxt_i2eJ1hK6tR7aKUt4T+6sitfxdMaBfN-ASR6-UPRE2A@mail.gmail.com>	<4F6CC346.9000703@alvestrand.no>	<CAD5OKxtNLawChET2AJmokJBkGQN-MCVfecqQrf+SgYPum+OuCg@mail.gmail.com>	<4F8400E8.6010102@ericsson.com> <CAD5OKxvrdsp30GofOmF7YgjVd_yg4f+pOFS-CHo+A3oDCZF3bw@mail.gmail.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 04/10/2012 08:22 PM, Roman Shpount wrote:
> I thought about customer provided TURN servers and I do not think they 
> will be sufficient, due to the fact that no information describing the 
> media (URL of the application that initiated this media call, 
> codec,any codec related parameters, keys). I think some sort of 
> network based hook that will allow browser to send all the signaling 
> information to some sort of WebRTC signaling proxy server would be 
> required to enable any type of managed corporate use. Since we gave up 
> the standard signaling protocol we gave up a lot of functionality 
> enterprise customers expect from real time communications. In case of 
> SIP enterprise can deploy some sort of proxy server and enforce any 
> type of enterprise specific policy. The only way to fill this gap is 
> to provide an ability to modify signaling coming from and being sent 
> via WebRTC API in a manner independent of the application code, by 
> configuring a policy enforcement server in web browser. The protocol 
> used to communicate with the policy enforcement server can be 
> something as simple as HTTP post with SDP data with response being 
> policy server modified SDP.  Any additional requirements such as 
> authentication and encryption of communications between WebRTC client 
> and WebRTC policy server will be provided via standard HTTP means 
> (HTTPS, and Digest authentication).

This requires the code that sends the SDP to the policy server to be 
part of the browser, since you obviously can't trust the Javascript to 
do the call-out to the server. (If you could trust the Javascript, you 
could also trust it to enforce policy.)

In that case, why don't you just require that the browser do the 
wiretapping for you by copying the media? Much simpler, and you don't 
have to worry about interpreting the SIP.

                           Harald


From dean.willis@softarmor.com  Tue Apr 10 11:50:44 2012
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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--e89a8f83a03b80513b04bd57995b
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On Tue, Apr 3, 2012 at 12:54 PM, Serge Lachapelle <sergel@google.com> wrote:

>
> Google confirms that the VP8 patent grant applies to both third-party
> hardware and software implementations of VP8.
> *
> *
>
>
That's very nice of Google, but I'm personally not worried about Google
filing suit against me (as a small developer) if I implement using VP8.

What I'm worried about is either a troll who has acquired a patent through
the usual fire-sale process, or a Big Old Research Lab that has a gazillion
patents but no revenue and decides to start monetizing patents its
management team had forgotten existed. Both sorts of entities are very
active in today's world of patent fights. And there are patents on design
patterns going way back; we're approaching a world of "solid invention"
where every alternative design for anything we know how to do has already
been described in a patent filing.

So if you really want me to be comfortable with implementing using VP8, buy
me an insurance policy that will indemnify me (and any other developer) for
any costs related to defending against any future infringement claims on
VP8. Perhaps I'd even settle for a free software stack that includes such
indemnification in its license. This wouldn't help the hardware
implementer, much, however.

--
Dean

--e89a8f83a03b80513b04bd57995b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Tue, Apr 3, 2012 at 12:54 PM, Serge L=
achapelle <span dir=3D"ltr">&lt;<a href=3D"mailto:sergel@google.com">sergel=
@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><font face=3D"Arial"><span style=3D"font-size:15px;white-space:pre-wra=
p"><br></span></font></div><div><span style=3D"font-family:Times;font-size:=
medium">

<span style=3D"font-size:15px;font-family:Arial;font-weight:normal;vertical=
-align:baseline;white-space:pre-wrap">Google confirms that the VP8 patent g=
rant applies to both third-party hardware and software implementations of V=
P8.=A0</span></span></div>


<div><b style=3D"font-family:Times;font-size:medium"><span style=3D"font-si=
ze:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-=
space:pre-wrap"><br></span></b></div><div><font face=3D"Arial"><span style=
=3D"font-size:15px;white-space:pre-wrap"><br>
</span></font></div></blockquote><div><br></div><div>That&#39;s very nice o=
f Google, but I&#39;m personally not worried about Google filing suit again=
st me (as a small developer) if I implement using VP8.=A0</div><div><br></d=
iv>
<div>What I&#39;m worried about is either a troll who has acquired a patent=
 through the usual fire-sale process, or a Big Old Research Lab that has a =
gazillion patents but no revenue and decides to start monetizing patents it=
s management team had forgotten existed. Both sorts of entities are very ac=
tive in today&#39;s world of patent fights. And there are patents on design=
 patterns going way back; we&#39;re approaching a world of &quot;solid inve=
ntion&quot; where every alternative design for anything we know how to do h=
as already been described in a patent filing.</div>
<div><br></div><div>So if you really want me to be comfortable with impleme=
nting using VP8, buy me an insurance policy that will indemnify me (and any=
 other developer) for any costs related to defending against any future inf=
ringement claims on VP8. Perhaps I&#39;d even settle for a free software st=
ack that includes such indemnification in its license. This wouldn&#39;t he=
lp the hardware implementer, much, however.</div>
<div><br></div><div>--</div><div>Dean</div></div>

--e89a8f83a03b80513b04bd57995b--

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Subject: Re: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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--047d7b33d7c474d54704bd57adce
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On Tue, Apr 10, 2012 at 4:18 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Actually TURN forces one to setup at least new permissions for each peer
> that you want to accept flows from. Thus, even cloning an existing local
> state will require additional actions for any TURN candidates. I haven't
> looked into the detail for TURN TCP.
>
>
You are correct, when a new remote peer is added to a given allocation TURN
allocation, a new permission needs to be added if remote peer address is
different (permission is per address), new channel needs to be
created(optional but highly recommended for RTP), and ICE exchange should
be completed to confirm the connectivity. This procedure is the same for
adding new remote peer due to normal session re-negotiation (you can send
an offer with the same SDP on existing connetion and get a new response
with different peers back), reusing TURN allocation in case of bundle, or
in case we discuss -- forking. I would expect that a functional WebRTC
implementation must be able to reuse the same TURN allocation for multiple
connections within the same call. I do not think that adding support for
this to enable reuse of the same allocation across multiple calls would
complicate things too much.
_____________
Roman Shpount

--047d7b33d7c474d54704bd57adce
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<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Apr 10, 2012 at 4:18 A=
M, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerl=
und@ericsson.com">magnus.westerlund@ericsson.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">
Actually TURN forces one to setup at least new permissions for each peer<br=
>
that you want to accept flows from. Thus, even cloning an existing local<br=
>
state will require additional actions for any TURN candidates. I haven&#39;=
t<br>
looked into the detail for TURN TCP.<br>
<div><div></div><br></div></blockquote><div><br>You are correct, when a new=
 remote peer is added to a given allocation TURN allocation, a new permissi=
on needs to be added if remote peer address is different (permission is per=
 address), new channel needs to be created(optional but highly recommended =
for RTP), and ICE exchange should be completed to confirm the connectivity.=
 This procedure is the same for adding new remote peer due to normal sessio=
n re-negotiation (you can send an offer with the same SDP on existing conne=
tion and get a new response with different peers back), reusing TURN alloca=
tion in case of bundle, or in case we discuss -- forking. I would expect th=
at a functional WebRTC implementation must be able to reuse the same TURN a=
llocation for multiple connections within the same call. I do not think tha=
t adding support for this to enable reuse of the same allocation across mul=
tiple calls would complicate things too much.<br>
</div></div>_____________<br>Roman Shpount<br>

--047d7b33d7c474d54704bd57adce--

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Date: Tue, 10 Apr 2012 15:06:37 -0400
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From: Roman Shpount <roman@telurix.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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--e89a8ffbae9f706d5304bd57d218
Content-Type: text/plain; charset=ISO-8859-1

On Tue, Apr 10, 2012 at 2:39 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

>
> This requires the code that sends the SDP to the policy server to be part
> of the browser, since you obviously can't trust the Javascript to do the
> call-out to the server. (If you could trust the Javascript, you could also
> trust it to enforce policy.)
>
> In that case, why don't you just require that the browser do the
> wiretapping for you by copying the media? Much simpler, and you don't have
> to worry about interpreting the SIP.
>

I do think this should be part of the browser, but I am not sure why you
insist on limiting this to wiretapping. Wiretapping is just one of the use
cases. Other use cases can be supported are things like:

1. Allocate only certain amount of bandwidth to WebRTC, disallow WebRTC
calls when bandwidth is exausted
2. Disable video calling from my network. Voice calling would still be
non-monitored encrypted and allowed
3. Force certain codec preference from my network (disable anything above
16KB/s)
4. Allow WebRTC communications outside of my network only when SDP
descriptions are signed by certain white listed companies.

There is a reason why proxy servers, both SIP and HTTP, exist. What I am
trying to explain is that in a lot of environments user network experience
is managed. This management requires tools. WebRTC, due to its consumer
oriented user rights focus and prioritization of security, provides none,
which, I believe, will create a barrier to its deployment in enterprise
environments.

_____________
Roman Shpount

--e89a8ffbae9f706d5304bd57d218
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Apr 10, 2012 at 2:39 P=
M, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestr=
and.no">harald@alvestrand.no</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 class=3D"im"><br></div>
This requires the code that sends the SDP to the policy server to be part o=
f the browser, since you obviously can&#39;t trust the Javascript to do the=
 call-out to the server. (If you could trust the Javascript, you could also=
 trust it to enforce policy.)<br>

<br>
In that case, why don&#39;t you just require that the browser do the wireta=
pping for you by copying the media? Much simpler, and you don&#39;t have to=
 worry about interpreting the SIP.<font color=3D"#888888"></font><br></bloc=
kquote>
</div><br>I do think this should be part of the browser, but I am not sure =
why you insist on limiting this to wiretapping. Wiretapping is just one of =
the use cases. Other use cases can be supported are things like:<br><br>
1. Allocate only certain amount of bandwidth to WebRTC, disallow WebRTC cal=
ls when bandwidth is exausted<br>2. Disable video calling from my network. =
Voice calling would still be non-monitored encrypted and allowed<br>3. Forc=
e certain codec preference from my network (disable anything above 16KB/s)<=
br>
4. Allow WebRTC communications outside of my network only when SDP descript=
ions are signed by certain white listed companies.<br><br>There is a reason=
 why proxy servers, both SIP and HTTP, exist. What I am trying to explain i=
s that in a lot of environments user network experience is managed. This ma=
nagement requires tools. WebRTC, due to its consumer oriented user rights f=
ocus and prioritization of security, provides none, which, I believe, will =
create a barrier to its deployment in enterprise environments.<br>
<br>_____________<br>Roman Shpount<br>
<br>

--e89a8ffbae9f706d5304bd57d218--

From randell-ietf@jesup.org  Tue Apr 10 12:32:44 2012
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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On 4/10/2012 2:50 PM, Dean Willis wrote:
>
> On Tue, Apr 3, 2012 at 12:54 PM, Serge Lachapelle <sergel@google.com 
> <mailto:sergel@google.com>> wrote:
>
>
>     Google confirms that the VP8 patent grant applies to both
>     third-party hardware and software implementations of VP8.
>     *
>     *
>
>
> That's very nice of Google, but I'm personally not worried about 
> Google filing suit against me (as a small developer) if I implement 
> using VP8.
>
> What I'm worried about is either a troll who has acquired a patent 
> through the usual fire-sale process, or a Big Old Research Lab that 
> has a gazillion patents but no revenue and decides to start monetizing 
> patents its management team had forgotten existed. Both sorts of 
> entities are very active in today's world of patent fights. And there 
> are patents on design patterns going way back; we're approaching a 
> world of "solid invention" where every alternative design for anything 
> we know how to do has already been described in a patent filing.
>
> So if you really want me to be comfortable with implementing using 
> VP8, buy me an insurance policy that will indemnify me (and any other 
> developer) for any costs related to defending against any future 
> infringement claims on VP8. Perhaps I'd even settle for a free 
> software stack that includes such indemnification in its license. This 
> wouldn't help the hardware implementer, much, however.

Such is the world of patents (especially software/algorithmic patents).  
Stick to stuff that's ~25-30 years old and you're probably ok (though 
someone may ding you for combining old thing A with old thing B).  
Original MPEG-1 for video is probably ok, just don't add anything to it 
or extend it or use it  in a way not originally envisioned (like over 
the internet).

Or live with non-0 risk of patent trolls.

-- 
Randell Jesup
randell-ietf@jesup.org


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(not actually a quote):

> So if you really want me to be comfortable with implementing using h264, buy
> me an insurance policy that will indemnify me (and any other developer) for
> any costs related to defending against any future infringement claims on
> h264. Perhaps I'd even settle for a fully licensed software stack that includes such
> indemnification in its license. This wouldn't help the hardware implementer,
> much, however.

Well darn, that isn't going to fly either!

Monty
Xiph.Org

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Date: Wed, 11 Apr 2012 00:18:59 -0400
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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> (not actually a quote):
> 
> > So if you really want me to be comfortable with implementing using
> > h264, buy me an insurance policy that will indemnify me (and any other
> > developer) for any costs related to defending against any future
> > infringement claims on h264. Perhaps I'd even settle for a fully
> > licensed software stack that includes such indemnification in its
> > license. This wouldn't help the hardware implementer, much, however.
> 
> Well darn, that isn't going to fly either!

We're still debating this on this?

I said before, either direction is a gamble.  But, the odds of not getting
successfully sued are in your favor with H.264, in my opinion.  Here's why:

1) Virtually all companies that have IPR on H.264 are known and the odds of
new companies entering the scene at this late stage are slim (seriously,
what troll would not have already made a claim by now)
2) Companies who developed H.264 have a financial interest in ensuring
H.264's success and would work to defend against bogus claims (I cannot
offer up their legal resources, but clearly they will not want some bogus
claim to stand in the way of profits, either from use of H.264 themselves or
licensing H.264 technology)
3) H.264 is EXTREMELY WIDELY deployed now and anything less than a concrete
claim is going to scrutinized heavily, the costs to pursue claims very high,
etc.  Legitimate companies with IPR should have already filed statements and
serious questions would have been asked.  To say such "holdout" companies
would not win friends at this point with some claim would be an
understatement.
4) While law and logic are sometimes at odds, I have to work from logic
since I'm not a lawyer.  Logically speaking, if there are 200+ patents
covering various aspects of H.264 and one more company comes in to make
claims, what should any reasonable court agree as to the value of that lone
patent versus the 200+ others on the same piece of technology?  (OK, this
might be reaching, but I seriously do think that damage awards would have to
considered in the context of that background.)
5) A damaging claim against H.264 at this point would affect entire
industries, and not just one industry.  I think even governments should
care, because it has an economic impact on a country.  I cannot imagine that
any court would issue an in injunction to halt sale of videos or streaming
video services world-wide, for example, on the basis of some claim relating
to some unknown patent nearly 10 years after the standard was produced.  If
such a claimant had valid IPR and even remotely claimed to be a subject
matter expert, they should be fully aware that every effort has been made to
identify IPR holders.  If I were a judge, I would not be too tolerant of
this (what I classify) deliberate, malicious, destructive behavior.

Now, should you choose to go down the path of VP8:

1) The only known IPR holder is Google, who has kindly offered to make their
IPR available for free
2) There are many, many companies who have IPR on H.264 because their
business includes, entirely or as a part, video coding.  It's impossible for
me to believe that with so much IPR on H.264 and so much research that went
into H.264, much of that research NOT finding its way into H.264, that not a
single one of those inventions by one of those companies does not cover VP8
3) There are legitimate companies who have invested millions in the
development of H.264 and own IPR on H.264.  Should VP8 becomes successful
and those companies lose revenue at the expense of VP8 and they hold IPR on
VP8, they would have every valid reason to assess royalties on your use of
VP8.  What do you want to bet that the royalties will be even higher than
use of H.264, too?  (That's a different bet, but since we're gambling
here...)
4) VP8 is not a standard and nobody was obligated to file any IPR statements
on VP8 (whereas participants in the development of H.264 were all requested
to submit statements; company representatives are asked at nearly every
meeting)
5) VP8 is not widely deployed and, quite possibly, not widely studied, even
by a good many experts in the video coding space  (I'm sure some have
studied it, but I am sure others are busy building "the next great thing").
As such, nobody should be surprised one day if VP8 does become widely
deployed that a company other than Google comes along and lays claim to the
technology
6) Should you get sued for embedding VP8 in your hardware or using it in
your software, Google is not going to indemnify you.  You are on your own.
The industry is not going to back you up (whereas I think they would for
H.264's benefit), because we all know that using VP8 is done so at our own
risk.  Perhaps with really good luck an H.264 patent holder might come to
your defense with evidence to refute, but don't count on it.  Those IPR
holders would prefer you switch to H.264.

As I said before, my opinions say nothing about the technical merits of one
codec vs. the other.  This is just my ramblings as I think through the legal
minefield here.  I personally see H.264 as relatively safe as compared to
VP8.

Of course, we will never know until VP8 gets widely deployed.  No IPR holder
who would be seeking to profit on VP8 will disclose a thing until VP8 is
widely deployed.

It is a gamble.  Place your bets.  My money is on H.264 as the safer option.

Paul



From magnus.westerlund@ericsson.com  Wed Apr 11 00:47:19 2012
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WG,

I would like to remind about the poll for the date of the Interim
meeting between now and IETF 84. We will make a decision after Friday
(13th of April).

Locations considered so far are Stockholm, Boston Area or Silicon
Valley. The meeting is planned to be a 2 day meeting. Please indicate
which of the suggest times that works for you.

http://doodle.com/nm3pp69znr3286cy

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From harald@alvestrand.no  Wed Apr 11 04:46:15 2012
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Just so that members are aware of this...

if it's permissible by W3C rules and logistics, the W3C WEBRTC WG plans 
to hold a WG meeting in conjunction with this RTCWEB meeting. It will 
likely be an one-day meeting.

                     Harald

On 04/11/2012 09:47 AM, Magnus Westerlund wrote:
> WG,
>
> I would like to remind about the poll for the date of the Interim
> meeting between now and IETF 84. We will make a decision after Friday
> (13th of April).
>
> Locations considered so far are Stockholm, Boston Area or Silicon
> Valley. The meeting is planned to be a 2 day meeting. Please indicate
> which of the suggest times that works for you.
>
> http://doodle.com/nm3pp69znr3286cy
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 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
>


From magnus.westerlund@ericsson.com  Wed Apr 11 05:52:59 2012
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WG,

We chair would like to discuss some policies we WG chairs intended to
use for interim meetings going forward.

1) For physical meetings the WG chairs select the location in a
round-robin fashion among regions where we have significant amount of
active contributors. This is to spread the inconvenience and cost of
long distance travel.

2) We will in the future pick the location prior to any poll for
suitable dates so that participants can weigh the location fully into
there response to any such poll.

3) For Virtual Interims the time of the day when the meeting is held
will be alternated among possible times if necessary to enable
reasonable participation times for regions with significant amount of
active contributors. This is also to spread the pain of inconvenient
time periods of the day.

Please comment on these policies. Additional policies that you want us
WG chairs to use?

Cheers

Magnus Westerlund
(WG chair)

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Ericsson AB                | Phone  +46 10 7148287
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Subject: [rtcweb] WebRTC 2012: CFP deadline extension
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The WebRTC 2012 CFP deadline has been extended to April 30.
More details at:
http://www.uppersideconferences.com/webrtc2012/webrtc2012intro.html

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11 apr 2012 kl. 15:38 skrev Remi Scavenius:

> 
> The WebRTC 2012 CFP deadline has been extended to April 30.
> More details at:
> http://www.uppersideconferences.com/webrtc2012/webrtc2012intro.html

What's the IETF policy in regards to this kind of messages. 
This conference is highly commercial and not related to the IETF.

In my view, it's very close to spamming. 

/O


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References: <4F740AB6.1080609@ericsson.com> <4F853704.1030000@ericsson.com> <4F856F04.5050301@alvestrand.no>
Date: Wed, 11 Apr 2012 10:15:08 -0500
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Subject: Re: [rtcweb] Poll for dates for Interim Meeting before IETF 84
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--20cf3079bc9e73434e04bd68b4a9
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And, just so folks are aware, we are also trying to schedule a CLUE meeting
(as announced at IETF-83) in conjunction with the RTCWEB meeting as there
is overlap in participation as well as technical issues:

The doodle for CLUE was crafted around the dates originally proposed for
the RTCWEB meeting:  http://doodle.com/zd4mhfsb63wkm7ww

It still remains a possibility given the current results that we could hold
the meetings the same week.

Mary.
CLUE WG co-chair

On Wed, Apr 11, 2012 at 6:46 AM, Harald Alvestrand <harald@alvestrand.no>wr=
ote:

> Just so that members are aware of this...
>
> if it's permissible by W3C rules and logistics, the W3C WEBRTC WG plans t=
o
> hold a WG meeting in conjunction with this RTCWEB meeting. It will likely
> be an one-day meeting.
>
>                    Harald
>
>
> On 04/11/2012 09:47 AM, Magnus Westerlund wrote:
>
>> WG,
>>
>> I would like to remind about the poll for the date of the Interim
>> meeting between now and IETF 84. We will make a decision after Friday
>> (13th of April).
>>
>> Locations considered so far are Stockholm, Boston Area or Silicon
>> Valley. The meeting is planned to be a 2 day meeting. Please indicate
>> which of the suggest times that works for you.
>>
>> http://doodle.com/**nm3pp69znr3286cy <http://doodle.com/nm3pp69znr3286cy=
>
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ------------------------------**------------------------------**
>> ----------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ------------------------------**------------------------------**
>> ----------
>> 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<https://www.ietf.org/mail=
man/listinfo/rtcweb>
>>
>>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailm=
an/listinfo/rtcweb>
>

--20cf3079bc9e73434e04bd68b4a9
Content-Type: text/html; charset=ISO-8859-1
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And, just so folks are aware, we are also trying to schedule a CLUE meeting=
 (as announced at IETF-83) in conjunction with the RTCWEB meeting as there =
is overlap in participation as well as technical issues:<div>=A0=A0</div><d=
iv>
The doodle for CLUE was crafted around the dates originally proposed for th=
e RTCWEB meeting: =A0<a href=3D"http://doodle.com/zd4mhfsb63wkm7ww">http://=
doodle.com/zd4mhfsb63wkm7ww</a></div><div><br><div>It still remains a possi=
bility given the current results that we could hold the meetings the same w=
eek.=A0</div>
<div><br></div><div>Mary.=A0</div><div>CLUE WG co-chair</div><div><br></div=
><div><div class=3D"gmail_quote">On Wed, Apr 11, 2012 at 6:46 AM, Harald Al=
vestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no">hara=
ld@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">Just so that members are aware of this...<br=
>
<br>
if it&#39;s permissible by W3C rules and logistics, the W3C WEBRTC WG plans=
 to hold a WG meeting in conjunction with this RTCWEB meeting. It will like=
ly be an one-day meeting.<br><font color=3D"#888888">
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Harald</font><div><div></div><div c=
lass=3D"h5"><br>
<br>
On 04/11/2012 09:47 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">
WG,<br>
<br>
I would like to remind about the poll for the date of the Interim<br>
meeting between now and IETF 84. We will make a decision after Friday<br>
(13th of April).<br>
<br>
Locations considered so far are Stockholm, Boston Area or Silicon<br>
Valley. The meeting is planned to be a 2 day meeting. Please indicate<br>
which of the suggest times that works for you.<br>
<br>
<a href=3D"http://doodle.com/nm3pp69znr3286cy" target=3D"_blank">http://doo=
dle.com/<u></u>nm3pp69znr3286cy</a><br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
------------------------------<u></u>------------------------------<u></u>-=
---------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
------------------------------<u></u>------------------------------<u></u>-=
---------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287" target=3D"_blank">+46 10 7148287</a>=
<br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079" target=3D"_blank">+46 73 0949079</=
a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
------------------------------<u></u>------------------------------<u></u>-=
---------<br>
<br>
______________________________<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>
<br>
</blockquote>
<br>
______________________________<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>

--20cf3079bc9e73434e04bd68b4a9--

From harald@alvestrand.no  Wed Apr 11 11:00:10 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Considerations for Interim Meetings
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On 04/11/2012 02:52 PM, Magnus Westerlund wrote:
> WG,
>
> We chair would like to discuss some policies we WG chairs intended to
> use for interim meetings going forward.
>
> 1) For physical meetings the WG chairs select the location in a
> round-robin fashion among regions where we have significant amount of
> active contributors. This is to spread the inconvenience and cost of
> long distance travel.
When considering where to put the next instalment of the round-robin, 
will the location of IETF meetings be considered too?
I note that the 3 next IETF meetings are all in North America 
(Vancouver, Atlanta and Orlando). It would be very strange if a meeting 
between Atlanta and Orlando were held in the eastern US.
>
> 2) We will in the future pick the location prior to any poll for
> suitable dates so that participants can weigh the location fully into
> there response to any such poll.
>
> 3) For Virtual Interims the time of the day when the meeting is held
> will be alternated among possible times if necessary to enable
> reasonable participation times for regions with significant amount of
> active contributors. This is also to spread the pain of inconvenient
> time periods of the day.
>
> Please comment on these policies. Additional policies that you want us
> WG chairs to use?
For logistics, long lead times are a Good Thing; the formal W3C 
announcement lead time is 8 weeks. This means that it will be good to 
announce the intent of holding an interim *before* the prior IETF meeting.

For interims in response to breaking issues, we should use 
teleconferences, IMHO.

>
> Cheers
>
> Magnus Westerlund
> (WG chair)
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 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
>


From dean.willis@softarmor.com  Wed Apr 11 23:29:18 2012
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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--f46d0444ee237edf8304bd757954
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>
>
>
> I said before, either direction is a gamble.  But, the odds of not getting
> successfully sued are in your favor with H.264, in my opinion.  Here's why:
>
>
That's arguable. As a small developer (and a poor one), I'm not able to pay
for licensing H.264. Therefore, my odds of getting sued (if anybody notices
my work) are pretty high. Just ask Microsoft what the odds of getting sued
for using H.264 are ... or the costs of trying to license it (reports are
that Moto was asking $4 billion for their patents alone).

http://www.rethink-wireless.com/article.asp?article_id=23154

I'm not saying VP8 is any safer in the long run, but it's certainly easier
to comply with its licensing terms up front. So I might accidentally
infringe, but it wouldn't be willful (at least on the essential core; there
are a lot of patents about there about stuff you might want to do with the
video that cover both codecs).

Think of it as picking up a random snake that MIGHT be venomous, versus
picking up one that's already buzzing its tail and striking at movement
(caught one in my yard Monday, actually).

The thing to remember is that you need to handle both with respect. They're
still snakes. Even an unenvenomed bite can get infected. And the quiet one
might be a cobra instead of a rattlesnake.

--
Dean

--f46d0444ee237edf8304bd757954
Content-Type: text/html; charset=ISO-8859-1
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<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
<br>
I said before, either direction is a gamble. =A0But, the odds of not gettin=
g<br>
successfully sued are in your favor with H.264, in my opinion. =A0Here&#39;=
s why:<br><br></blockquote><div><br></div><div>That&#39;s arguable. As a sm=
all developer (and a poor one), I&#39;m not able to pay for licensing H.264=
. Therefore, my odds of getting sued (if anybody notices my work) are prett=
y high. Just ask Microsoft what the odds of getting sued for using H.264 ar=
e ... or the costs of trying to license it (reports are that Moto was askin=
g $4 billion for their patents alone).</div>
<div><br></div><div><a href=3D"http://www.rethink-wireless.com/article.asp?=
article_id=3D23154">http://www.rethink-wireless.com/article.asp?article_id=
=3D23154</a>
</div><div><br></div><div>I&#39;m not saying VP8 is any safer in the long r=
un, but it&#39;s certainly easier to comply with its licensing terms up fro=
nt. So I might accidentally infringe, but it wouldn&#39;t be willful (at le=
ast on the essential core; there are a lot of patents about there about stu=
ff you might want to do with the video that cover both codecs).</div>
<div><br></div><div>Think of it as picking up a random snake that MIGHT be =
venomous, versus picking up one that&#39;s already buzzing its tail and str=
iking at movement (caught one in my yard Monday, actually).</div><div><br>
</div><div>The thing to remember is that you need to handle both with respe=
ct. They&#39;re still snakes. Even an unenvenomed bite can get infected. An=
d the quiet one might be a cobra instead of a rattlesnake.</div><div><br>
</div><div>--</div><div>Dean</div></div>

--f46d0444ee237edf8304bd757954--

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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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On Thu, Apr 5, 2012 at 1:47 AM, Paul E. Jones <p
>
> While there may be undisclosed IPR on H.264, I find it hard to believe that
> there would be much, if any, at this point.  Keep in mind that H.264 is
> product of the joint effort of a bunch of people who are experts in the
> field.  I would be suspicious of any company not involved in the
> development
> of H.264 claiming to have IPR, because somebody in the joint committee
> probably owns that IPR.  Sure, there might be something ... something very
> minor.  Perhaps.
>
>
Both H.264 and  VP8 are motion-vector codings, right?

A quick search on Google Patents shows about 32,400 existing motion vector
video patents. What do you want to bet some of those apply  to one or the
other? Or both? Or both plus H.263?

Just because they haven't been asserted yet doesn't mean they're not real.
It might mean that the people that own them aren't paying attention. Or
weren't, but have started. Or don't care, but might be willing to sell
their patent to someone who does.

The simple fact is that there are a crapload of patents out there, and
creative reading can get a patent owner fired up enough to start trying to
collect revenue on even a very bad patent -- and with hundreds of thousands
of patents in the area, some of them are likely to be very good. And in
certain venues (the Eastern District of Texas comes to mind), there's a
strong presumption that patents are valid and the courts are known to grant
damages even though a reexamination is pending. And even if the patent is
bogus and you win, you've still spent a lot of money. I'd guess your firm
spends more than my yearly revenue on defending even the lamest individual
suit.

So what I'm saying is that it doesn't matter whether you use H.264 or VP8;
you're eventually going to get sued by somebody, and it's going to be
expensive and annoying. Build in a revenue margin to plan for it. But there
is an argument to make that if you use H.264, you know that somebody with
an M in their name is going to be involved SOON unless you fork out cash in
advance, whereas if you use VP8, you can hope to maybe wait longer for the
surprise and can save your money until then.

That said, they both still make me nervous. Can't we just use tin cans and
string? ;-)

--
Dean

--e89a8fb1f86203701204bd75c557
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<br><br><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 1:47 AM, Paul E. =
Jones <span dir=3D"ltr">&lt;p</span><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
While there may be undisclosed IPR on H.264, I find it hard to believe that=
<br>
there would be much, if any, at this point. =A0Keep in mind that H.264 is<b=
r>
product of the joint effort of a bunch of people who are experts in the<br>
field. =A0I would be suspicious of any company not involved in the developm=
ent<br>
of H.264 claiming to have IPR, because somebody in the joint committee<br>
probably owns that IPR. =A0Sure, there might be something ... something ver=
y<br>
minor. =A0Perhaps.<br>
<br></blockquote><div><br></div><div>Both H.264 and =A0VP8 are motion-vecto=
r codings, right?</div><div><br></div><div>A quick search on Google Patents=
 shows about 32,400 existing motion vector video patents. What do you want =
to bet some of those apply =A0to one or the other? Or both? Or both plus H.=
263?</div>
<div><br></div><div>Just because they haven&#39;t been asserted yet doesn&#=
39;t mean they&#39;re not real. It might mean that the people that own them=
 aren&#39;t paying attention. Or weren&#39;t, but have started. Or don&#39;=
t care, but might be willing to sell their patent to someone who does.</div=
>
<div><br></div><div>The simple fact is that there are a crapload of patents=
 out there, and creative reading can get a patent owner fired up enough to =
start trying to collect revenue on even a very bad patent -- and with hundr=
eds of thousands of patents in the area, some of them are likely to be very=
 good. And in certain venues (the Eastern District of Texas comes to mind),=
 there&#39;s a strong presumption that patents are valid and the courts are=
 known to grant damages even though a reexamination is pending. And even if=
 the patent is bogus and you win, you&#39;ve still spent a lot of money. I&=
#39;d guess your firm spends more than my yearly revenue on defending even =
the lamest individual suit.</div>
<div><br></div><div>So what I&#39;m saying is that it doesn&#39;t matter wh=
ether you use H.264 or VP8; you&#39;re eventually going to get sued by some=
body, and it&#39;s going to be expensive and annoying. Build in a revenue m=
argin to plan for it. But there is an argument to make that if you use H.26=
4, you know that somebody with an M in their name is going to be involved S=
OON unless you fork out cash in advance, whereas if you use VP8, you can ho=
pe to maybe wait longer for the surprise and can save your money until then=
.</div>
<div><br></div><div>That said, they both still make me nervous. Can&#39;t w=
e just use tin cans and string? ;-)</div><div><br></div><div>--<br>Dean</di=
v></div>

--e89a8fb1f86203701204bd75c557--

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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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On Thu, Apr 5, 2012 at 1:50 PM, Paul E. Jones <paulej@packetizer.com> wrote:

>
> You're ignoring one point I made.  Those involved in the creation of H.264
> have quite likely filed IPR statements with ISO or ITU.  If some other
> company were to come out of the closet on this with something that
> blindsides the industry, the industry would fight it.  At the very least,
> whatever bit of IPR that is claimed would have to be put into perspective
> with respect to the other hundreds of patents on H.264.
>
>
This actually is a good point. Since the existing H.264 claimants gain
revenue therefrom, they are likely to act to protect their share of the
revenue. It's kind of like the local mob keeping outsiders from putting the
squeeze on businesses in their domain.

On the other hand, if I can't afford to "buy in" to their protection racket
up front, my kneecaps might be in trouble.

So do you want a nice safe Northeastern city with a godfather who will make
you an offer you can't refuse, or a Wild West free-for-all with a mix of
hard working frontiersmen, militant ranch-barons, drunken cowboys, and the
occasional marauding aborigine?

Different strokes for different folks, I think.

--
Dean

--e89a8fb1f862502a5b04bd75f5ce
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<br><br><div class=3D"gmail_quote">On Thu, Apr 5, 2012 at 1:50 PM, Paul E. =
Jones <span dir=3D"ltr">&lt;<a href=3D"mailto:paulej@packetizer.com">paulej=
@packetizer.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">
<br>
You&#39;re ignoring one point I made. =A0Those involved in the creation of =
H.264<br>
have quite likely filed IPR statements with ISO or ITU. =A0If some other<br=
>
company were to come out of the closet on this with something that<br>
blindsides the industry, the industry would fight it. =A0At the very least,=
<br>
whatever bit of IPR that is claimed would have to be put into perspective<b=
r>
with respect to the other hundreds of patents on H.264.<br><br></blockquote=
><div><br></div><div>This actually is a good point. Since the existing H.26=
4 claimants gain revenue therefrom, they are likely to act to protect their=
 share of the revenue. It&#39;s kind of like the local mob keeping outsider=
s from putting the squeeze on businesses in their domain.=A0</div>
<div><br></div><div>On the other hand, if I can&#39;t afford to &quot;buy i=
n&quot; to their protection racket up front, my kneecaps might be in troubl=
e.</div><div><br></div><div>So do you want a nice safe Northeastern city wi=
th a godfather who will make you an offer you can&#39;t refuse, or a Wild W=
est free-for-all with a mix of hard working frontiersmen, militant ranch-ba=
rons, drunken cowboys, and the occasional marauding aborigine?=A0</div>
<div><br></div><div>Different strokes for different folks, I think.</div><d=
iv><br></div><div>--<br>Dean</div><div><br></div></div>

--e89a8fb1f862502a5b04bd75f5ce--

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
References: <CAMKM2Ly-xnVEciL941uOu1Bgwc-wssZ7HNkQuBhsCcgyqfuk5Q@mail.gmail.com>	<03ac01cd120d$0ffe95f0$2ffbc1d0$@packetizer.com>	<4F7BCD1A.7020508@librevideo.org>	<03e301cd1223$153e6b60$3fbb4220$@packetizer.com>	<4F7C4FB4.4070703@librevideo.org>	<007b01cd12f7$fbcd72e0$f36858a0$@packetizer.com> <CAOHm=4tqcwmU9OJNYv4-x4GO6z4AYijd3LFcQq=q1_210FhyJg@mail.gmail.com>
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Date: Thu, 12 Apr 2012 03:09:52 -0400
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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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To be fair, Microsoft and Motorola are just having a lovers quarrel.  Do
recall that Microsoft has sued Motorola over Android, now they're suing
back. It would not surprise me if there were other spats along the way, but
I don't keep up with all of these law suits.

 

If Motorola goes knocking on the doors of the thousands of other H.264 users
out there, I might believe there is a concern and that Motorola is trying to
kill H.264.  I don't believe that's the case, though.  And I doubt that
Motorola would charge you the same $4B to use their patents, either.  Go ask
them.

 

In any case, when Google swallows Motorola, Google has an opportunity once
again to prove that it does no evil.  Google gets to manipulate those odds I
mentioned.

 

Paul

 

From: Dean Willis [mailto:dean.willis@softarmor.com] 
Sent: Thursday, April 12, 2012 2:50 AM
To: Paul E. Jones
Cc: Basil Mohamed Gohar; rtcweb@ietf.org
Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was
Re:Proposal for H.263 baseline codec]

 

 

On Thu, Apr 5, 2012 at 1:47 AM, Paul E. Jones <p

While there may be undisclosed IPR on H.264, I find it hard to believe that
there would be much, if any, at this point.  Keep in mind that H.264 is
product of the joint effort of a bunch of people who are experts in the
field.  I would be suspicious of any company not involved in the development
of H.264 claiming to have IPR, because somebody in the joint committee
probably owns that IPR.  Sure, there might be something ... something very
minor.  Perhaps.

 

Both H.264 and  VP8 are motion-vector codings, right?

 

A quick search on Google Patents shows about 32,400 existing motion vector
video patents. What do you want to bet some of those apply  to one or the
other? Or both? Or both plus H.263?

 

Just because they haven't been asserted yet doesn't mean they're not real.
It might mean that the people that own them aren't paying attention. Or
weren't, but have started. Or don't care, but might be willing to sell their
patent to someone who does.

 

The simple fact is that there are a crapload of patents out there, and
creative reading can get a patent owner fired up enough to start trying to
collect revenue on even a very bad patent -- and with hundreds of thousands
of patents in the area, some of them are likely to be very good. And in
certain venues (the Eastern District of Texas comes to mind), there's a
strong presumption that patents are valid and the courts are known to grant
damages even though a reexamination is pending. And even if the patent is
bogus and you win, you've still spent a lot of money. I'd guess your firm
spends more than my yearly revenue on defending even the lamest individual
suit.

 

So what I'm saying is that it doesn't matter whether you use H.264 or VP8;
you're eventually going to get sued by somebody, and it's going to be
expensive and annoying. Build in a revenue margin to plan for it. But there
is an argument to make that if you use H.264, you know that somebody with an
M in their name is going to be involved SOON unless you fork out cash in
advance, whereas if you use VP8, you can hope to maybe wait longer for the
surprise and can save your money until then.

 

That said, they both still make me nervous. Can't we just use tin cans and
string? ;-)

 

--
Dean


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
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style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>To be fair, Microsoft and Motorola are just having a lovers =
quarrel.&nbsp; Do recall that Microsoft has sued Motorola over Android, =
now they&#8217;re suing back. It would not surprise me if there were =
other spats along the way, but I don&#8217;t keep up with all of these =
law suits.<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><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>If Motorola goes knocking on the doors of the thousands of other =
H.264 users out there, I might believe there is a concern and that =
Motorola is trying to kill H.264.&nbsp; I don&#8217;t believe =
that&#8217;s the case, though.&nbsp; And I doubt that Motorola would =
charge you the same $4B to use their patents, either.&nbsp; Go ask =
them.<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><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>In any case, when Google swallows Motorola, Google has an opportunity =
once again to prove that it does no evil.&nbsp; Google gets to =
manipulate those odds I mentioned.<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><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Paul<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 =
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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"'> =
Dean Willis [mailto:dean.willis@softarmor.com] <br><b>Sent:</b> =
Thursday, April 12, 2012 2:50 AM<br><b>To:</b> Paul E. =
Jones<br><b>Cc:</b> Basil Mohamed Gohar; =
rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] Google VP8 Patent Grant =
for third parties [Was Re:Proposal for H.263 baseline =
codec]<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Thu, Apr 5, 2012 at 1:47 AM, Paul E. Jones =
&lt;p<o:p></o:p></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>While there may be undisclosed IPR on =
H.264, I find it hard to believe that<br>there would be much, if any, at =
this point. &nbsp;Keep in mind that H.264 is<br>product of the joint =
effort of a bunch of people who are experts in the<br>field. &nbsp;I =
would be suspicious of any company not involved in the development<br>of =
H.264 claiming to have IPR, because somebody in the joint =
committee<br>probably owns that IPR. &nbsp;Sure, there might be =
something ... something very<br>minor. =
&nbsp;Perhaps.<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Both H.264 and &nbsp;VP8 are motion-vector codings, =
right?<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>A =
quick search on Google Patents shows about 32,400 existing motion vector =
video patents. What do you want to bet some of those apply &nbsp;to one =
or the other? Or both? Or both plus H.263?<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Just because they haven't been asserted yet doesn't =
mean they're not real. It might mean that the people that own them =
aren't paying attention. Or weren't, but have started. Or don't care, =
but might be willing to sell their patent to someone who =
does.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>The simple fact is that there are a crapload of =
patents out there, and creative reading can get a patent owner fired up =
enough to start trying to collect revenue on even a very bad patent -- =
and with hundreds of thousands of patents in the area, some of them are =
likely to be very good. And in certain venues (the Eastern District of =
Texas comes to mind), there's a strong presumption that patents are =
valid and the courts are known to grant damages even though a =
reexamination is pending. And even if the patent is bogus and you win, =
you've still spent a lot of money. I'd guess your firm spends more than =
my yearly revenue on defending even the lamest individual =
suit.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>So what I'm saying is that it doesn't matter whether =
you use H.264 or VP8; you're eventually going to get sued by somebody, =
and it's going to be expensive and annoying. Build in a revenue margin =
to plan for it. But there is an argument to make that if you use H.264, =
you know that somebody with an M in their name is going to be involved =
SOON unless you fork out cash in advance, whereas if you use VP8, you =
can hope to maybe wait longer for the surprise and can save your money =
until then.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>That said, they both still make me nervous. Can't we =
just use tin cans and string? ;-)<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>--<br>Dean<o:p></o:p></p></div></div></div></div></body=
></html>
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To: Dean Willis <dean.willis@softarmor.com>
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12 apr 2012 kl. 08:50 skrev Dean Willis:

> So what I'm saying is that it doesn't matter whether you use H.264 or =
VP8; you're eventually going to get sued by somebody, and it's going to =
be expensive and annoying. Build in a revenue margin to plan for it. But =
there is an argument to make that if you use H.264, you know that =
somebody with an M in their name is going to be involved SOON unless you =
fork out cash in advance, whereas if you use VP8, you can hope to maybe =
wait longer for the surprise and can save your money until then.
>=20
> That said, they both still make me nervous. Can't we just use tin cans =
and string? ;-)
>=20

This is really bad news for Open Source. Even if we try to get a way to =
pay for licenses, our business model is far away from understandable for =
most of the syndicates. I tried with the AMR codec once and it stopped =
at initial order quantity. Anything under 10.000 licenses was not up for =
discussion.

G.729 is available on one-by-one basis which both Asterisk and =
FreeSwitch sell to users... I have no insight into how these agreements =
was worked out, but it at least indicates that someone tries to open up.

/O



From harald@alvestrand.no  Thu Apr 12 01:46:03 2012
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Subject: [rtcweb] Resolution negotiation - a contribution
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Hello folks,

after the IETF, I've attempted to gather together information about 
resolution negotiation and what we might need to do there.

I pulled together the paragraphs below - this may want to be 
incorporated in an I-D on "SDP usage in RTCWEB", or it may want to go 
somewhere else. Chairs' guidance is appreciated.

At the moment, I have it in xml2rfc format, so it should be easy to do 
cut-and-paste on it, but don't want to push it as a separate I-D unless 
that's what the chairs desire.
Note - there are some QUERY sections in there - I'm unsure what those 
should be resolved to.

Comments?

                          Harald

2.  Requirements

    The relevant requirement for video resolution negotiation from the
    RTCWEB use cases document
    [I-D.ietf-rtcweb-use-cases-and-requirements] is:

    o  F25 The browser SHOULD use encoding of streams suitable for the
       current rendering (e.g. video display size) and SHOULD change
       parameters if the rendering changes during the session.

    The resulting need is:

    o  To negotiate a maximum spatial resolution for all videos at call
       setup time

    o  To negotiate a maximum temporal resolution ("frame rate") across
       all videos at call setup time

    o  To indicate the desire of the recipient for a particular spatial
       or temporal resolution on a particular video source

    o  To indicate the intent of the sender to send a video source in a
       particular spatial or temporal resolution

    This document does not mention other requirements.


3.  SDP negotiation of video resolution

    An RTCWEB client MUST support negotiation of resolution using the
    "imageattr" attribute, as documented in [RFC6236].

    An RTCWEB client MUST support a SAR value of 1.0 (square pixels), and
    MAY choose to support only the 1.0 value of the "sar" attribute.

    The interpretation of the negotiation is that any video stream in the
    m= line containing the a=imageattr attribute will have a resolution
    within the bounds established by the negotiation.

    An RTCWEB client MUST support negotiation of the "a=framerate"
    attribute, as specified in [RFC4566] section 6.  Note that this is an
    upper bound on framerate; there is no lower bound negotiated.

    These requirements are necessary and sufficient for establishing the
    maximum spatial and temporal resolution for all videos at call setup
    time.  These bounds MAY be renegotiated over the course of the call,
    but MUST NOT be renegotiated to render any currently transmitted
    video stream out of bounds; the sending video stream MUST first be
    changed to be of a resolution that fits within both the old and the
    new bounds, or halted. <<QUERY: Is this necessary or sufficient
    restriction???>>

    An RTCWEB client MUST support per-SSRC requests for video
    resolutions, as described in
    [I-D.lennox-mmusic-sdp-source-selection].  This satisfies the
    requirement to indicate the desire of the recipient for a particular
    spatial or temporal resolution.

    We assume that the media sent from a sender to a receiver contains
    enough information inside the media format to tell what the
    resolution and framerate is. <<QUERY: Do we need to extend RFC 5576
    with an "imageattr" or "framerate" attribute in the forward direction
    instead?>>


4.  Non-SDP negotiation of video resolution

    An RTCWEB client MAY support the COP mechanism
    [I-D.westerlund-avtext-codec-operation-point] to negotiate the
    resolution of video within the limits established by the SDP
    negotiation without the need for additional SDP exchanges.


5.  Relation to WebRTC API constraint

    It is intended that the resolution negotiation be influenced by the
    constraints set by the application of either mandatory or optional
    constraints at the WebRTC API, as registered in the registry
    established by [I-D.burnett-rtcweb-constraints-registry].

    The following relationships hold for all attributes that the
    implementation intends to satisfy (note that the constraints listed
    here have NOT been registered yet):

    video-min-height >= value of imageattr y= xyrange lower bound

    video-max-height <= value of imageattr y= xyrange upper bound

    video-min-framerate is not represented in SDP

    video-max-framerate <= value of a=framerate attribute

    video-min-aspect-ratio <= value of imageattr "par=" prange lower
    bound

    video-max-aspect-ratio >= value of imageattr "par=" prange upper
    bound

    The implementation is free to increase "min" values or decrease "max"
    values (make requirements more restrictive) and add "step" in order
    to fit with its implementation restrictions. <<NOTE - check the value
    of the direction of those assumptions>>.

    Constraints specified at PeerConnection creation time are reflected
    as SDP-wide values.  Constraints specified when creating a
    MediaStream or attaching a MediaStream to a PeerConnection are
    reflected as ssrc-specific values.



From tterriberry@mozilla.com  Thu Apr 12 02:09:47 2012
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Harald Alvestrand wrote:
> time. These bounds MAY be renegotiated over the course of the call,
> but MUST NOT be renegotiated to render any currently transmitted
> video stream out of bounds; the sending video stream MUST first be
> changed to be of a resolution that fits within both the old and the
> new bounds, or halted. <<QUERY: Is this necessary or sufficient
> restriction???>>

Is this restriction intended to be imposed on the sender only? I.e., can 
the recipient still attempt renegotiation (using SDP source selection) 
of a smaller resolution than is currently allowed (e.g., because it's 
CPU-bound)?

From harald@alvestrand.no  Thu Apr 12 05:47:48 2012
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Subject: Re: [rtcweb] Resolution negotiation - a contribution
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On 04/12/2012 11:09 AM, Timothy B. Terriberry wrote:
> Harald Alvestrand wrote:
>> time. These bounds MAY be renegotiated over the course of the call,
>> but MUST NOT be renegotiated to render any currently transmitted
>> video stream out of bounds; the sending video stream MUST first be
>> changed to be of a resolution that fits within both the old and the
>> new bounds, or halted. <<QUERY: Is this necessary or sufficient
>> restriction???>>
>
> Is this restriction intended to be imposed on the sender only? I.e., 
> can the recipient still attempt renegotiation (using SDP source 
> selection) of a smaller resolution than is currently allowed (e.g., 
> because it's CPU-bound)?

What do you think it should say?

What I was thinking about was the silly state you can get into if you 
have a running 1024x768 stream, based on negotiating "width between 1000 
and 1500", and want to negotiate max resolution down to 480x360 by 
saying "width between 400 and 500".

The possibilities that may make sense on the sender side:

- Negotiate width between 400 and 1500, then change to 480x360, then 
negotiate width 400-500
- Stop the stream, negotiate width between 400 and 500, then turn on 
stream at new size
- Keep sending 1024x768 until negotiation is complete, then change at 
your leisure
- Switch to 480x360 even though it's not allowed by current parameters, 
then renegotiate

The two first look ugly, but are (as I understand) compatible with 
current standards. The two last ones seem to break the "only send what 
you have negotiated" model.

If the negotiation is receiver-initiated, it makes more sense (just 
thinking on the fly here) to say that it can be started at any time, and 
that the sender is free to change resolution to the new params at any 
time after he's decided the offer is acceptable. Perhaps he MUST do so 
before sending the answer?

(This is one particular example of Cullen's "what are the steps to 
change a codec" scenario.....)

                  Harald




From ted.ietf@gmail.com  Thu Apr 12 08:20:58 2012
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Subject: [rtcweb] Dates for upcoming interim
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After discussion between the WEBRTC and RTCWEB chairs, consulting the
doodle poll, and working through the potential for co-location with
CLUE, we have decided to set the date for the RTCWEB interim at June
12 & 13th, to be hosted by Ericsson in Stockholm.  We understand that
the WEBRTC chairs will hold their meeting on June 11th.  If CLUE
wishes to co-locate their meeting, the 14th and 15th will be
available.  This would not have been possible on the previous week,
given the national holiday in Sweden on the Wednesday of that week.

We very much appreciated the work Eric put into his analysis, and in
general we will try to prefer hubs; in this particular case, however,
a large fraction of the Western European participants are either based
in Stockholm or must fly through it to reach one of the larger hubs.
Given the availability of a host and this data, we decided to select
Stockholm for this meeting.

regards,

Ted Hardie, for the chairs.

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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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As a poor and rather nervous developer trying to avoid any slithering reptil=
es laying await in the grass, VP8 seems to be the not so obvious choice here=
. Would be better if Google could get behind this a little more as the vague=
ness could drive some to using H.263 (plegh), liability or not, most perceiv=
e 263 as royalty free.

As it stands...

VP8 +1

On 2012-04-11, at 11:29 PM, Dean Willis <dean.willis@softarmor.com> wrote:

>=20
>=20
> I said before, either direction is a gamble.  But, the odds of not getting=

> successfully sued are in your favor with H.264, in my opinion.  Here's why=
:
>=20
>=20
> That's arguable. As a small developer (and a poor one), I'm not able to pa=
y for licensing H.264. Therefore, my odds of getting sued (if anybody notice=
s my work) are pretty high. Just ask Microsoft what the odds of getting sued=
 for using H.264 are ... or the costs of trying to license it (reports are t=
hat Moto was asking $4 billion for their patents alone).
>=20
> http://www.rethink-wireless.com/article.asp?article_id=3D23154
>=20
> I'm not saying VP8 is any safer in the long run, but it's certainly easier=
 to comply with its licensing terms up front. So I might accidentally infrin=
ge, but it wouldn't be willful (at least on the essential core; there are a l=
ot of patents about there about stuff you might want to do with the video th=
at cover both codecs).
>=20
> Think of it as picking up a random snake that MIGHT be venomous, versus pi=
cking up one that's already buzzing its tail and striking at movement (caugh=
t one in my yard Monday, actually).
>=20
> The thing to remember is that you need to handle both with respect. They'r=
e still snakes. Even an unenvenomed bite can get infected. And the quiet one=
 might be a cobra instead of a rattlesnake.
>=20
> --
> Dean
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head></head><body bgcolor="#FFFFFF"><div>As a poor and rather nervous developer trying to avoid any slithering reptiles laying await in the grass, VP8 seems to be the not so obvious choice here. Would be better if Google could get behind this a little more as the vagueness could drive some to using H.263 (plegh), liability or not, most perceive 263 as royalty free.</div><div><br></div><div>As it stands...</div><div><br></div><div>VP8 +1</div><div><br>On 2012-04-11, at 11:29 PM, Dean Willis &lt;<a href="mailto:dean.willis@softarmor.com">dean.willis@softarmor.com</a>&gt; wrote:<br><br></div><div></div><blockquote type="cite"><div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
<br>
I said before, either direction is a gamble. &nbsp;But, the odds of not getting<br>
successfully sued are in your favor with H.264, in my opinion. &nbsp;Here's why:<br><br></blockquote><div><br></div><div>That's arguable. As a small developer (and a poor one), I'm not able to pay for licensing H.264. Therefore, my odds of getting sued (if anybody notices my work) are pretty high. Just ask Microsoft what the odds of getting sued for using H.264 are ... or the costs of trying to license it (reports are that Moto was asking $4 billion for their patents alone).</div>
<div><br></div><div><a href="http://www.rethink-wireless.com/article.asp?article_id=23154">http://www.rethink-wireless.com/article.asp?article_id=23154</a>
</div><div><br></div><div>I'm not saying VP8 is any safer in the long run, but it's certainly easier to comply with its licensing terms up front. So I might accidentally infringe, but it wouldn't be willful (at least on the essential core; there are a lot of patents about there about stuff you might want to do with the video that cover both codecs).</div>
<div><br></div><div>Think of it as picking up a random snake that MIGHT be venomous, versus picking up one that's already buzzing its tail and striking at movement (caught one in my yard Monday, actually).</div><div><br>
</div><div>The thing to remember is that you need to handle both with respect. They're still snakes. Even an unenvenomed bite can get infected. And the quiet one might be a cobra instead of a rattlesnake.</div><div><br>
</div><div>--</div><div>Dean</div></div>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>rtcweb mailing list</span><br><span><a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>
--Apple-Mail-140D86EF-150A-469B-8D18-8F51276FDF20--

From ekr@rtfm.com  Thu Apr 12 08:50:08 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 12 Apr 2012 08:49:26 -0700
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Subject: Re: [rtcweb] Dates for upcoming interim
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On Thu, Apr 12, 2012 at 8:20 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
> After discussion between the WEBRTC and RTCWEB chairs, consulting the
> doodle poll, and working through the potential for co-location with
> CLUE, we have decided to set the date for the RTCWEB interim at June
> 12 & 13th, to be hosted by Ericsson in Stockholm. =A0We understand that
> the WEBRTC chairs will hold their meeting on June 11th. =A0If CLUE
> wishes to co-locate their meeting, the 14th and 15th will be
> available. =A0This would not have been possible on the previous week,
> given the national holiday in Sweden on the Wednesday of that week.
>
> We very much appreciated the work Eric put into his analysis, and in
> general we will try to prefer hubs; in this particular case, however,
> a large fraction of the Western European participants are either based
> in Stockholm or must fly through it to reach one of the larger hubs.
> Given the availability of a host and this data, we decided to select
> Stockholm for this meeting.

Thanks for resolving this.

Since I'm assuming we're going to be having more of these, I'd like
to ask the chairs to publish ASAP whatever rotation they are planning to
use in the future. This would be very useful for planning purposes,
as well as to allow discussion of the proposes rota well in advance

Thanks,
-Ekr

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On 04/12/2012 10:20 AM, Ted Hardie wrote:
> After discussion between the WEBRTC and RTCWEB chairs, consulting the
> doodle poll, and working through the potential for co-location with
> CLUE, we have decided to set the date for the RTCWEB interim at June
> 12&  13th, to be hosted by Ericsson in Stockholm.  We understand that
> the WEBRTC chairs will hold their meeting on June 11th.  If CLUE
> wishes to co-locate their meeting, the 14th and 15th will be
> available.  This would not have been possible on the previous week,
> given the national holiday in Sweden on the Wednesday of that week.
>
> We very much appreciated the work Eric put into his analysis, and in
> general we will try to prefer hubs; in this particular case, however,
> a large fraction of the Western European participants are either based
> in Stockholm or must fly through it to reach one of the larger hubs.
> Given the availability of a host and this data, we decided to select
> Stockholm for this meeting.

Thanks for the update; please let us know once the details of remote 
participation (if any) are available. Airfare from the US to Stockholm 
for those dates is well north of US$1400 already, so a number of us 
won't be able to get travel dollars to attend in person.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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On 04/12/2012 02:21 AM, Olle E. Johansson wrote:

> This is really bad news for Open Source. Even if we try to get a way to pay for licenses, our business model is far away from understandable for most of the syndicates. I tried with the AMR codec once and it stopped at initial order quantity. Anything under 10.000 licenses was not up for discussion.
>
> G.729 is available on one-by-one basis which both Asterisk and FreeSwitch sell to users... I have no insight into how these agreements was worked out, but it at least indicates that someone tries to open up.

It is my understanding that the licensing arrangements that have been 
made for G.729 are no longer an option for new licensees (not offered by 
the consortium), and to my knowledge such arrangements have never been 
an option for any of the other voice/video codecs that our customers 
have asked us about (including H.264, the AMR family and others).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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On Thu, Apr 12, 2012 at 8:58 AM, Kevin P. Fleming <kpfleming@digium.com> wrote:
>
> Thanks for the update; please let us know once the details of remote
> participation (if any) are available. Airfare from the US to Stockholm for
> those dates is well north of US$1400 already, so a number of us won't be
> able to get travel dollars to attend in person.
>

We will work to get the details to you as soon as possible.  We
definitely will have options for remote participation.

regards,

Ted Hardie

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Subject: Re: [rtcweb] Resolution negotiation - a contribution
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On Thu, Apr 12, 2012 at 4:46 AM, Harald Alvestrand <harald@alvestrand.no> w=
rote:
> Hello folks,
>
> after the IETF, I've attempted to gather together information about
> resolution negotiation and what we might need to do there.
>
> I pulled together the paragraphs below - this may want to be incorporated=
 in
> an I-D on "SDP usage in RTCWEB", or it may want to go somewhere else.
> Chairs' guidance is appreciated.
>
> At the moment, I have it in xml2rfc format, so it should be easy to do
> cut-and-paste on it, but don't want to push it as a separate I-D unless
> that's what the chairs desire.
> Note - there are some QUERY sections in there - I'm unsure what those sho=
uld
> be resolved to.
>
> Comments?
>
> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Harald
>
> 2. =A0Requirements
>
> =A0 The relevant requirement for video resolution negotiation from the
> =A0 RTCWEB use cases document
> =A0 [I-D.ietf-rtcweb-use-cases-and-requirements] is:
>
> =A0 o =A0F25 The browser SHOULD use encoding of streams suitable for the
> =A0 =A0 =A0current rendering (e.g. video display size) and SHOULD change
> =A0 =A0 =A0parameters if the rendering changes during the session.
>
> =A0 The resulting need is:
>
> =A0 o =A0To negotiate a maximum spatial resolution for all videos at call
> =A0 =A0 =A0setup time
>
> =A0 o =A0To negotiate a maximum temporal resolution ("frame rate") across
> =A0 =A0 =A0all videos at call setup time
>

It is not clear if this is "per session" or "per browser" and that
should be made clear. Here is why it makes a difference:

Per session maximum video limits can give rise to a "race to the
bottom" in the case of a mixed multiparty conference.

Suppose, for example, you have 4 parties. One is on a low powered
phone with a low bandwidth connection, and
all they can handle is SIF at 384 kbps and 15 fps. The other three
have the power to handle 1048p at > 1 Mbps and 30 fps.

Restriction of all uses to the capability of the least-capable
participant is, for example, not
likely to go over well in mixed RTCWEB / CLUE Telepresence sessions.

This is a very common situation, and the solution is to do this negotiation
"per browser." Of course, in practice, this generally requires
middleware (4.3.3. in
draft-ietf-rtcweb-use-cases-and-requirements ).

Regards
Marshall

> =A0 o =A0To indicate the desire of the recipient for a particular spatial
> =A0 =A0 =A0or temporal resolution on a particular video source
>
> =A0 o =A0To indicate the intent of the sender to send a video source in a
> =A0 =A0 =A0particular spatial or temporal resolution
>
> =A0 This document does not mention other requirements.
>
>
> 3. =A0SDP negotiation of video resolution
>
> =A0 An RTCWEB client MUST support negotiation of resolution using the
> =A0 "imageattr" attribute, as documented in [RFC6236].
>
> =A0 An RTCWEB client MUST support a SAR value of 1.0 (square pixels), and
> =A0 MAY choose to support only the 1.0 value of the "sar" attribute.
>
> =A0 The interpretation of the negotiation is that any video stream in the
> =A0 m=3D line containing the a=3Dimageattr attribute will have a resoluti=
on
> =A0 within the bounds established by the negotiation.
>
> =A0 An RTCWEB client MUST support negotiation of the "a=3Dframerate"
> =A0 attribute, as specified in [RFC4566] section 6. =A0Note that this is =
an
> =A0 upper bound on framerate; there is no lower bound negotiated.
>
> =A0 These requirements are necessary and sufficient for establishing the
> =A0 maximum spatial and temporal resolution for all videos at call setup
> =A0 time. =A0These bounds MAY be renegotiated over the course of the call=
,
> =A0 but MUST NOT be renegotiated to render any currently transmitted
> =A0 video stream out of bounds; the sending video stream MUST first be
> =A0 changed to be of a resolution that fits within both the old and the
> =A0 new bounds, or halted. <<QUERY: Is this necessary or sufficient
> =A0 restriction???>>
>
> =A0 An RTCWEB client MUST support per-SSRC requests for video
> =A0 resolutions, as described in
> =A0 [I-D.lennox-mmusic-sdp-source-selection]. =A0This satisfies the
> =A0 requirement to indicate the desire of the recipient for a particular
> =A0 spatial or temporal resolution.
>
> =A0 We assume that the media sent from a sender to a receiver contains
> =A0 enough information inside the media format to tell what the
> =A0 resolution and framerate is. <<QUERY: Do we need to extend RFC 5576
> =A0 with an "imageattr" or "framerate" attribute in the forward direction
> =A0 instead?>>
>
>
> 4. =A0Non-SDP negotiation of video resolution
>
> =A0 An RTCWEB client MAY support the COP mechanism
> =A0 [I-D.westerlund-avtext-codec-operation-point] to negotiate the
> =A0 resolution of video within the limits established by the SDP
> =A0 negotiation without the need for additional SDP exchanges.
>
>
> 5. =A0Relation to WebRTC API constraint
>
> =A0 It is intended that the resolution negotiation be influenced by the
> =A0 constraints set by the application of either mandatory or optional
> =A0 constraints at the WebRTC API, as registered in the registry
> =A0 established by [I-D.burnett-rtcweb-constraints-registry].
>
> =A0 The following relationships hold for all attributes that the
> =A0 implementation intends to satisfy (note that the constraints listed
> =A0 here have NOT been registered yet):
>
> =A0 video-min-height >=3D value of imageattr y=3D xyrange lower bound
>
> =A0 video-max-height <=3D value of imageattr y=3D xyrange upper bound
>
> =A0 video-min-framerate is not represented in SDP
>
> =A0 video-max-framerate <=3D value of a=3Dframerate attribute
>
> =A0 video-min-aspect-ratio <=3D value of imageattr "par=3D" prange lower
> =A0 bound
>
> =A0 video-max-aspect-ratio >=3D value of imageattr "par=3D" prange upper
> =A0 bound
>
> =A0 The implementation is free to increase "min" values or decrease "max"
> =A0 values (make requirements more restrictive) and add "step" in order
> =A0 to fit with its implementation restrictions. <<NOTE - check the value
> =A0 of the direction of those assumptions>>.
>
> =A0 Constraints specified at PeerConnection creation time are reflected
> =A0 as SDP-wide values. =A0Constraints specified when creating a
> =A0 MediaStream or attaching a MediaStream to a PeerConnection are
> =A0 reflected as ssrc-specific values.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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To: "Kevin P. Fleming" <kpfleming@digium.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was Re:Proposal for H.263 baseline codec]
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12 apr 2012 kl. 18:09 skrev Kevin P. Fleming:

> On 04/12/2012 02:21 AM, Olle E. Johansson wrote:
>=20
>> This is really bad news for Open Source. Even if we try to get a way =
to pay for licenses, our business model is far away from understandable =
for most of the syndicates. I tried with the AMR codec once and it =
stopped at initial order quantity. Anything under 10.000 licenses was =
not up for discussion.
>>=20
>> G.729 is available on one-by-one basis which both Asterisk and =
FreeSwitch sell to users... I have no insight into how these agreements =
was worked out, but it at least indicates that someone tries to open up.
>=20
> It is my understanding that the licensing arrangements that have been =
made for G.729 are no longer an option for new licensees (not offered by =
the consortium), and to my knowledge such arrangements have never been =
an option for any of the other voice/video codecs that our customers =
have asked us about (including H.264, the AMR family and others).

And isn't that a shame. The question here for the legal department is =
wether Google's policies in regards to VP8 is enough for companies like =
you to work with it in either binary addons or Open Source code? Does it =
provide enough liability assurance?

I think that question is where we are right now.

If not, will Google try to offer a solution?

/O=

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Subject: Re: [rtcweb] Resolution negotiation - a contribution
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On 04/12/2012 06:19 PM, Marshall Eubanks wrote:
> On Thu, Apr 12, 2012 at 4:46 AM, Harald Alvestrand<harald@alvestrand.no>  wrote:
>> Hello folks,
>>
>> after the IETF, I've attempted to gather together information about
>> resolution negotiation and what we might need to do there.
>>
>> I pulled together the paragraphs below - this may want to be incorporated in
>> an I-D on "SDP usage in RTCWEB", or it may want to go somewhere else.
>> Chairs' guidance is appreciated.
>>
>> At the moment, I have it in xml2rfc format, so it should be easy to do
>> cut-and-paste on it, but don't want to push it as a separate I-D unless
>> that's what the chairs desire.
>> Note - there are some QUERY sections in there - I'm unsure what those should
>> be resolved to.
>>
>> Comments?
>>
>>                          Harald
>>
>> 2.  Requirements
>>
>>    The relevant requirement for video resolution negotiation from the
>>    RTCWEB use cases document
>>    [I-D.ietf-rtcweb-use-cases-and-requirements] is:
>>
>>    o  F25 The browser SHOULD use encoding of streams suitable for the
>>       current rendering (e.g. video display size) and SHOULD change
>>       parameters if the rendering changes during the session.
>>
>>    The resulting need is:
>>
>>    o  To negotiate a maximum spatial resolution for all videos at call
>>       setup time
>>
>>    o  To negotiate a maximum temporal resolution ("frame rate") across
>>       all videos at call setup time
>>
> It is not clear if this is "per session" or "per browser" and that
> should be made clear.
Since this is negotiation, it has to be observable across the wire, 
which means that it's negotiated per SDP session - that is, per 
PeerConnection pair, which has an SDP session between them. I don't see 
how it could be otherwise.
>   Here is why it makes a difference:
>
> Per session maximum video limits can give rise to a "race to the
> bottom" in the case of a mixed multiparty conference.
>
> Suppose, for example, you have 4 parties. One is on a low powered
> phone with a low bandwidth connection, and
> all they can handle is SIF at 384 kbps and 15 fps. The other three
> have the power to handle 1048p at>  1 Mbps and 30 fps.
>
> Restriction of all uses to the capability of the least-capable
> participant is, for example, not
> likely to go over well in mixed RTCWEB / CLUE Telepresence sessions.
I break this up differently than  you do..... we might be talking about 
the same thing still.

for the mesh conferencing situation, each participant has a direct 
connection to the "weakling".
That session will need to negotiate down to what's available. It's up to 
the app to decide whether it produces multiple streams at different 
resolutions or not (and up to the browser to figure out if it can 
support that).

For the centralized scenario, either the central unit provides 
downsampling (or stripping, if the end nodes are doing some kind of AVC 
or simulcast), or the "race to the bottom" you worry about will happen.

The word "session" has become one of the words I try to avoid using 
without a qualifier....

> This is a very common situation, and the solution is to do this negotiation
> "per browser." Of course, in practice, this generally requires
> middleware (4.3.3. in
> draft-ietf-rtcweb-use-cases-and-requirements ).
>
> Regards
> Marshall
>
>>    o  To indicate the desire of the recipient for a particular spatial
>>       or temporal resolution on a particular video source
>>
>>    o  To indicate the intent of the sender to send a video source in a
>>       particular spatial or temporal resolution
>>
>>    This document does not mention other requirements.
>>
>>
>> 3.  SDP negotiation of video resolution
>>
>>    An RTCWEB client MUST support negotiation of resolution using the
>>    "imageattr" attribute, as documented in [RFC6236].
>>
>>    An RTCWEB client MUST support a SAR value of 1.0 (square pixels), and
>>    MAY choose to support only the 1.0 value of the "sar" attribute.
>>
>>    The interpretation of the negotiation is that any video stream in the
>>    m= line containing the a=imageattr attribute will have a resolution
>>    within the bounds established by the negotiation.
>>
>>    An RTCWEB client MUST support negotiation of the "a=framerate"
>>    attribute, as specified in [RFC4566] section 6.  Note that this is an
>>    upper bound on framerate; there is no lower bound negotiated.
>>
>>    These requirements are necessary and sufficient for establishing the
>>    maximum spatial and temporal resolution for all videos at call setup
>>    time.  These bounds MAY be renegotiated over the course of the call,
>>    but MUST NOT be renegotiated to render any currently transmitted
>>    video stream out of bounds; the sending video stream MUST first be
>>    changed to be of a resolution that fits within both the old and the
>>    new bounds, or halted.<<QUERY: Is this necessary or sufficient
>>    restriction???>>
>>
>>    An RTCWEB client MUST support per-SSRC requests for video
>>    resolutions, as described in
>>    [I-D.lennox-mmusic-sdp-source-selection].  This satisfies the
>>    requirement to indicate the desire of the recipient for a particular
>>    spatial or temporal resolution.
>>
>>    We assume that the media sent from a sender to a receiver contains
>>    enough information inside the media format to tell what the
>>    resolution and framerate is.<<QUERY: Do we need to extend RFC 5576
>>    with an "imageattr" or "framerate" attribute in the forward direction
>>    instead?>>
>>
>>
>> 4.  Non-SDP negotiation of video resolution
>>
>>    An RTCWEB client MAY support the COP mechanism
>>    [I-D.westerlund-avtext-codec-operation-point] to negotiate the
>>    resolution of video within the limits established by the SDP
>>    negotiation without the need for additional SDP exchanges.
>>
>>
>> 5.  Relation to WebRTC API constraint
>>
>>    It is intended that the resolution negotiation be influenced by the
>>    constraints set by the application of either mandatory or optional
>>    constraints at the WebRTC API, as registered in the registry
>>    established by [I-D.burnett-rtcweb-constraints-registry].
>>
>>    The following relationships hold for all attributes that the
>>    implementation intends to satisfy (note that the constraints listed
>>    here have NOT been registered yet):
>>
>>    video-min-height>= value of imageattr y= xyrange lower bound
>>
>>    video-max-height<= value of imageattr y= xyrange upper bound
>>
>>    video-min-framerate is not represented in SDP
>>
>>    video-max-framerate<= value of a=framerate attribute
>>
>>    video-min-aspect-ratio<= value of imageattr "par=" prange lower
>>    bound
>>
>>    video-max-aspect-ratio>= value of imageattr "par=" prange upper
>>    bound
>>
>>    The implementation is free to increase "min" values or decrease "max"
>>    values (make requirements more restrictive) and add "step" in order
>>    to fit with its implementation restrictions.<<NOTE - check the value
>>    of the direction of those assumptions>>.
>>
>>    Constraints specified at PeerConnection creation time are reflected
>>    as SDP-wide values.  Constraints specified when creating a
>>    MediaStream or attaching a MediaStream to a PeerConnection are
>>    reflected as ssrc-specific values.
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From ted.ietf@gmail.com  Thu Apr 12 10:42:41 2012
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From: Ted Hardie <ted.ietf@gmail.com>
To: rtcweb@ietf.org, Magnus Westerlund <magnus.westerlund@ericsson.com>,  Cullen Jennings <fluffy@cisco.com>
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Subject: [rtcweb] URGENT: Dates for upcoming interim ON HOLD
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Please hold off making travel plans based on this timing.   Because
neither RAI area director will be available during this timing, they
have asked to consider shifting this timing to the previous week, June
7th and 8th.  Until the chairs have had a chance to discuss this,
please hold off on booking travel plans.

My apologies for this scramble,

Ted Hardie

On Thu, Apr 12, 2012 at 8:20 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
> After discussion between the WEBRTC and RTCWEB chairs, consulting the
> doodle poll, and working through the potential for co-location with
> CLUE, we have decided to set the date for the RTCWEB interim at June
> 12 & 13th, to be hosted by Ericsson in Stockholm. =A0We understand that
> the WEBRTC chairs will hold their meeting on June 11th. =A0If CLUE
> wishes to co-locate their meeting, the 14th and 15th will be
> available. =A0This would not have been possible on the previous week,
> given the national holiday in Sweden on the Wednesday of that week.
>
> We very much appreciated the work Eric put into his analysis, and in
> general we will try to prefer hubs; in this particular case, however,
> a large fraction of the Western European participants are either based
> in Stockholm or must fly through it to reach one of the larger hubs.
> Given the availability of a host and this data, we decided to select
> Stockholm for this meeting.
>
> regards,
>
> Ted Hardie, for the chairs.

From stewe@stewe.org  Thu Apr 12 11:20:23 2012
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From: Stephan Wenger <stewe@stewe.org>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Resolution negotiation - a contribution
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Hi Harald,
Thanks for this strawman.  I believe it should work, but I fail to see how
a two dimensional negotiation requirement (negotiating max values for
framerate and image size--which, in turn, also has two-dimensional
properties) leads to better interop than a one dimensional negotiation
(pixels per second processing requirement).  Let alone a one dimensional
negotiation with a fragmented value space and constraints imposed an
maxima and minima for both resolution and frame rates to values that
represent common operation points (such as MPEG-style levels).  Actually,
I fail to see ANY positive aspect, except perhaps that dimension/framerate
may be a bit more intuitive than processing power demands.  This goes for
both the API and the protocol.
One can argue a lot about the value of MPEG's profiles in today's software
implementation world (and people do, myself included).  But the level
concept appears to make sense to me.
As a side note, something along the lines of CLUE's simultaneous
transmission sets concept (or however that thing is currently called; I
lost track) would also be beneficial for a system that is inherently
capable of displaying multiple video signals, such as  a web browser.
Regards,
Stephan


On 4.12.2012 01:46 , "Harald Alvestrand" <harald@alvestrand.no> wrote:

>Hello folks,
>
>after the IETF, I've attempted to gather together information about
>resolution negotiation and what we might need to do there.
>
>I pulled together the paragraphs below - this may want to be
>incorporated in an I-D on "SDP usage in RTCWEB", or it may want to go
>somewhere else. Chairs' guidance is appreciated.
>
>At the moment, I have it in xml2rfc format, so it should be easy to do
>cut-and-paste on it, but don't want to push it as a separate I-D unless
>that's what the chairs desire.
>Note - there are some QUERY sections in there - I'm unsure what those
>should be resolved to.
>
>Comments?
>
>                          Harald
>
>2.  Requirements
>
>    The relevant requirement for video resolution negotiation from the
>    RTCWEB use cases document
>    [I-D.ietf-rtcweb-use-cases-and-requirements] is:
>
>    o  F25 The browser SHOULD use encoding of streams suitable for the
>       current rendering (e.g. video display size) and SHOULD change
>       parameters if the rendering changes during the session.
>
>    The resulting need is:
>
>    o  To negotiate a maximum spatial resolution for all videos at call
>       setup time
>
>    o  To negotiate a maximum temporal resolution ("frame rate") across
>       all videos at call setup time
>
>    o  To indicate the desire of the recipient for a particular spatial
>       or temporal resolution on a particular video source
>
>    o  To indicate the intent of the sender to send a video source in a
>       particular spatial or temporal resolution
>
>    This document does not mention other requirements.
>
>
>3.  SDP negotiation of video resolution
>
>    An RTCWEB client MUST support negotiation of resolution using the
>    "imageattr" attribute, as documented in [RFC6236].
>
>    An RTCWEB client MUST support a SAR value of 1.0 (square pixels), and
>    MAY choose to support only the 1.0 value of the "sar" attribute.
>
>    The interpretation of the negotiation is that any video stream in the
>    m=3D line containing the a=3Dimageattr attribute will have a resolutio=
n
>    within the bounds established by the negotiation.
>
>    An RTCWEB client MUST support negotiation of the "a=3Dframerate"
>    attribute, as specified in [RFC4566] section 6.  Note that this is an
>    upper bound on framerate; there is no lower bound negotiated.
>
>    These requirements are necessary and sufficient for establishing the
>    maximum spatial and temporal resolution for all videos at call setup
>    time.  These bounds MAY be renegotiated over the course of the call,
>    but MUST NOT be renegotiated to render any currently transmitted
>    video stream out of bounds; the sending video stream MUST first be
>    changed to be of a resolution that fits within both the old and the
>    new bounds, or halted. <<QUERY: Is this necessary or sufficient
>    restriction???>>
>
>    An RTCWEB client MUST support per-SSRC requests for video
>    resolutions, as described in
>    [I-D.lennox-mmusic-sdp-source-selection].  This satisfies the
>    requirement to indicate the desire of the recipient for a particular
>    spatial or temporal resolution.
>
>    We assume that the media sent from a sender to a receiver contains
>    enough information inside the media format to tell what the
>    resolution and framerate is. <<QUERY: Do we need to extend RFC 5576
>    with an "imageattr" or "framerate" attribute in the forward direction
>    instead?>>
>
>
>4.  Non-SDP negotiation of video resolution
>
>    An RTCWEB client MAY support the COP mechanism
>    [I-D.westerlund-avtext-codec-operation-point] to negotiate the
>    resolution of video within the limits established by the SDP
>    negotiation without the need for additional SDP exchanges.
>
>
>5.  Relation to WebRTC API constraint
>
>    It is intended that the resolution negotiation be influenced by the
>    constraints set by the application of either mandatory or optional
>    constraints at the WebRTC API, as registered in the registry
>    established by [I-D.burnett-rtcweb-constraints-registry].
>
>    The following relationships hold for all attributes that the
>    implementation intends to satisfy (note that the constraints listed
>    here have NOT been registered yet):
>
>    video-min-height >=3D value of imageattr y=3D xyrange lower bound
>
>    video-max-height <=3D value of imageattr y=3D xyrange upper bound
>
>    video-min-framerate is not represented in SDP
>
>    video-max-framerate <=3D value of a=3Dframerate attribute
>
>    video-min-aspect-ratio <=3D value of imageattr "par=3D" prange lower
>    bound
>
>    video-max-aspect-ratio >=3D value of imageattr "par=3D" prange upper
>    bound
>
>    The implementation is free to increase "min" values or decrease "max"
>    values (make requirements more restrictive) and add "step" in order
>    to fit with its implementation restrictions. <<NOTE - check the value
>    of the direction of those assumptions>>.
>
>    Constraints specified at PeerConnection creation time are reflected
>    as SDP-wide values.  Constraints specified when creating a
>    MediaStream or attaching a MediaStream to a PeerConnection are
>    reflected as ssrc-specific values.
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



From harald@alvestrand.no  Thu Apr 12 12:08:10 2012
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On 04/12/2012 08:19 PM, Stephan Wenger wrote:
> Hi Harald,
> Thanks for this strawman.  I believe it should work, but I fail to see how
> a two dimensional negotiation requirement (negotiating max values for
> framerate and image size--which, in turn, also has two-dimensional
> properties) leads to better interop than a one dimensional negotiation
> (pixels per second processing requirement).
Stephan,

I do not see this (or the requirement from the use-cases document) first 
and foremost a decoder complexity negotiation; it is a negotiation of 
how much data it is useful to send, given the recipient's intended use 
of that data.

For instance, in Google+ Hangouts, we use a similar function to make 
sure the participants who are not currently displayed in large format on 
anyone's screen only send the data needed to display the "thumbnail"; 
this represents a huge saving in bandwidth at our central servers.

It just happens that the decoding complexity of the VP8 codec is 
sufficiently linear with the number of pixels displayed that we don't 
see any advantage in encumbering the negotiation of that codec with 
extra features like "profiles" - but that's a discussion for PAYLOAD, 
not RTCWEB.


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From: Stephan Wenger <stewe@stewe.org>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Resolution negotiation - a contribution
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Subject: Re: [rtcweb] Resolution negotiation - a contribution
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On 4.12.2012 12:08 , "Harald Alvestrand" <harald@alvestrand.no> wrote:

>On 04/12/2012 08:19 PM, Stephan Wenger wrote:
>> Hi Harald,
>> Thanks for this strawman.  I believe it should work, but I fail to see
>>how
>> a two dimensional negotiation requirement (negotiating max values for
>> framerate and image size--which, in turn, also has two-dimensional
>> properties) leads to better interop than a one dimensional negotiation
>> (pixels per second processing requirement).
>Stephan,
>
>I do not see this (or the requirement from the use-cases document) first
>and foremost a decoder complexity negotiation; it is a negotiation of
>how much data it is useful to send, given the recipient's intended use
>of that data.

Then such a negotiation should be executed in addition.  Decoder cycle
requirement do matter in practical implementations.

>
>For instance, in Google+ Hangouts, we use a similar function to make
>sure the participants who are not currently displayed in large format on
>anyone's screen only send the data needed to display the "thumbnail";
>this represents a huge saving in bandwidth at our central servers.

Oh, absolutely.  I'm all in favor of resolution adaptation based on
receiver (population) requirements.  But only after an upper limit is
established (that could be based on many factors, including screen size,
connectivity, and computational resources (sic!).

>
>It just happens that the decoding complexity of the VP8 codec is
>sufficiently linear with the number of pixels displayed that we don't
>see any advantage in encumbering the negotiation of that codec with
>extra features like "profiles" - but that's a discussion for PAYLOAD,
>not RTCWEB.

True, agreed, and not what I was arguing about.

Stephan
>
>



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Subject: Re: [rtcweb] Resolution negotiation - a contribution
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On 04/12/2012 11:13 PM, Stephan Wenger wrote:
>
> On 4.12.2012 12:08 , "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>
>> On 04/12/2012 08:19 PM, Stephan Wenger wrote:
>>> Hi Harald,
>>> Thanks for this strawman.  I believe it should work, but I fail to see
>>> how
>>> a two dimensional negotiation requirement (negotiating max values for
>>> framerate and image size--which, in turn, also has two-dimensional
>>> properties) leads to better interop than a one dimensional negotiation
>>> (pixels per second processing requirement).
>> Stephan,
>>
>> I do not see this (or the requirement from the use-cases document) first
>> and foremost a decoder complexity negotiation; it is a negotiation of
>> how much data it is useful to send, given the recipient's intended use
>> of that data.
> Then such a negotiation should be executed in addition.  Decoder cycle
> requirement do matter in practical implementations.
Feel free to propose language that captures this requirement. As noted, 
my I-D fragment doesn't.


From magnus.westerlund@ericsson.com  Fri Apr 13 04:06:14 2012
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Subject: [rtcweb] Draft minutes for Wednesday session of IETF 83
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WG,

I have uploaded a first version of minutes for the first WG session in
Paris. Please review so that the minutes accurately reflects your
activities in the WG meeting session.

http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.htm


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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 ibc@aliax.net  Fri Apr 13 04:33:57 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 13 Apr 2012 13:33:35 +0200
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Subject: [rtcweb] Just browser-browser and browser-"gateway"? is this an RTCweb assumption?
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Hi, by reading RTCWeb Minutes IETF 83 [1] it seems that most of the
folks here assume that WebRTC *media* communication can be done in two
ways:

  1) browser to browser (perhaps with a TURN server)
  2) browser to gateway

So... what about browser to SIP phone and browser to XMPP/Jingle
endpoint (assuming ICE and SDES-SRTP support)?

IMHO assuming that a media gateway is required for any browser to
non-browser media communication is an artificial limitation. If so,
why does RTCweb makes use of RTP and SDP? why not creating its own
"media streaming protocol for RTCweb"?

I hope this subject is clarified and such an assumption relaxed.
Interoperability (*whithout* media gateways) is a good feature and
makes RTCweb adoption easier. Not all the people interested in RTCweb
are gateways vendors.

NOTE: I do *not* mean supporting plain and insecure RTP.

Regards.



[1] http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.htm

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

From harald@alvestrand.no  Fri Apr 13 04:39:09 2012
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On 04/13/2012 01:33 PM, IÃ±aki Baz Castillo wrote:
> Hi, by reading RTCWeb Minutes IETF 83 [1] it seems that most of the
> folks here assume that WebRTC *media* communication can be done in two
> ways:
>
>    1) browser to browser (perhaps with a TURN server)
>    2) browser to gateway
>
> So... what about browser to SIP phone and browser to XMPP/Jingle
> endpoint (assuming ICE and SDES-SRTP support)?
 From -overview section 3:

    As for all protocol and API specifications, there is no restriction
    that the protocols can only be used to talk to another browser; since
    they are fully specified, any device that implements the protocols
    faithfully should be able to interoperate with the application
    running in the browser.

Not all of us have been making the assumption you cite.
> IMHO assuming that a media gateway is required for any browser to
> non-browser media communication is an artificial limitation. If so,
> why does RTCweb makes use of RTP and SDP? why not creating its own
> "media streaming protocol for RTCweb"?
>
> I hope this subject is clarified and such an assumption relaxed.
> Interoperability (*whithout* media gateways) is a good feature and
> makes RTCweb adoption easier. Not all the people interested in RTCweb
> are gateways vendors.
>
> NOTE: I do *not* mean supporting plain and insecure RTP.

I agree, and that's the way the documents have been written "forever".
But the burden is 1) on the RTCWEB group to define what the protocols 
are, and 2) on the other devices to conform to those protocols.

           Harald


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What about browser to a media server? Does anybody want to do, for
instance, video upload from the browser to a video sharing site?
_____________
Roman Shpount


On Fri, Apr 13, 2012 at 7:33 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Hi, by reading RTCWeb Minutes IETF 83 [1] it seems that most of the
> folks here assume that WebRTC *media* communication can be done in two
> ways:
>
>  1) browser to browser (perhaps with a TURN server)
>  2) browser to gateway
>
> So... what about browser to SIP phone and browser to XMPP/Jingle
> endpoint (assuming ICE and SDES-SRTP support)?
>
> IMHO assuming that a media gateway is required for any browser to
> non-browser media communication is an artificial limitation. If so,
> why does RTCweb makes use of RTP and SDP? why not creating its own
> "media streaming protocol for RTCweb"?
>
> I hope this subject is clarified and such an assumption relaxed.
> Interoperability (*whithout* media gateways) is a good feature and
> makes RTCweb adoption easier. Not all the people interested in RTCweb
> are gateways vendors.
>
> NOTE: I do *not* mean supporting plain and insecure RTP.
>
> Regards.
>
>
>
> [1] http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.htm
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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What about browser to a media server? Does anybody want to do, for instance=
, video upload from the browser to a video sharing site? <br clear=3D"all">=
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Fri, Apr 13, 2012 at 7:33 AM, I=F1aki=
 Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net">ibc@al=
iax.net</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">
Hi, by reading RTCWeb Minutes IETF 83 [1] it seems that most of the<br>
folks here assume that WebRTC *media* communication can be done in two<br>
ways:<br>
<br>
 =A01) browser to browser (perhaps with a TURN server)<br>
 =A02) browser to gateway<br>
<br>
So... what about browser to SIP phone and browser to XMPP/Jingle<br>
endpoint (assuming ICE and SDES-SRTP support)?<br>
<br>
IMHO assuming that a media gateway is required for any browser to<br>
non-browser media communication is an artificial limitation. If so,<br>
why does RTCweb makes use of RTP and SDP? why not creating its own<br>
&quot;media streaming protocol for RTCweb&quot;?<br>
<br>
I hope this subject is clarified and such an assumption relaxed.<br>
Interoperability (*whithout* media gateways) is a good feature and<br>
makes RTCweb adoption easier. Not all the people interested in RTCweb<br>
are gateways vendors.<br>
<br>
NOTE: I do *not* mean supporting plain and insecure RTP.<br>
<br>
Regards.<br>
<br>
<br>
<br>
[1] <a href=3D"http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb=
.htm" target=3D"_blank">http://www.ietf.org/proceedings/83/minutes/minutes-=
83-rtcweb.htm</a><br>
<font color=3D"#888888"><br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<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>
</font></blockquote></div><br>

--000e0cdfdb520f590104bd8dec1a--

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2012/4/13 Harald Alvestrand <harald@alvestrand.no>:
> But the burden is 1) on the RTCWEB group to define what the protocols are=
,
> and 2) on the other devices to conform to those protocols.

Right. But the more *new* RTCweb media specifications are required,
the harder it will interoperate with other protocols.

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

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On 04/13/2012 01:39 PM, Roman Shpount wrote:
> What about browser to a media server? Does anybody want to do, for 
> instance, video upload from the browser to a video sharing site?
Not necessarily real-time, so may be out of scope for RTCWEB, but people 
are definitely interested.

For those who want to read about scenarios that don't necessarily 
involve real-time communication, but will most probably involve 
getUserMedia, I recommend this document:

https://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/scenarios.html

It's being used to drive requirements on the getUserMedia API in the W3C.

                      Harald


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There's a W3C group that  is interested in this, in the form of speech =
enabling HTML 5 pages (i.e. adding text-to-speech and  speech-to-text =
capabilities).  We are hoping to use the rtcweb work as part of this.  =
We'd have a browser one end, a media/IVR server on the other, and voice =
and data channels.  There's been no specific mention of  video so far, =
but I don't see any reason why it couldn't  be added.

=20

-          Jim

=20

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Roman Shpount
Sent: Friday, April 13, 2012 7:39 AM
To: I=F1aki Baz Castillo
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Just browser-browser and browser-"gateway"? is =
this an RTCweb assumption?

=20

What about browser to a media server? Does anybody want to do, for =
instance, video upload from the browser to a video sharing site?=20
_____________
Roman Shpount



On Fri, Apr 13, 2012 at 7:33 AM, I=F1aki Baz Castillo <ibc@aliax.net> =
wrote:

Hi, by reading RTCWeb Minutes IETF 83 [1] it seems that most of the
folks here assume that WebRTC *media* communication can be done in two
ways:

 1) browser to browser (perhaps with a TURN server)
 2) browser to gateway

So... what about browser to SIP phone and browser to XMPP/Jingle
endpoint (assuming ICE and SDES-SRTP support)?

IMHO assuming that a media gateway is required for any browser to
non-browser media communication is an artificial limitation. If so,
why does RTCweb makes use of RTP and SDP? why not creating its own
"media streaming protocol for RTCweb"?

I hope this subject is clarified and such an assumption relaxed.
Interoperability (*whithout* media gateways) is a good feature and
makes RTCweb adoption easier. Not all the people interested in RTCweb
are gateways vendors.

NOTE: I do *not* mean supporting plain and insecure RTP.

Regards.



[1] http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.htm

--
I=F1aki Baz Castillo
<ibc@aliax.net>
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

=20


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>There&#8217;s a W3C group that=A0 is interested in this, in the form =
of speech enabling HTML 5 pages (i.e. adding text-to-speech and=A0 =
speech-to-text capabilities). =A0We are hoping to use the rtcweb work as =
part of this.=A0 We&#8217;d have a browser one end, a media/IVR server =
on the other, and voice and data channels.=A0 There&#8217;s been no =
specific mention of=A0 video so far, but I don&#8217;t see any reason =
why it couldn&#8217;t=A0 be added.<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><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Jim<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-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-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Roman Shpount<br><b>Sent:</b> Friday, April 13, 2012 7:39 =
AM<br><b>To:</b> I=F1aki Baz Castillo<br><b>Cc:</b> =
rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] Just browser-browser and =
browser-&quot;gateway&quot;? is this an RTCweb =
assumption?<o:p></o:p></span></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>What about browser to a media server? =
Does anybody want to do, for instance, video upload from the browser to =
a video sharing site? <br clear=3Dall>_____________<br>Roman =
Shpount<br><br><o:p></o:p></p><div><p class=3DMsoNormal>On Fri, Apr 13, =
2012 at 7:33 AM, I=F1aki Baz Castillo &lt;<a =
href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal>Hi, by reading RTCWeb Minutes =
IETF 83 [1] it seems that most of the<br>folks here assume that WebRTC =
*media* communication can be done in two<br>ways:<br><br>&nbsp;1) =
browser to browser (perhaps with a TURN server)<br>&nbsp;2) browser to =
gateway<br><br>So... what about browser to SIP phone and browser to =
XMPP/Jingle<br>endpoint (assuming ICE and SDES-SRTP =
support)?<br><br>IMHO assuming that a media gateway is required for any =
browser to<br>non-browser media communication is an artificial =
limitation. If so,<br>why does RTCweb makes use of RTP and SDP? why not =
creating its own<br>&quot;media streaming protocol for =
RTCweb&quot;?<br><br>I hope this subject is clarified and such an =
assumption relaxed.<br>Interoperability (*whithout* media gateways) is a =
good feature and<br>makes RTCweb adoption easier. Not all the people =
interested in RTCweb<br>are gateways vendors.<br><br>NOTE: I do *not* =
mean supporting plain and insecure =
RTP.<br><br>Regards.<br><br><br><br>[1] <a =
href=3D"http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.htm"=
 =
target=3D"_blank">http://www.ietf.org/proceedings/83/minutes/minutes-83-r=
tcweb.htm</a><br><span style=3D'color:#888888'><br>--<br>I=F1aki Baz =
Castillo<br>&lt;<a =
href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<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></span>=
<o:p></o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
------_=_NextPart_001_01CD1973.FFC20721--

From marshall.eubanks@gmail.com  Fri Apr 13 11:14:59 2012
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From: Marshall Eubanks <marshall.eubanks@gmail.com>
To: rtcweb@ietf.org
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Subject: [rtcweb] RTCWEB / CLUE Use Cases for Browser - Telepresence Interoperability
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I am posting this to both CLUE and RTCWEB. I am not sure which (if
either) WG would want to adopt it, but I think both should see it. If
this gains traction, I
will make sure both WG are kept abreast of developments.

Regards
Marshall

RTCWEB / CLUE Use Cases for Browser - Telepresence Interoperability

References :

RTCWEB Use Cases and Requirements :
http://www.ietf.org/id/draft-ietf-rtcweb-use-cases-and-requirements-07.txt

CLUE :  Use Cases for Telepresence Multi-streams
http://www.ietf.org/id/draft-ietf-clue-telepresence-use-cases-02.txt

What follows is a first-cut at use cases for interoperability between
a browser based videoconferencing system and Telepresence
videoconferencing. This is an attempt to describe what should be done,
without specifying in detail how. I have seen all of the below in the
field, with Cases 1-T, 1-R.c and 2-R.b being the most widely
supported. These cases are similar to the 4.2.10 use case in
ID.draft-ietf-rtcweb-use-cases-and-requirements, but of course in the
telepresence scenario there is a set of CLUE semantics overlaid on the
basic display of screens. In this scenario, the browser is more likely
to require higher level protocol changes than the telepresence units
but, of course, the telepresence units or middleware MUST be able to
accommodate RTCWEB codec and signaling choices.

Base scenario. There are one or more browser based videoconferencing
systems and one or more telepresence systems (each with at least two
cameras and two screens) participating in an immersive-telepresence
videoconferencing session. This may or may not require middleware (a
server or MCU) to accomplish, both use cases with and without
middleware SHOULD be supported. The telepresence units themselves are
assumed to use CLUE and underlying protocols to decide suitable bit
rates, resolutions, etc., between themselves; intra-telepresence
negotiations are not in scope for this text. In telepresence, "screen"
and "stream" are used more or less interchangeably, and will be done
so here. In the base scenario, there are thus at least 3 screens
(streams); conferences with 9 or more streams are not unusual, and
conferences with dozens or even hundreds of streams are a commercial
reality.

It is useful to divide use cases between Transmission (T, what the
browser sends) and Reception (R, what it receives). Browser based
units SHOULD be able to decide between transmission and reception use
cases independently, depending on capabilities and user choices.

Case 1 : The one screen use case.

In Case 1, it is possible for a CLUE Telepresence session and a
browser to interact without any knowledge of CLUE on the part of the
browser (i.e., by a telepresence unit or middleware  acting as a
single point videoconferencing unit). Such "Clueless" conferencing
sessions are not in scope for this text.

Case 1-Transmission : The browser sends one screen at the screen
characteristics of the participating telepresence units or, failing
that, at the maximum resolution and bandwidth it is capable or
authorized to do so. Likewise, audio SHOULD be sent at the bit rate
and using the codec negotiated by the CLUE telepresence session. Case
1 audio SHOULD be sent in mono. The browser MUST be able to use CLUE
to negotiate these characteristics (which may change with time), and
SHOULD be able to provide whatever meta-data is required by CLUE
(e.g., the user's name or location).

The browser may be one complete screen in the remote telepresence
units. If care is taken by the browser user and software (for example,
by matching head size and camera angle compared to the standards of
the participating telepresence units, together with sufficient audio
and video quality and resolution), the browser image and sound may
approach immersive telepresence on the remote ends. (It would be
useful if the system provided feedback or even automated zoom in/out
to help with this adjustment for remote immersion. I have seen this
done manually, but this has not to my knowledge been discussed to date
in CLUE.)

In the case of lower quality browser transmitted video, the receiving
telepresence units may chose to display such videos in a composited
form, with multiple browser transmissions sharing one screen. (This is
common with low quality video from multiple browser-based
participants.)

Case 1-Reception : The browser displays one screen for all of the
remote telepresence units. The browser receives one screen at the
screen characteristics of the participating telepresence units or,
failing that, at the maximum resolution and bandwidth it is capable
of. Likewise, it SHOULD be able to receive audio at the bit rate and
using the codec negotiated by the CLUE telepresence session. The
browser MUST be able to negotiate these characteristics (which may
change with time) using CLUE.

In all Case 1-R sub cases, the browser MAY also display "thumbnail"
(i.e., substantially reduced) images of other screens. These
thumbnails might be shown for all screens, or for recently active
screens (e.g., the last speaker), or for some static choice (e.g., the
conference chair). The browser SHOULD be able to signal its desire /
need to receive such thumbnails.

Sub Case 1-R a :  Static. The browser displays one screen only, as
selected by the user or by some other method. (A simultaneous
translator, for example, may prefer only to see the screen attached to
their assigned speaker, regardless of whether or not they are
speaking.)

Sub Case 1-R b : Switching. The browser displays the "active" stream,
typically of the current speaker. The browser MUST be able to switch
rapidly between resolutions and other stream configuration choices,
say if the active screen switches between a telepresence unit and
another browser's transmission.

Sub Case 1-R c : Compositing. The browser displays one screen,
consisting of a static compositing of all (or conceivably a selection)
of the other screens. This MAY indicate the active speaker (say, by
highlighting them) and MAY display composited metadata (such as the
attendees in each sub-screen, or their location). (If this display is
an NxM matrix of equal size sub-screens, this display type is
frequently called "Hollywood Squares," but other choices are
possible.) In general this will be a static screen assignment, which
MAY change with time (e.g., as participants come and go from the
conference). This compositing could be done by the browser, but in
current practice will most likely be done by either a telepresence
unit or by middleware.

Case 2 : Multiple Screens.  The browser sends and/or receives multiple
screens at the maximum resolution and bandwidth it is capable or
authorized to do so. (This case is NOT intended to cover "thumbnail"
sub screens, which may be sent or received in this case as well.)

Case 2-Transmission. The browser has access to multiple cameras and
sends multiple images to remote participants. If care is taken by the
browser user / software, the browser images may approach immersive
telepresence on the remote ends. For this to happen in the
multi-screen transmission case, the browser MUST be able to fully
participate in CLUE protocol negotiations, identifying, for example,
which screen is left, center and right in the case of a three camera
transmission.

In Case 2-T, the browser SHOULD send stereo or multi-channel audio and
SHOULD format any such audio to match the transmitted screens, e.g.,
with the left audio corresponding to the left screen. Such audio
choices, if made, MUST be indicated in the CLUE configuration setup.

Case 2-Reception. The browser displays multiple screens based on the
multiple screens available from other telepresence participants.

Sub Case 2-R a : Switched compositing. In this use case typically one
composited screen is shown, of some or all of the telepresence
participants, together with one or more full-sized screens of the
active participants (or, at times, of one static screen. Note that the
composited screens may have higher sub-screen resolution than for
thumbnails, or may have very different aspect ratios than is typical
for thumbnails. For example, a three screen telepresence unit, with an
individual screen aspect ratio of 16:9, may transmit a 48:9 aspect
ratio composite of all its screens in the proper display order.
Several such composites could be combined into a single 16:9 aspect
ratio to make the composited screen in this use case. This use case
includes multiple screens for selected speakers, for example a
composited screen to the left, the conference moderator in larger
resolution in the center, and (also in larger resolution) the current
or previous speaker (should the moderator be currently speaking) on
the right. In Case 2-R a, the browser SHOULD display screens and use
stereo or multi-channel audio conforming to the CLUE configuration,
with (in the above example), the conference moderator being assigned
the center audio channel and the current or previous speaker the right
audio channel.

Sub Case 2-R b : Telepresence mimicry. The browser displays screens as
if it were a telepresence unit involved in the telepresence session.
The browser MUST be able to fully participate in the CLUE protocol,
displaying, for example, left screens on the left and center screens
on the center. In general, these screens will be displayed at the same
aspect ratio, but with lower size / resolution, than in the full
telepresence session. Experience shows that, for participants who are
used to multi-screen telepresence, even lesser quality telepresence
mimicry is very popular and RTCWEB / CLUE SHOULD support this full
telepresence functionality.  At the high end of browser capabilities,
the full resolution and bit rates available could  be used to approach
a truly immersive telepresence at the browser receive side; this is
likely to become more common in the future.

In Case 2-R b, the browser SHOULD display screens and stereo or
multi-channel audio conforming to the CLUE configuration, with, e.g.,
the left audio channel corresponding to the left screen, etc.

From ibc@aliax.net  Fri Apr 13 13:49:55 2012
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Date: Fri, 13 Apr 2012 22:49:53 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] RTCWEB / CLUE Use Cases for Browser - Telepresence Interoperability
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Hi, an easy question... What does it mean CLUE?

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<p>Hi, an easy question... What does it mean CLUE?</p>

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http://datatracker.ietf.org/wg/clue/charter/

On Fri, Apr 13, 2012 at 4:49 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:
> Hi, an easy question... What does it mean CLUE?

From ted.ietf@gmail.com  Fri Apr 13 15:38:42 2012
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Subject: [rtcweb] UPDATED Draft minutes for Wednesday session of IETF 83
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The proceedings seem to be set up to prefer a single set of minutes,
rather than one per session.  I've uploaded a set now that includes
the ones Magnus originally put up, plus minutes from the second
session.  Many thanks to all of our note takers for their hard work.

Please review the minutes and send corrections,

thanks,

Ted Hardie

On Fri, Apr 13, 2012 at 4:06 AM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> WG,
>
> I have uploaded a first version of minutes for the first WG session in
> Paris. Please review so that the minutes accurately reflects your
> activities in the WG meeting session.
>
> http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.htm
>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| 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

From randell-ietf@jesup.org  Fri Apr 13 19:26:51 2012
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Subject: Re: [rtcweb] Interaction between MediaStream API and signaling
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A reply I hadn't sent...

On 4/2/2012 9:55 AM, Stefan Hakansson LK wrote:
> On 04/02/2012 05:16 AM, Justin Uberti wrote:
>>>> Yes. Just like in SIP. And so when you send an OFFER (or
>> modified
>> re-OFFER), you must be ready to receive data per that offer
>> even if no
>> ANSWER has been received - just like in SIP. And if its a
>> re-offer, you
>> need to accept the old, and accept the new (though you could
>> probably
>> use reception of obviously new-OFFER media to turn off
>> decoding/rendering old-OFFER in preparation for the ANSWER).
>>
>> The flip side of this is the responder has to infer when the
>> sender
>> switches over to the result of the ANSWER from the media.
>> For example:
>>
>> A B
>> <--- H.261 --->
>> re-OFFER(VP8) --->
>> <-- ANSWER(VP8) (delayed in reception)
>> <-----------VP8 (A should infer that B ANSWERed
>> and accepted VP8)
>> ----------> H.261
>> <-- ANSWER(VP8) (received)
>> <--------VP8----------> (B should infer by reception of
>> VP8 that ANSWER
>> was received)
>>
>> (Personally, I hate inferences, but without a 3 (or 4) way
>> handshake,
>> you have to). If you switches of codecs are staged, then
>> this isn't
>> (much) of a problem. Either leave old codec on the list, or
>> leave it on
>> the list until accept, and then re-OFFER to remove the
>> un-used codec.
>>

[ Stefan responded: ]

>> I think I understand what you mean, and this would work fine as
>> long as
>> you just switch codecs that are used in already set-up MediaStreams.

Ah, but how do you know which stream the packets are for:
in bundle, the packets are designated by SSRC, but without the ANSWER 
you can't figure it out, and on ANSWER the SSRCs could change.  Or if 
you change from non-bundle to bundle (admittedly, it's unclear when one 
would do so, but we're discussing a protocol, and that's not disallowed 
I assume).

We have to consider, since we're designing a protocol, not only the 
preferred and likely negotiations, but all possible negotiations. 
Painful, but true.

>> This may have been a very bad example. Probably you can tell them
>> apart on the SSRC. But even so, the A browser won't know what the
>> VP8 stream (if it has an unknown SSRC) represents without receiving
>> the ANSWER.
>>
>>
>> I think this is only an issue if you decide to add streams in the
>> ANSWER. But even so, eventually the ANSWER arrives and you can start
>> demuxing/decoding appropriately.
>
> Yes, I agree to that it is only an issue if you add streams in the
> answer. Perhaps it is a model we should move away from - but that is the
> model used in the basic examples of the JSEP draft.

Even if you don't add streams you could change the Bundle; or the SSRC's 
could change messing up demux until ANSWER, etc.

> I think there is a risk of clipping if you start sending immediately,
> but can only start demuxing/decoding once an answer is received.

I would agree.



-- 
Randell Jesup
randell-ietf@jesup.org

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Subject: Re: [rtcweb] DTLS-SRTP with end-to-end security: Short Authentication String
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On 4/6/12 6:57 AM, Randell Jesup wrote:
> tl;dr:  I believe SAS should be supported by the browser (or non-browser
> app/device) UI, and I believe that's the current plan though perhaps
> it's not written as a MUST.  I am much more hesitant to endorse
> key-chaining due to usability issues.

Well, that's an element of improvement that maybe strongly desiderable,
as it would enable the current DTLS-SRTP standard to provide end-to-end
security.

Otherwise, without independent fingerprint verification method
formalized and mandatory.

The browser would just need to report whenever the communication has:
- end-to-site security (trusting google/signaling provider)
- end-to-end security (verifying it independently)

The JS API would just allow the calling app to specify whenever the
media security authentication would go trough:
- without verification
- with a third party IDP
- with end-to-end authentication (via user to user authentication)

This would really be a strong security improvement to the WebRTC
standard as it would satisfy also end-to-end encryption needs.

And user would be aware of the level of security of their call in a
simple and user-friendly way.

That way the level of security can be evaluated by the end-user, with
security options to be provided by the calling app provider by the
signaling service.

This seems to me quite a reasonable, not so complicated, but very
*powerful* security improvement as it bring end-to-end security to the
standard.

Otherwise we would never be able to say that "WebRTC provider end-to-end
security"
.
>> I would like to bring the attention of the WG to provide a MANDATORY
>> requirement to enable SAS verification.
> 
> I think the question was asked at IETF (paraphrased): will SAS be
> available through the UI in some manner?  Yes.

This would be really cool.
It's just that current IETF internet-draft does not seems to reflect
this statement (but i maybe wrong, in case point me to the right
sentence highlighting the support for SAS)

Fabio

From stefan.lk.hakansson@ericsson.com  Mon Apr 16 00:01:39 2012
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Date: Mon, 16 Apr 2012 09:01:36 +0200
From: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
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Subject: Re: [rtcweb] Interaction between MediaStream API and signaling
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On 04/14/2012 04:22 AM, Randell Jesup wrote:
> A reply I hadn't sent...
>
> On 4/2/2012 9:55 AM, Stefan Hakansson LK wrote:
>> On 04/02/2012 05:16 AM, Justin Uberti wrote:
>>>>> Yes. Just like in SIP. And so when you send an OFFER (or
>>> modified
>>> re-OFFER), you must be ready to receive data per that offer
>>> even if no
>>> ANSWER has been received - just like in SIP. And if its a
>>> re-offer, you
>>> need to accept the old, and accept the new (though you could
>>> probably
>>> use reception of obviously new-OFFER media to turn off
>>> decoding/rendering old-OFFER in preparation for the ANSWER).
>>>
>>> The flip side of this is the responder has to infer when the
>>> sender
>>> switches over to the result of the ANSWER from the media.
>>> For example:
>>>
>>> A B
>>> <--- H.261 --->
>>> re-OFFER(VP8) --->
>>> <-- ANSWER(VP8) (delayed in reception)
>>> <-----------VP8 (A should infer that B ANSWERed
>>> and accepted VP8)
>>> ---------->  H.261
>>> <-- ANSWER(VP8) (received)
>>> <--------VP8---------->  (B should infer by reception of
>>> VP8 that ANSWER
>>> was received)
>>>
>>> (Personally, I hate inferences, but without a 3 (or 4) way
>>> handshake,
>>> you have to). If you switches of codecs are staged, then
>>> this isn't
>>> (much) of a problem. Either leave old codec on the list, or
>>> leave it on
>>> the list until accept, and then re-OFFER to remove the
>>> un-used codec.
>>>
>
> [ Stefan responded: ]
>
>>> I think I understand what you mean, and this would work fine as
>>> long as
>>> you just switch codecs that are used in already set-up MediaStreams.
>
> Ah, but how do you know which stream the packets are for:
> in bundle, the packets are designated by SSRC, but without the ANSWER
> you can't figure it out, and on ANSWER the SSRCs could change.

If I read the spec right, the ANSWER can't change the SSRCs for streams 
going offerer -> answerer (but I agree to the first point you are making).



> Or if
> you change from non-bundle to bundle (admittedly, it's unclear when one
> would do so, but we're discussing a protocol, and that's not disallowed
> I assume).

Right.

>
> We have to consider, since we're designing a protocol, not only the
> preferred and likely negotiations, but all possible negotiations.
> Painful, but true.

Agree. IMO, there is a lot that is unclear, and need to be speced out. 
JSEP basically allows you to modify anything at any time, and to be able 
to build interoperable implementations we need to define what should happen.

>
>>> This may have been a very bad example. Probably you can tell them
>>> apart on the SSRC. But even so, the A browser won't know what the
>>> VP8 stream (if it has an unknown SSRC) represents without receiving
>>> the ANSWER.
>>>
>>>
>>> I think this is only an issue if you decide to add streams in the
>>> ANSWER. But even so, eventually the ANSWER arrives and you can start
>>> demuxing/decoding appropriately.
>>
>> Yes, I agree to that it is only an issue if you add streams in the
>> answer. Perhaps it is a model we should move away from - but that is the
>> model used in the basic examples of the JSEP draft.
>
> Even if you don't add streams you could change the Bundle; or the SSRC's
> could change messing up demux until ANSWER, etc.
>
>> I think there is a risk of clipping if you start sending immediately,
>> but can only start demuxing/decoding once an answer is received.
>
> I would agree.
>
>
>


From magnus.westerlund@ericsson.com  Mon Apr 16 01:37:02 2012
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Date: Mon, 16 Apr 2012 10:36:59 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Consensus call regarding media security
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WG, EKR

The deadline for this call of consensus has ended. I note that there has
been discussion of a number of details and additional views on the
actual consensus statement made. Some supporting the consensus from the
meeting room and some opposing it. In the end I determine that the two
consensus statements are confirmed:

 The first was that there was overwhelming consensus that all RTP
 packets SHALL be protected by SRTP.

 Secondly that no one objected against making DTLS-SRTP a mandatory to
 implement and the default keying mechanism. Additional mechanisms are
 not precluded.

>From the discussion in the thread I would note the following:

1) Null cipher for encryption is strongly desired but an actual proposal
is needed for how to express this.

2) Details of the SRTP and DTLS-SRTP usage in this WG needs to be
defined. This is for continued discussion. I hope our editor of the
security architecture can make a proposal to the WG to consider.

3) The discussion of additional keying mechanisms such as Security
Descriptions and Encrypted Key Transport will continue.

Cheers

Magnus

On 2012-03-28 16:50, Magnus Westerlund wrote:
> WG,
> 
> In todays RTCWEB WG meeting there was discussion around media security
> mechanism. In this meeting there was some clear consensus in the
> meeting which we would like to confirm on the list.
> 
> The first was that there was overwhelming consensus that all RTP
> packets SHALL be protected by SRTP.
> 
> Secondly that no one objected against making DTLS-SRTP a mandatory to
> implement and the default keying mechanism. Additional mechanisms are
> not precluded.
> 
> WG participants may state their position regarding these consensus calls
> until 12th of April when the chairs will declare the final consensus. If
> you where present in the meeting room and comment on this, please
> indicate that.
> 
> Best Regards
> 
> Magnus Westerlund
> For the WG chairs
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Mon Apr 16 01:41:55 2012
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Subject: Re: [rtcweb] UPDATED Draft minutes for Wednesday session of IETF 83
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On 2012-04-14 00:38, Ted Hardie wrote:
> The proceedings seem to be set up to prefer a single set of minutes,
> rather than one per session.  I've uploaded a set now that includes
> the ones Magnus originally put up, plus minutes from the second
> session.  Many thanks to all of our note takers for their hard work.
> 
> Please review the minutes and send corrections,

A link to the version containing both sessions are:

http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.txt

Cheers

Magnus Westerlund

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From ekr@rtfm.com  Mon Apr 16 04:58:26 2012
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On Mon, Apr 16, 2012 at 4:36 AM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> WG, EKR
>
> The deadline for this call of consensus has ended. I note that there has
> been discussion of a number of details and additional views on the
> actual consensus statement made. Some supporting the consensus from the
> meeting room and some opposing it. In the end I determine that the two
> consensus statements are confirmed:
>
> =A0The first was that there was overwhelming consensus that all RTP
> =A0packets SHALL be protected by SRTP.
>
> =A0Secondly that no one objected against making DTLS-SRTP a mandatory to
> =A0implement and the default keying mechanism. Additional mechanisms are
> =A0not precluded.
>
> From the discussion in the thread I would note the following:
>
> 1) Null cipher for encryption is strongly desired but an actual proposal
> is needed for how to express this.
>
> 2) Details of the SRTP and DTLS-SRTP usage in this WG needs to be
> defined. This is for continued discussion. I hope our editor of the
> security architecture can make a proposal to the WG to consider.

Acknowledged.

I will do so.

-Ekr

From harald@alvestrand.no  Mon Apr 16 05:01:19 2012
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Subject: Re: [rtcweb] UPDATED Draft minutes for Wednesday session of IETF 83
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Thanks for these!

The only serious comments I have are on the codec discussion near the 
end; otherwise, there are grammar nits.

Notes:
- Please convert to ASCII. Quotes and non-ASCII characters are broken.
- "One attack by claim to link to something, but what you really connect 
to is some other site"
Grammar issue: by claim -> by claiming
- "In Jingle to enable early response with the ICE candidates and the 
username and password are anyway needed." -> could not parse this 
sentence. Please reword.
- One instance of "Collin Perkins" (misspelled name)
- "straight forward" -> "straightforward"
- "Regarding the key-exchange and the round trip time. We will
encounter cases, like the trombone where the signaling server is far
away and the media peer is close." - the first full stop should be a 
comma for grammatical correctness.
- "no overhead in a waste majority of cases." -> "no overhead in the 
vast majority of cases"
- "The chairs declared No Consensus between these two options due to very
evenly strength hums." - grammatically, it needs to be "even strength 
hums"; I would formulate it as "approximately even strength hums", given 
that I don't remember them as *that* even (but I was not at the front of 
the room).

- "Harald: I hope we can limit the number of cases where SDP manipulation
is limited." - I certainly hope I said "needed" as the last word.
- "Harald regards hints as steering wheel and brake pedal,
while SDP is plugging in to engine control. Hopes can do what we need
via hints." - "in to" -> "into"; "Hopes can do" -> "He hopes that we can 
do".

- "you are talking of SCT/DTLS/UDP." -> SCTP/DTLS/UDP
- "Tim (Mozilla)" -> Tim Terriberry
- Tim P -> Tim Panton

"Cullen: raise your hand in favor of H.261
                                     H.264
                                     VP8"

The numbers (or proportions) need to be part of the minutes, and the 
minutes need to make it clear that this was a straw poll, not a hum for 
consensus (Ted made that very clear, but it's not clear from the minutes.)

Allistair ??: The discussion was the same in H.323, and G.711 was
selected as audio codec because of IPR free. None believes it would be
useful, but this stopped the discussions.

I believe this was followed by a remark saying a huge amount of current 
traffic is G.711 or worse.


On 04/16/2012 10:41 AM, Magnus Westerlund wrote:
> On 2012-04-14 00:38, Ted Hardie wrote:
>> The proceedings seem to be set up to prefer a single set of minutes,
>> rather than one per session.  I've uploaded a set now that includes
>> the ones Magnus originally put up, plus minutes from the second
>> session.  Many thanks to all of our note takers for their hard work.
>>
>> Please review the minutes and send corrections,
> A link to the version containing both sessions are:
>
> http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.txt
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 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
>


From Henry.Lum@genesyslab.com  Mon Apr 16 08:52:58 2012
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It's unclear to me what the purpose of SDP_PRANSWER is really for and
why it needs to be there. Originally I thought it was meant for early
media, but section 6.1.4 paragraph 3 says that it should not result in
starting of media transmission. Is there a reason to provide an answer
before media is ready to be transmitted? There are separate handles for
processing ICE candidates so I don't see a need to have provisional
answers.=20

=20

I am not proposing PRANSWER for handling early media though, since I
believe early media is more of an application level issue (it's more
like late-billing on the application side) and should be considered as a
separate offer/answer if needed.

=20

Regards,

Henry=20


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<p class=3DMsoNormal>It&#8217;s unclear to me what the purpose of =
SDP_PRANSWER is
really for and why it needs to be there. Originally I thought it was =
meant for early
media, but section 6.1.4 paragraph 3 says that it should not result in =
starting
of media transmission. Is there a reason to provide an answer before =
media is
ready to be transmitted? There are separate handles for processing ICE =
candidates
so I don&#8217;t see a need to have provisional answers. <o:p></o:p></p>

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

<p class=3DMsoNormal>I am not proposing PRANSWER for handling early =
media though,
since I believe early media is more of an application level issue =
(it&#8217;s
more like late-billing on the application side) and should be considered =
as a
separate offer/answer if needed.<o:p></o:p></p>

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

<p class=3DMsoNormal>Regards,<o:p></o:p></p>

<p class=3DMsoNormal>Henry <o:p></o:p></p>

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The CLUE and WebRTC groups are making plans based on WebRTC meeting June =
12 and 13 in Stockholm. The chairs believe that picking a different time =
or locations at this point will not be beneficial so we plan to try and =
confirm that date.=20

I'd like to try and answer the question about locations of RTCWeb =
interim meetings after the ones currently planned.

First of all, the chairs would are going to declare that there is WG =
consensus for a 1:1:1 meeting rotation between west coast of north =
america, europe, and east coast of north america and plan to run =
approximately equal number of meetings in these locations.  In counting =
the number of past meetings in a given region, we will include all the =
face to face RTCWeb and WebRTC meetings as this work is closely joined =
and a large number of the participants travel to both sets of meetings. =
The face to face meetings that happen at the main IETF or W3C meetings =
are included in this count. Collocated meetings will be counted just =
once as they only require one set of travel to that location. =20

Cullen  <RTCWeb co-chair>





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Just one clarification - I thought the W3C WebRTC group planned to meet on
June 11th and then the IETF RTCWEB WG on June 12-13?

The CLUE WG meeting still plans to meet on June 7-8.

Regards,
Mary.

On Mon, Apr 16, 2012 at 2:45 PM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> The CLUE and WebRTC groups are making plans based on WebRTC meeting June
> 12 and 13 in Stockholm. The chairs believe that picking a different time or
> locations at this point will not be beneficial so we plan to try and
> confirm that date.
>
> I'd like to try and answer the question about locations of RTCWeb interim
> meetings after the ones currently planned.
>
> First of all, the chairs would are going to declare that there is WG
> consensus for a 1:1:1 meeting rotation between west coast of north america,
> europe, and east coast of north america and plan to run approximately equal
> number of meetings in these locations.  In counting the number of past
> meetings in a given region, we will include all the face to face RTCWeb and
> WebRTC meetings as this work is closely joined and a large number of the
> participants travel to both sets of meetings. The face to face meetings
> that happen at the main IETF or W3C meetings are included in this count.
> Collocated meetings will be counted just once as they only require one set
> of travel to that location.
>
> Cullen  <RTCWeb co-chair>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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Just one clarification - I thought the W3C WebRTC group planned to meet on =
June 11th and then the IETF RTCWEB WG on June 12-13? =A0<div><br></div><div=
>The CLUE WG meeting still plans to meet on June 7-8. =A0</div><div><br></d=
iv>
<div>Regards,</div><div>Mary.=A0</div><div><br><div class=3D"gmail_quote">O=
n Mon, Apr 16, 2012 at 2:45 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt;</span> wrote:<br><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex">
<br>
The CLUE and WebRTC groups are making plans based on WebRTC meeting June 12=
 and 13 in Stockholm. The chairs believe that picking a different time or l=
ocations at this point will not be beneficial so we plan to try and confirm=
 that date.<br>

<br>
I&#39;d like to try and answer the question about locations of RTCWeb inter=
im meetings after the ones currently planned.<br>
<br>
First of all, the chairs would are going to declare that there is WG consen=
sus for a 1:1:1 meeting rotation between west coast of north america, europ=
e, and east coast of north america and plan to run approximately equal numb=
er of meetings in these locations. =A0In counting the number of past meetin=
gs in a given region, we will include all the face to face RTCWeb and WebRT=
C meetings as this work is closely joined and a large number of the partici=
pants travel to both sets of meetings. The face to face meetings that happe=
n at the main IETF or W3C meetings are included in this count. Collocated m=
eetings will be counted just once as they only require one set of travel to=
 that location.<br>

<br>
Cullen =A0&lt;RTCWeb co-chair&gt;<br>
<br>
<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>

--20cf3071d0a2c874e204bdd22b58--

From fluffy@cisco.com  Mon Apr 16 14:18:37 2012
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From: Cullen Jennings <fluffy@cisco.com>
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Mary, I think you are correct that the dates being considered are=20

Clue - June 7 and 8=20

WebRTC - June 11

RTCWeb - June 12 and 13

Sorry for confusion.=20

On Apr 16, 2012, at 3:04 PM, Mary Barnes wrote:

> Just one clarification - I thought the W3C WebRTC group planned to =
meet on June 11th and then the IETF RTCWEB WG on June 12-13? =20
>=20
> The CLUE WG meeting still plans to meet on June 7-8. =20
>=20
> Regards,
> Mary.=20
>=20
> On Mon, Apr 16, 2012 at 2:45 PM, Cullen Jennings <fluffy@cisco.com> =
wrote:
>=20
> The CLUE and WebRTC groups are making plans based on WebRTC meeting =
June 12 and 13 in Stockholm. The chairs believe that picking a different =
time or locations at this point will not be beneficial so we plan to try =
and confirm that date.
>=20
> I'd like to try and answer the question about locations of RTCWeb =
interim meetings after the ones currently planned.
>=20
> First of all, the chairs would are going to declare that there is WG =
consensus for a 1:1:1 meeting rotation between west coast of north =
america, europe, and east coast of north america and plan to run =
approximately equal number of meetings in these locations.  In counting =
the number of past meetings in a given region, we will include all the =
face to face RTCWeb and WebRTC meetings as this work is closely joined =
and a large number of the participants travel to both sets of meetings. =
The face to face meetings that happen at the main IETF or W3C meetings =
are included in this count. Collocated meetings will be counted just =
once as they only require one set of travel to that location.
>=20
> Cullen  <RTCWeb co-chair>
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From ron.even.tlv@gmail.com  Mon Apr 16 14:39:46 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Hi,

I would like to suggest that the RTCweb meeting will be on the 11 and 12
followed by the webrtc allowing people to attend CLUE on the 7,8 and RTCweb
without having to stay one more extra day after the weekend.

Thanks

Roni Even

 

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Mary Barnes
Sent: Tuesday, April 17, 2012 12:05 AM
To: Cullen Jennings
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Update of future interim meeting locations

 

Just one clarification - I thought the W3C WebRTC group planned to meet on
June 11th and then the IETF RTCWEB WG on June 12-13?  

 

The CLUE WG meeting still plans to meet on June 7-8.  

 

Regards,

Mary. 

 

On Mon, Apr 16, 2012 at 2:45 PM, Cullen Jennings <fluffy@cisco.com> wrote:


The CLUE and WebRTC groups are making plans based on WebRTC meeting June 12
and 13 in Stockholm. The chairs believe that picking a different time or
locations at this point will not be beneficial so we plan to try and confirm
that date.

I'd like to try and answer the question about locations of RTCWeb interim
meetings after the ones currently planned.

First of all, the chairs would are going to declare that there is WG
consensus for a 1:1:1 meeting rotation between west coast of north america,
europe, and east coast of north america and plan to run approximately equal
number of meetings in these locations.  In counting the number of past
meetings in a given region, we will include all the face to face RTCWeb and
WebRTC meetings as this work is closely joined and a large number of the
participants travel to both sets of meetings. The face to face meetings that
happen at the main IETF or W3C meetings are included in this count.
Collocated meetings will be counted just once as they only require one set
of travel to that location.

Cullen  <RTCWeb co-chair>




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

 


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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'>I would like to suggest that the RTCweb meeting will be on the 11 and =
12 followed by the webrtc allowing people to attend CLUE on the 7,8 and =
RTCweb without having to stay one more extra day after the =
weekend.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Thanks<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 Even<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 =
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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-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Mary Barnes<br><b>Sent:</b> Tuesday, April 17, 2012 12:05 =
AM<br><b>To:</b> Cullen Jennings<br><b>Cc:</b> =
rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] Update of future interim =
meeting locations<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Just one =
clarification - I thought the W3C WebRTC group planned to meet on June =
11th and then the IETF RTCWEB WG on June 12-13? =
&nbsp;<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>The CLUE WG meeting still plans to meet on June 7-8. =
&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Regards,<o:p></o:p></p></div><div><p =
class=3DMsoNormal>Mary.&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>On Mon, =
Apr 16, 2012 at 2:45 PM, Cullen Jennings &lt;<a =
href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal><br>The CLUE and WebRTC groups =
are making plans based on WebRTC meeting June 12 and 13 in Stockholm. =
The chairs believe that picking a different time or locations at this =
point will not be beneficial so we plan to try and confirm that =
date.<br><br>I'd like to try and answer the question about locations of =
RTCWeb interim meetings after the ones currently planned.<br><br>First =
of all, the chairs would are going to declare that there is WG consensus =
for a 1:1:1 meeting rotation between west coast of north america, =
europe, and east coast of north america and plan to run approximately =
equal number of meetings in these locations. &nbsp;In counting the =
number of past meetings in a given region, we will include all the face =
to face RTCWeb and WebRTC meetings as this work is closely joined and a =
large number of the participants travel to both sets of meetings. The =
face to face meetings that happen at the main IETF or W3C meetings are =
included in this count. Collocated meetings will be counted just once as =
they only require one set of travel to that location.<br><br>Cullen =
&nbsp;&lt;RTCWeb =
co-chair&gt;<br><br><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><o:p></=
o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
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From: Stephan Wenger <stewe@stewe.org>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: VP8 payload, decoder processing capabilities  (was Re: [rtcweb] Resolution negotiation - a contribution)
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Date: Mon, 16 Apr 2012 21:40:09 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: [rtcweb] VP8 payload, decoder processing capabilities (was Re: Resolution negotiation - a contribution)
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Hi all,

For context: Harald and myself have been at odds for a while now about the
lack of support for a code point in the VP8 payload that can be used to
negotiate a maximum decoder/bitstream complexity.  Specifically, Harald
(and other VP8 payload folks) suggested that generic mechanisms, such as
the a=3Dframerate attribute of RFC4566 in conjunction with the picture size
aspect of the imageattr of RFC 6236 can be used, at least in the rtcweb
context.  However, as far as I understood our argument, these two
mechanisms in combination are not meant as a limit for decoder complexity
(in terms of samples/sec processing rate), but rather as an indication,
from receiver to sender, of an upper bound of what is "useful to send".
See the email below.  To me, it's quite obvious that an indication of
"useful to send" includes "my decoder can handle this"; however, it could
be more restrictive in that factors other than decoder horsepower could
also be at play, such as screen size, user interface settings, and so on.

I believe that the combination of what can be signaled using the above
mechanisms should be sufficient for rtcweb.  However, I also believe that
it is insufficient for general purpose use, mostly because it requires the
support of RFC 6236, which is not exactly a widely deployed technology.
Further, the a=3Dframerate attribute is not a particularly useful attribute
these days anymore, because variable frame rates, at least for software
encoding/decoding, are the norm.

In previous posts on the payload list (in response to the VP8 payload
WGLC), I have commented on the practical shortcomings of the (lack of)
complexity negotiation, and suggested that this needs to be fixed.

Two options:

1) codify Harald's mechanism (based on a=3Dframerate and imageattr in the
VP8 payload draft, at MUST strength.  "In a declarative context, a
prospective media sender supporting this (VP8 payload) specification MUST
support RFC 4566 a=3Dframerate and RFC6236 imageattr, and MUST include code
points according to both mechanisms to identify the properties of the
media stream.  In an offer-answer context, both offerer and answerer
receiver supporting this VP8 payload specification MUST support
a=3Dframertate and imageattr, and MUST include both in their respective
offer/answer messages, so to identify an operation point that will not
overload the media decoder's capabilities.

The issue with this approach, IMO, is that we are dealing here with three
individual code points (framerate, horizontal and vertical picture size),
where a single code point ought to be sufficient for determining whether a
d=E9cor is capable of decoding a stream, at least from a complexity
viewpoint).

2) include, in the V8 payload, a negotiable SDP code point indicating the
complexity of a stream, in units of samples per second processing
requirements or a derivative thereof (such as: levels as used in the MPEG
world).  For example, the VP8 payload could include a single, optional,
negotiable parameter "SamplePerSecond".  If SamplePerSecond were absent in
the SDP, a value of xxxxx must be inferred.  (a sensible value for xxxxx
could be, for example 9216000, which is the number of samples per second
for VGA resolution at 30 Hz).  If SamplePerSecond is present in a
declarative context, it indicates the minimum processing requirements a
decoder must support in order to successfully decode the stream.  In a
symmetric offer-answer context, SamplePerSecond can be used to "dial down"
the complexity of the stream to a value that both encoder and decoder can
support.

My preference is obviously the second proposal, but I'm willing to help
fleshing out either or both of them, just not today :-)

Regards,
Stephan
=20


On 4.13.2012 00:45 , "Harald Alvestrand" <harald@alvestrand.no> wrote:

>On 04/12/2012 11:13 PM, Stephan Wenger wrote:
>>
>> On 4.12.2012 12:08 , "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>>
>>> On 04/12/2012 08:19 PM, Stephan Wenger wrote:
>>>> Hi Harald,
>>>> Thanks for this strawman.  I believe it should work, but I fail to see
>>>> how
>>>> a two dimensional negotiation requirement (negotiating max values for
>>>> framerate and image size--which, in turn, also has two-dimensional
>>>> properties) leads to better interop than a one dimensional negotiation
>>>> (pixels per second processing requirement).
>>> Stephan,
>>>
>>> I do not see this (or the requirement from the use-cases document)
>>>first
>>> and foremost a decoder complexity negotiation; it is a negotiation of
>>> how much data it is useful to send, given the recipient's intended use
>>> of that data.
>> Then such a negotiation should be executed in addition.  Decoder cycle
>> requirement do matter in practical implementations.
>Feel free to propose language that captures this requirement. As noted,
>my I-D fragment doesn't.
>
>



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From: "Yuepeiyu (Roy)" <yuepeiyu@huawei.com>
To: "payload@ietf.org" <payload@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [payload] VP8 payload, decoder processing capabilities (was Re: [rtcweb] Resolution negotiation - a contribution)
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Subject: Re: [rtcweb] [payload] VP8 payload, decoder processing capabilities (was Re: Resolution negotiation - a contribution)
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From ekr@rtfm.com  Tue Apr 17 13:58:22 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 17 Apr 2012 16:57:40 -0400
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Subject: Re: [rtcweb] Update of future interim meeting locations
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On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com> wrote:
> First of all, the chairs would are going to declare that there is WG
> consensus for a 1:1:1 meeting rotation between west coast of north
> america, europe, and east coast of north america and plan to run
> approximately equal number of meetings in these locations.  In
> counting the number of past meetings in a given region, we will
> include all the face to face RTCWeb and WebRTC meetings as this work
> is closely joined and a large number of the participants travel to
> both sets of meetings. The face to face meetings that happen at the
> main IETF or W3C meetings are included in this count. Collocated
> meetings will be counted just once as they only require one set of
> travel to that location.

Cullen,

Thanks for this clarification. Just to be totally sure I understand,
do you mean the following algorithm?

   For an interim meeting at time X, sum up all the prior meetings
   for each of the regions. The region with the fewest prior
   meetings is then selected for the next interim. [0]

Is this what you guys have in mind?

-Ekr

[0] For extra credit, ties need to be resolved somehow. I can
suggest a mechanism if you need one :)

From marshall.eubanks@gmail.com  Tue Apr 17 14:06:28 2012
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On Tue, Apr 17, 2012 at 4:57 PM, Eric Rescorla <ekr@rtfm.com> wrote:
> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com> wrote=
:
>> First of all, the chairs would are going to declare that there is WG
>> consensus for a 1:1:1 meeting rotation between west coast of north
>> america, europe, and east coast of north america and plan to run
>> approximately equal number of meetings in these locations. =A0In
>> counting the number of past meetings in a given region, we will
>> include all the face to face RTCWeb and WebRTC meetings as this work
>> is closely joined and a large number of the participants travel to
>> both sets of meetings. The face to face meetings that happen at the
>> main IETF or W3C meetings are included in this count. Collocated
>> meetings will be counted just once as they only require one set of
>> travel to that location.
>
> Cullen,
>
> Thanks for this clarification. Just to be totally sure I understand,
> do you mean the following algorithm?
>
> =A0 For an interim meeting at time X, sum up all the prior meetings
> =A0 for each of the regions. The region with the fewest prior
> =A0 meetings is then selected for the next interim. [0]
>

Which implies that  Paris doesn't count.

Regards
Marshall


> Is this what you guys have in mind?
>
> -Ekr
>
> [0] For extra credit, ties need to be resolved somehow. I can
> suggest a mechanism if you need one :)
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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On Tue, Apr 17, 2012 at 5:06 PM, Marshall Eubanks
<marshall.eubanks@gmail.com> wrote:
> On Tue, Apr 17, 2012 at 4:57 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com> wrot=
e:
>>> First of all, the chairs would are going to declare that there is WG
>>> consensus for a 1:1:1 meeting rotation between west coast of north
>>> america, europe, and east coast of north america and plan to run
>>> approximately equal number of meetings in these locations. =A0In
>>> counting the number of past meetings in a given region, we will
>>> include all the face to face RTCWeb and WebRTC meetings as this work
>>> is closely joined and a large number of the participants travel to
>>> both sets of meetings. The face to face meetings that happen at the
>>> main IETF or W3C meetings are included in this count. Collocated
>>> meetings will be counted just once as they only require one set of
>>> travel to that location.
>>
>> Cullen,
>>
>> Thanks for this clarification. Just to be totally sure I understand,
>> do you mean the following algorithm?
>>
>> =A0 For an interim meeting at time X, sum up all the prior meetings
>> =A0 for each of the regions. The region with the fewest prior
>> =A0 meetings is then selected for the next interim. [0]
>>
>
> Which implies that =A0Paris doesn't count.

That's certainly not what I meant. Perhaps it would help to rewrite
"next interim" as "interim X". Concretely, to select an interim meeting
in mid-September, I would expect to count meetings up to and
including Vancouver.

-Ekr

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Cc: Cullen Jennings <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Update of future interim meeting locations
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At 04:09 PM 4/17/2012, Eric Rescorla wrote:
>On Tue, Apr 17, 2012 at 5:06 PM, Marshall Eubanks
><marshall.eubanks@gmail.com> wrote:
> > On Tue, Apr 17, 2012 at 4:57 PM, Eric Rescorla <ekr@rtfm.com> wrote:
> >> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com> wrote:
> >>> First of all, the chairs would are going to declare that there is WG
> >>> consensus for a 1:1:1 meeting rotation between west coast of north
> >>> america, europe, and east coast of north america and plan to run
> >>> approximately equal number of meetings in these locations.  In
> >>> counting the number of past meetings in a given region, we will
> >>> include all the face to face RTCWeb and WebRTC meetings as this work
> >>> is closely joined and a large number of the participants travel to
> >>> both sets of meetings. The face to face meetings that happen at the
> >>> main IETF or W3C meetings are included in this count. Collocated
> >>> meetings will be counted just once as they only require one set of
> >>> travel to that location.
> >>
> >> Cullen,
> >>
> >> Thanks for this clarification. Just to be totally sure I understand,
> >> do you mean the following algorithm?
> >>
> >>   For an interim meeting at time X, sum up all the prior meetings
> >>   for each of the regions. The region with the fewest prior
> >>   meetings is then selected for the next interim. [0]
> >>
> >
> > Which implies that  Paris doesn't count.
>
>That's certainly not what I meant. Perhaps it would help to rewrite
>"next interim" as "interim X". Concretely, to select an interim meeting
>in mid-September, I would expect to count meetings up to and
>including Vancouver.

why don't Paris and Boston (I think that was the last interim, though 
that might have been CLUE) count here?

James


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


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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 17 Apr 2012 17:28:40 -0400
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On Tue, Apr 17, 2012 at 5:15 PM, James M. Polk <jmpolk@cisco.com> wrote:
> At 04:09 PM 4/17/2012, Eric Rescorla wrote:
>>
>> On Tue, Apr 17, 2012 at 5:06 PM, Marshall Eubanks
>> <marshall.eubanks@gmail.com> wrote:
>> > On Tue, Apr 17, 2012 at 4:57 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>> >> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com>
>> >> wrote:
>> >>> First of all, the chairs would are going to declare that there is WG
>> >>> consensus for a 1:1:1 meeting rotation between west coast of north
>> >>> america, europe, and east coast of north america and plan to run
>> >>> approximately equal number of meetings in these locations. =A0In
>> >>> counting the number of past meetings in a given region, we will
>> >>> include all the face to face RTCWeb and WebRTC meetings as this work
>> >>> is closely joined and a large number of the participants travel to
>> >>> both sets of meetings. The face to face meetings that happen at the
>> >>> main IETF or W3C meetings are included in this count. Collocated
>> >>> meetings will be counted just once as they only require one set of
>> >>> travel to that location.
>> >>
>> >> Cullen,
>> >>
>> >> Thanks for this clarification. Just to be totally sure I understand,
>> >> do you mean the following algorithm?
>> >>
>> >> =A0 For an interim meeting at time X, sum up all the prior meetings
>> >> =A0 for each of the regions. The region with the fewest prior
>> >> =A0 meetings is then selected for the next interim. [0]
>> >>
>> >
>> > Which implies that =A0Paris doesn't count.
>>
>> That's certainly not what I meant. Perhaps it would help to rewrite
>> "next interim" as "interim X". Concretely, to select an interim meeting
>> in mid-September, I would expect to count meetings up to and
>> including Vancouver.
>
>
> why don't Paris and Boston (I think that was the last interim, though tha=
t
> might have been CLUE) count here?

They would. That's why I said "Up to and including". Paris was before
September 2012.

Obviously I'm being unclear. Here's a slightly more formal description.

Each scheduled or past meeting is represented as a tuple [region, date].
E.g., Paris would be something like [Europe, 20120325]

To select an interim at date X, take the subset of all meetings with date <=
 X
and count the number in each region. To just make up some data, this
would look like:

{
  'Europe':1,
  'West Coast':2:,
  'East Coast':0
}

Then, select the region with the smallest number of meetings for the
interim at date X.
So, in this case, the interim would be held on the East Coast.

For additional concretenss, iff we were executing this algorithm to select =
this
interim, we would be building a table of all meetings up through Paris. I.e=
.,
the last two would be IETF 83 and the Jan/Feb Interim in Mountain View.
[I'm assuming for the purposes of this discussion that we are not counting
CLUE interims, since the chairs referred to "RTCWeb and WebRTC".

One detail: you need to select the interims in chronological order because =
the
previous interim selections obviously affect the next ones.

Best,
-Ekr

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On Apr 17, 2012, at 2:57 PM, Eric Rescorla wrote:

> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com> =
wrote:
> > First of all, the chairs would are going to declare that there is WG
> > consensus for a 1:1:1 meeting rotation between west coast of north
> > america, europe, and east coast of north america and plan to run
> > approximately equal number of meetings in these locations.  In
> > counting the number of past meetings in a given region, we will
> > include all the face to face RTCWeb and WebRTC meetings as this work
> > is closely joined and a large number of the participants travel to
> > both sets of meetings. The face to face meetings that happen at the
> > main IETF or W3C meetings are included in this count. Collocated
> > meetings will be counted just once as they only require one set of
> > travel to that location.
>=20
> Cullen,
>=20
> Thanks for this clarification. Just to be totally sure I understand,
> do you mean the following algorithm?
>=20
>    For an interim meeting at time X, sum up all the prior meetings
>    for each of the regions. The region with the fewest prior
>    meetings is then selected for the next interim. [0]
>=20
> Is this what you guys have in mind?
>=20
> -Ekr
>=20
> [0] For extra credit, ties need to be resolved somehow. I can
> suggest a mechanism if you need one :)
>=20

Yes - that is what I was thinking and I had not really thought much =
about ties. So unless you want a maximally self serving tie breaking =
algorithm of selecting the location closest to YYC, it would be great if =
someone could suggest the tie breaking algorithm.=20

Cullen




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Cc: Cullen Jennings <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Update of future interim meeting locations
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At 04:28 PM 4/17/2012, Eric Rescorla wrote:
>On Tue, Apr 17, 2012 at 5:15 PM, James M. Polk <jmpolk@cisco.com> wrote:
> > At 04:09 PM 4/17/2012, Eric Rescorla wrote:
> >>
> >> On Tue, Apr 17, 2012 at 5:06 PM, Marshall Eubanks
> >> <marshall.eubanks@gmail.com> wrote:
> >> > On Tue, Apr 17, 2012 at 4:57 PM, Eric Rescorla <ekr@rtfm.com> wrote:
> >> >> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com>
> >> >> wrote:
> >> >>> First of all, the chairs would are going to declare that there is WG
> >> >>> consensus for a 1:1:1 meeting rotation between west coast of north
> >> >>> america, europe, and east coast of north america and plan to run
> >> >>> approximately equal number of meetings in these locations.  In
> >> >>> counting the number of past meetings in a given region, we will
> >> >>> include all the face to face RTCWeb and WebRTC meetings as this work
> >> >>> is closely joined and a large number of the participants travel to
> >> >>> both sets of meetings. The face to face meetings that happen at the
> >> >>> main IETF or W3C meetings are included in this count. Collocated
> >> >>> meetings will be counted just once as they only require one set of
> >> >>> travel to that location.
> >> >>
> >> >> Cullen,
> >> >>
> >> >> Thanks for this clarification. Just to be totally sure I understand,
> >> >> do you mean the following algorithm?
> >> >>
> >> >>   For an interim meeting at time X, sum up all the prior meetings
> >> >>   for each of the regions. The region with the fewest prior
> >> >>   meetings is then selected for the next interim. [0]
> >> >>
> >> >
> >> > Which implies that  Paris doesn't count.
> >>
> >> That's certainly not what I meant. Perhaps it would help to rewrite
> >> "next interim" as "interim X". Concretely, to select an interim meeting
> >> in mid-September, I would expect to count meetings up to and
> >> including Vancouver.
> >
> >
> > why don't Paris and Boston (I think that was the last interim, though that
> > might have been CLUE) count here?
>
>They would. That's why I said "Up to and including". Paris was before
>September 2012.
>
>Obviously I'm being unclear.

No, I think I knew what you wanted to get to the first time, but what 
you have below articulates it quite nicely though.

And, as you pointed out, we need a means for solving for ties - 
mostly because the goal of this is to have a tied once a year between 
3 (if we keep the 1:1:1 format), then once a year between 2, right?

James

>Here's a slightly more formal description.
>
>Each scheduled or past meeting is represented as a tuple [region, date].
>E.g., Paris would be something like [Europe, 20120325]
>
>To select an interim at date X, take the subset of all meetings with date < X
>and count the number in each region. To just make up some data, this
>would look like:
>
>{
>   'Europe':1,
>   'West Coast':2:,
>   'East Coast':0
>}
>
>Then, select the region with the smallest number of meetings for the
>interim at date X.
>So, in this case, the interim would be held on the East Coast.
>
>For additional concretenss, iff we were executing this algorithm to 
>select this
>interim, we would be building a table of all meetings up through Paris. I.e.,
>the last two would be IETF 83 and the Jan/Feb Interim in Mountain View.
>[I'm assuming for the purposes of this discussion that we are not counting
>CLUE interims, since the chairs referred to "RTCWeb and WebRTC".
>
>One detail: you need to select the interims in chronological order because the
>previous interim selections obviously affect the next ones.
>
>Best,
>-Ekr


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At 04:33 PM 4/17/2012, Cullen Jennings wrote:

>On Apr 17, 2012, at 2:57 PM, Eric Rescorla wrote:
>
> > On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com> wrote:
> > > First of all, the chairs would are going to declare that there is WG
> > > consensus for a 1:1:1 meeting rotation between west coast of north
> > > america, europe, and east coast of north america and plan to run
> > > approximately equal number of meetings in these locations.  In
> > > counting the number of past meetings in a given region, we will
> > > include all the face to face RTCWeb and WebRTC meetings as this work
> > > is closely joined and a large number of the participants travel to
> > > both sets of meetings. The face to face meetings that happen at the
> > > main IETF or W3C meetings are included in this count. Collocated
> > > meetings will be counted just once as they only require one set of
> > > travel to that location.
> >
> > Cullen,
> >
> > Thanks for this clarification. Just to be totally sure I understand,
> > do you mean the following algorithm?
> >
> >    For an interim meeting at time X, sum up all the prior meetings
> >    for each of the regions. The region with the fewest prior
> >    meetings is then selected for the next interim. [0]
> >
> > Is this what you guys have in mind?
> >
> > -Ekr
> >
> > [0] For extra credit, ties need to be resolved somehow. I can
> > suggest a mechanism if you need one :)
> >
>
>Yes - that is what I was thinking and I had not really thought much 
>about ties. So unless you want a maximally self serving tie breaking 
>algorithm of selecting the location closest to YYC, it would be 
>great if someone could suggest the tie breaking algorithm.

how about the previous and next IETF meetings disqualify that 
location (and potentially each location). Something like that should 
work in favor of covering the currently least travelled to location 
(in theory).

I.e., last was Europe, next is West Coast NA, that means this interim 
must be on east coast NA (sorry Magnus  ;-)

perhaps this one ought to be that Europe now can't be for a year, 
because it is getting two RTCweb meetings in a row (Paris in March 
and Stockholm in June).

James


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


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From: Eric Rescorla <ekr@rtfm.com>
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On Tue, Apr 17, 2012 at 5:33 PM, Cullen Jennings <fluffy@cisco.com> wrote:
>
> On Apr 17, 2012, at 2:57 PM, Eric Rescorla wrote:

> Yes - that is what I was thinking and I had not really thought much about=
 ties. So unless you want a maximally self serving tie breaking algorithm o=
f selecting the location closest to YYC, it would be great if someone could=
 suggest the tie breaking algorithm.

I would suggest picking the region (in the equivalence set)
we were at least recently in the case of ties.

-Ekr

From Jim.Barnett@genesyslab.com  Tue Apr 17 15:48:03 2012
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Possible tie-breaking algorithm:

  If season=3D=3Dsummer =3D=3D> Europe
  elsif season=3D=3Dwinter =3D=3D> west coast
  elsif season=3D=3Dsorta-in-between =3D=3D> east coast

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Cullen Jennings
Sent: Tuesday, April 17, 2012 5:33 PM
To: Eric Rescorla
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Update of future interim meeting locations


On Apr 17, 2012, at 2:57 PM, Eric Rescorla wrote:

> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings <fluffy@cisco.com>
wrote:
> > First of all, the chairs would are going to declare that there is WG

> > consensus for a 1:1:1 meeting rotation between west coast of north=20
> > america, europe, and east coast of north america and plan to run=20
> > approximately equal number of meetings in these locations.  In=20
> > counting the number of past meetings in a given region, we will=20
> > include all the face to face RTCWeb and WebRTC meetings as this work

> > is closely joined and a large number of the participants travel to=20
> > both sets of meetings. The face to face meetings that happen at the=20
> > main IETF or W3C meetings are included in this count. Collocated=20
> > meetings will be counted just once as they only require one set of=20
> > travel to that location.
>=20
> Cullen,
>=20
> Thanks for this clarification. Just to be totally sure I understand,=20
> do you mean the following algorithm?
>=20
>    For an interim meeting at time X, sum up all the prior meetings
>    for each of the regions. The region with the fewest prior
>    meetings is then selected for the next interim. [0]
>=20
> Is this what you guys have in mind?
>=20
> -Ekr
>=20
> [0] For extra credit, ties need to be resolved somehow. I can suggest=20
> a mechanism if you need one :)
>=20

Yes - that is what I was thinking and I had not really thought much
about ties. So unless you want a maximally self serving tie breaking
algorithm of selecting the location closest to YYC, it would be great if
someone could suggest the tie breaking algorithm.=20

Cullen



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

From harald@alvestrand.no  Wed Apr 18 00:27:38 2012
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Cc: Cullen Jennings <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Update of future interim meeting locations
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1) I'd like to let the chairs have some flexibility
2) It's important when we start counting, too

So far, my current list of meetings for this WG starts off with the 
October 2010 unofficial get-together, and has 1 interim meeting on it: 
Jan/Feb 2012, in Mountain View.

Inbetween are 5 meetings at IETFs including the BOF.

That makes my count:

NA-West: 2 (both interims)
NA-East: 1
Europe: 2
Asia: 2

The next 3 IETF meetings are 1 NA-West and 2 NA-East, so after that set 
plus the upcoming Europe interim, we'll have:

NA-West: 3
NA-East: 3
Europe: 3
Asia: 2

Not too uneven. But just after the upcoming interim, it will seem like 
the maths of the situation should definitely force the chairs to 
schedule an NA-East interim; I'd like the chairs to have leeway about that.

On 04/17/2012 10:57 PM, Eric Rescorla wrote:
> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings<fluffy@cisco.com>  wrote:
>> First of all, the chairs would are going to declare that there is WG
>> consensus for a 1:1:1 meeting rotation between west coast of north
>> america, europe, and east coast of north america and plan to run
>> approximately equal number of meetings in these locations.  In
>> counting the number of past meetings in a given region, we will
>> include all the face to face RTCWeb and WebRTC meetings as this work
>> is closely joined and a large number of the participants travel to
>> both sets of meetings. The face to face meetings that happen at the
>> main IETF or W3C meetings are included in this count. Collocated
>> meetings will be counted just once as they only require one set of
>> travel to that location.
> Cullen,
>
> Thanks for this clarification. Just to be totally sure I understand,
> do you mean the following algorithm?
>
>     For an interim meeting at time X, sum up all the prior meetings
>     for each of the regions. The region with the fewest prior
>     meetings is then selected for the next interim. [0]
>
> Is this what you guys have in mind?
>
> -Ekr
>
> [0] For extra credit, ties need to be resolved somehow. I can
> suggest a mechanism if you need one :)
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From HKaplan@acmepacket.com  Wed Apr 18 01:57:55 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Update of future interim meeting locations
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This is all getting a little silly isn't it? :)
The real issue is we have two WG chairs from NA-West and one from Europe - =
there is no fair representation from NA-East!!  (kidding)

-hadriel

Sent from my iPhone

On Apr 18, 2012, at 3:27 AM, "Harald Alvestrand" <harald@alvestrand.no> wro=
te:

> 1) I'd like to let the chairs have some flexibility
> 2) It's important when we start counting, too
>=20
> So far, my current list of meetings for this WG starts off with the Octob=
er 2010 unofficial get-together, and has 1 interim meeting on it: Jan/Feb 2=
012, in Mountain View.
>=20
> Inbetween are 5 meetings at IETFs including the BOF.
>=20
> That makes my count:
>=20
> NA-West: 2 (both interims)
> NA-East: 1
> Europe: 2
> Asia: 2
>=20
> The next 3 IETF meetings are 1 NA-West and 2 NA-East, so after that set p=
lus the upcoming Europe interim, we'll have:
>=20
> NA-West: 3
> NA-East: 3
> Europe: 3
> Asia: 2
>=20
> Not too uneven. But just after the upcoming interim, it will seem like th=
e maths of the situation should definitely force the chairs to schedule an =
NA-East interim; I'd like the chairs to have leeway about that.
>=20
> On 04/17/2012 10:57 PM, Eric Rescorla wrote:
>> On Mon, Apr 16, 2012 at 3:45 PM, Cullen Jennings<fluffy@cisco.com>  wrot=
e:
>>> First of all, the chairs would are going to declare that there is WG
>>> consensus for a 1:1:1 meeting rotation between west coast of north
>>> america, europe, and east coast of north america and plan to run
>>> approximately equal number of meetings in these locations.  In
>>> counting the number of past meetings in a given region, we will
>>> include all the face to face RTCWeb and WebRTC meetings as this work
>>> is closely joined and a large number of the participants travel to
>>> both sets of meetings. The face to face meetings that happen at the
>>> main IETF or W3C meetings are included in this count. Collocated
>>> meetings will be counted just once as they only require one set of
>>> travel to that location.
>> Cullen,
>>=20
>> Thanks for this clarification. Just to be totally sure I understand,
>> do you mean the following algorithm?
>>=20
>>    For an interim meeting at time X, sum up all the prior meetings
>>    for each of the regions. The region with the fewest prior
>>    meetings is then selected for the next interim. [0]
>>=20
>> Is this what you guys have in mind?
>>=20
>> -Ekr
>>=20
>> [0] For extra credit, ties need to be resolved somehow. I can
>> suggest a mechanism if you need one :)
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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On Wed, Apr 18, 2012 at 12:27 AM, Harald Alvestrand
<harald@alvestrand.no> wrote:
> 1) I'd like to let the chairs have some flexibility

Frankly, I'm a lot less interested in the chairs having some flexibility than
predictability.

As it stands now, we're in the position of being less than 8 weeks out
from the proposed interim and still not knowing whether it's happening
or when or where it will be. Cullens last message says that the chairs will
"try and confirm" Stockholm for June 12-13, but I for one don't
want to plan a couple thousand dollars worth of travel on
"try and confirm."

We know precisely when the IETF and W3C meetings are for the next
year or so, so there's really nothing that stops the chairs from promulgating
a policy and then announcing the locations of the meetings now.

From andrew.hutton@siemens-enterprise.com  Wed Apr 18 08:27:05 2012
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Hi,

Totally agree with Eric what I want is to be able to plan my time and budge=
t well in advance and the main thing I need to know is the date.

If the choice is between Europe, NA-West and NA-East then flight costs and =
time tend not to be very variable.

The main variable is actually hotel costs so just choose somewhere that is =
close to a major international airport and that has reasonable hotel prices=
 and let us know well in advance.

Regards
Andy





> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Eric Rescorla
> Sent: 18 April 2012 16:01
> To: Harald Alvestrand
> Cc: Cullen Jennings; rtcweb@ietf.org
> Subject: Re: [rtcweb] Update of future interim meeting locations
>=20
> On Wed, Apr 18, 2012 at 12:27 AM, Harald Alvestrand
> <harald@alvestrand.no> wrote:
> > 1) I'd like to let the chairs have some flexibility
>=20
> Frankly, I'm a lot less interested in the chairs having some
> flexibility than
> predictability.
>=20
> As it stands now, we're in the position of being less than 8 weeks out
> from the proposed interim and still not knowing whether it's happening
> or when or where it will be. Cullens last message says that the chairs
> will
> "try and confirm" Stockholm for June 12-13, but I for one don't
> want to plan a couple thousand dollars worth of travel on
> "try and confirm."
>=20
> We know precisely when the IETF and W3C meetings are for the next
> year or so, so there's really nothing that stops the chairs from
> promulgating
> a policy and then announcing the locations of the meetings now.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From fluffy@cisco.com  Wed Apr 18 09:10:45 2012
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From: Cullen Jennings <fluffy@cisco.com>
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+1 Eric and Andy.=20

I don't think the chairs need flexibility - Flexible chairs tend to not =
be a pretty unstable.=20


On Apr 18, 2012, at 9:26 AM, Hutton, Andrew wrote:

> Hi,
>=20
> Totally agree with Eric what I want is to be able to plan my time and =
budget well in advance and the main thing I need to know is the date.
>=20
> If the choice is between Europe, NA-West and NA-East then flight costs =
and time tend not to be very variable.
>=20
> The main variable is actually hotel costs so just choose somewhere =
that is close to a major international airport and that has reasonable =
hotel prices and let us know well in advance.
>=20
> Regards
> Andy
>=20
>=20
>=20
>=20
>=20
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Eric Rescorla
>> Sent: 18 April 2012 16:01
>> To: Harald Alvestrand
>> Cc: Cullen Jennings; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Update of future interim meeting locations
>>=20
>> On Wed, Apr 18, 2012 at 12:27 AM, Harald Alvestrand
>> <harald@alvestrand.no> wrote:
>>> 1) I'd like to let the chairs have some flexibility
>>=20
>> Frankly, I'm a lot less interested in the chairs having some
>> flexibility than
>> predictability.
>>=20
>> As it stands now, we're in the position of being less than 8 weeks =
out
>> from the proposed interim and still not knowing whether it's =
happening
>> or when or where it will be. Cullens last message says that the =
chairs
>> will
>> "try and confirm" Stockholm for June 12-13, but I for one don't
>> want to plan a couple thousand dollars worth of travel on
>> "try and confirm."
>>=20
>> We know precisely when the IETF and W3C meetings are for the next
>> year or so, so there's really nothing that stops the chairs from
>> promulgating
>> a policy and then announcing the locations of the meetings now.
>> _______________________________________________
>> 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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On Apr 18, 2012, at 2:57 AM, Hadriel Kaplan wrote:

> This is all getting a little silly isn't it? :)
> The real issue is we have two WG chairs from NA-West and one from =
Europe - there is no fair representation from NA-East!!  (kidding)
>=20
> -hadriel

I nominate Hadriel for chair :-)



From martin.thomson@gmail.com  Wed Apr 18 11:22:07 2012
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On 18 April 2012 09:30, Cullen Jennings <fluffy@cisco.com> wrote:
>> The real issue is we have two WG chairs from NA-West and one from Europe=
 - there is no fair representation from NA-East!! =C2=A0(kidding)
>>
>> -hadriel
>
> I nominate Hadriel for chair :-)

Seconded.  It sounds like one of the NA-West chairs would have to step
down to make this work though ;)

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On Apr 18, 2012, at 12:22 PM, Martin Thomson wrote:

> On 18 April 2012 09:30, Cullen Jennings <fluffy@cisco.com> wrote:
>>> The real issue is we have two WG chairs from NA-West and one from =
Europe - there is no fair representation from NA-East!!  (kidding)
>>>=20
>>> -hadriel
>>=20
>> I nominate Hadriel for chair :-)
>=20
> Seconded.  It sounds like one of the NA-West chairs would have to step
> down to make this work though ;)

Note my phrasing was "chair", not "co-chair"=20


From HKaplan@acmepacket.com  Wed Apr 18 15:06:47 2012
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On Apr 18, 2012, at 12:30 PM, Cullen Jennings wrote:

> On Apr 18, 2012, at 2:57 AM, Hadriel Kaplan wrote:
>=20
>> This is all getting a little silly isn't it? :)
>> The real issue is we have two WG chairs from NA-West and one from Europe=
 - there is no fair representation from NA-East!!  (kidding)
>>=20
>> -hadriel
>=20
> I nominate Hadriel for chair :-)

Oh yes I would make a very unstable chair...

My solution for interims would be to split the difference between all the l=
ocations, by having every interim in Hawaii, and making it last exactly two=
 hours: one hour on Monday morning, one on Friday night.
;)

-hadriel


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Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was	Re:Proposal for H.263 baseline codec]
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Olle,

> 12 apr 2012 kl. 18:09 skrev Kevin P. Fleming:
> 
> > On 04/12/2012 02:21 AM, Olle E. Johansson wrote:
> >
> >> This is really bad news for Open Source. Even if we try to get a way to
> pay for licenses, our business model is far away from understandable for
> most of the syndicates. I tried with the AMR codec once and it stopped at
> initial order quantity. Anything under 10.000 licenses was not up for
> discussion.

Would a patent holder even both with litigation if the patent owner fely
that licensing in quantities of less than 10,000 units is not worth the
money?  Likely not.  Litigation is expensive and patent trolls will only do
it if there is money to be made.  Legitimate companies do not sue unless
there is either profit to be made or (as with the Microsoft/Motorola case)
there is big money at stake related to something bigger than what the legal
case is over.

That said, there is always the risk of getting sued before you can get the
front doors to your small business propped open.  Still, one cannot stop out
of fear of legal action.  Half the companies in Silicon Valley would never
have opened if they concerned themselves too much with patents.

However, if successful, a day will come when it's "time to pay the fiddler
his due."  And when that day comes, there is some comfort in knowing who you
go to in order to license the technology you are using.

> >> G.729 is available on one-by-one basis which both Asterisk and
> FreeSwitch sell to users... I have no insight into how these agreements
> was worked out, but it at least indicates that someone tries to open up.
> >
> > It is my understanding that the licensing arrangements that have been
> made for G.729 are no longer an option for new licensees (not offered by
> the consortium), and to my knowledge such arrangements have never been an
> option for any of the other voice/video codecs that our customers have
> asked us about (including H.264, the AMR family and others).
> 
> And isn't that a shame. The question here for the legal department is
> wether Google's policies in regards to VP8 is enough for companies like
> you to work with it in either binary addons or Open Source code? Does it
> provide enough liability assurance?

Google's policies will not prevent you from getting sued one day when
successful.  What would make me feel better is if Google could indemnify
people.  If they feel they own all of the technology in VP8, they could
indemnify people.  I would be pleasantly surprised if their legal team truly
feels so confident that they own all of the technology in VP8, though.

With H.264, you know where to go license technology once you start to see a
little success.  The IPR holders are known.  I cannot imagine a troll out
there who wouldn't have already filed a lawsuit by now.  (And I'll note
again that VP8 might even have elements that are covered by IPR holders who
also own IPR on H.264 and who may not be so willing to license it for use
with VP8.  I see a definite business risk there that I don't see with AMR or
H.264.)

> I think that question is where we are right now.
>
> If not, will Google try to offer a solution?

I would not suggest you hold your breath :-)  I applaud Google for taking
the steps it has taken so far, but I do not think there is more they can
offer than what they have already.

Paul



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I hate auto-correct.

"even both" --> "even bother"
"owner fely that" --> "owner felt that"
...

I hope the text can be parsed by most ;-)

Paul

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Paul E. Jones
> Sent: Wednesday, April 18, 2012 11:49 PM
> To: 'Olle E. Johansson'; 'Kevin P. Fleming'
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Google VP8 Patent Grant for third parties [Was
> Re:Proposal for H.263 baseline codec]
> 
> Olle,
> 
> > 12 apr 2012 kl. 18:09 skrev Kevin P. Fleming:
> >
> > > On 04/12/2012 02:21 AM, Olle E. Johansson wrote:
> > >
> > >> This is really bad news for Open Source. Even if we try to get a
> > >> way to
> > pay for licenses, our business model is far away from understandable
> > for most of the syndicates. I tried with the AMR codec once and it
> > stopped at initial order quantity. Anything under 10.000 licenses was
> > not up for discussion.
> 
> Would a patent holder even both with litigation if the patent owner fely
> that licensing in quantities of less than 10,000 units is not worth the
> money?  Likely not.  Litigation is expensive and patent trolls will only
> do it if there is money to be made.  Legitimate companies do not sue
> unless there is either profit to be made or (as with the
> Microsoft/Motorola case) there is big money at stake related to something
> bigger than what the legal case is over.
> 
> That said, there is always the risk of getting sued before you can get the
> front doors to your small business propped open.  Still, one cannot stop
> out of fear of legal action.  Half the companies in Silicon Valley would
> never have opened if they concerned themselves too much with patents.
> 
> However, if successful, a day will come when it's "time to pay the fiddler
> his due."  And when that day comes, there is some comfort in knowing who
> you go to in order to license the technology you are using.
> 
> > >> G.729 is available on one-by-one basis which both Asterisk and
> > FreeSwitch sell to users... I have no insight into how these
> > agreements was worked out, but it at least indicates that someone tries
> to open up.
> > >
> > > It is my understanding that the licensing arrangements that have
> > > been
> > made for G.729 are no longer an option for new licensees (not offered
> > by the consortium), and to my knowledge such arrangements have never
> > been an option for any of the other voice/video codecs that our
> > customers have asked us about (including H.264, the AMR family and
> others).
> >
> > And isn't that a shame. The question here for the legal department is
> > wether Google's policies in regards to VP8 is enough for companies
> > like you to work with it in either binary addons or Open Source code?
> > Does it provide enough liability assurance?
> 
> Google's policies will not prevent you from getting sued one day when
> successful.  What would make me feel better is if Google could indemnify
> people.  If they feel they own all of the technology in VP8, they could
> indemnify people.  I would be pleasantly surprised if their legal team
> truly feels so confident that they own all of the technology in VP8,
> though.
> 
> With H.264, you know where to go license technology once you start to see
> a little success.  The IPR holders are known.  I cannot imagine a troll
> out there who wouldn't have already filed a lawsuit by now.  (And I'll
> note again that VP8 might even have elements that are covered by IPR
> holders who also own IPR on H.264 and who may not be so willing to license
> it for use with VP8.  I see a definite business risk there that I don't
> see with AMR or
> H.264.)
> 
> > I think that question is where we are right now.
> >
> > If not, will Google try to offer a solution?
> 
> I would not suggest you hold your breath :-)  I applaud Google for taking
> the steps it has taken so far, but I do not think there is more they can
> offer than what they have already.
> 
> Paul
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From randell-ietf@jesup.org  Thu Apr 19 00:15:59 2012
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On 3/29/2012 1:12 PM, Silvia Pfeiffer wrote:
> Very interesting discussion. Out of personal curiosity, I have some
> questions inline.
>
> On Thu, Mar 29, 2012 at 7:20 AM, Stephan Wenger<stewe@stewe.org>  wrote:
>>
>> The most commonly cited timeline for a widely in use technology to be
>> "save" from a patent viewpoint, based on equitable defenses such as laches
>> (in the US) is six years.
>
> Very interesting to know the usually used limit on the number of
> years. IIUC that would make Speex, Vorbis and Theora "save" codecs,
> since they were released in 2003, 2000, and 2004 respectively?

IANAL:  No, since the defense (with some notable exceptions) is by a 
specific person against delayed assertion of a suit against them, so 
someone using Speex for example could not use that defense for circa 6 
years after they began using it.  Note that publication of a spec 
doesn't violate a patent (though using it may).  In practice the defense 
might be even more limited, for example if the infringer made very 
limited use of the patent for years such that the owner didn't even know 
they were infringing; it's possible the court might start the clock when 
the owner knew or could/should have known of the infringement.


-- 
Randell Jesup
randell-ietf@jesup.org

From harald@alvestrand.no  Thu Apr 19 00:46:43 2012
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Date: Thu, 19 Apr 2012 09:46:40 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory resolutions
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When discussing the issue of resolution negotiation locally, we found a 
quandary:

We have two different negotiation needs:
- Maximum resolution that's possible to use in a given context
- The preferred resolution for a given context

The current imageattr= spec (RFC 6236) doesn't specify clearly whether 
the result of the negotiation is to be considered mandatory or advisory 
(it uses the term "wishes" a lot), but the description reads most 
consistently if one assumes that it is a "MUST" type request.

One interpretation is that the highest functionality available in the 
spec is the "limit", while the highest q value is the "preference", thus:

recv [x=320,y=200,q=1.0] [x=200:1024,y=160:768,q=0.1]

would indicate that the receiver is capable of 200x160 to 1024x768, but 
wants 320x200.

This maps reasonably to the capabilities spec's talk about "mandatory" 
vs "optional" capabilities.

Does this interpretation make sense?

                       Harald


On 04/12/2012 10:46 AM, Harald Alvestrand wrote:
> Hello folks,
>
> after the IETF, I've attempted to gather together information about 
> resolution negotiation and what we might need to do there.
>
> I pulled together the paragraphs below - this may want to be 
> incorporated in an I-D on "SDP usage in RTCWEB", or it may want to go 
> somewhere else. Chairs' guidance is appreciated.
>
> At the moment, I have it in xml2rfc format, so it should be easy to do 
> cut-and-paste on it, but don't want to push it as a separate I-D 
> unless that's what the chairs desire.
> Note - there are some QUERY sections in there - I'm unsure what those 
> should be resolved to.
>
> Comments?
>
>                          Harald
>
> 2.  Requirements
>
>    The relevant requirement for video resolution negotiation from the
>    RTCWEB use cases document
>    [I-D.ietf-rtcweb-use-cases-and-requirements] is:
>
>    o  F25 The browser SHOULD use encoding of streams suitable for the
>       current rendering (e.g. video display size) and SHOULD change
>       parameters if the rendering changes during the session.
>
>    The resulting need is:
>
>    o  To negotiate a maximum spatial resolution for all videos at call
>       setup time
>
>    o  To negotiate a maximum temporal resolution ("frame rate") across
>       all videos at call setup time
>
>    o  To indicate the desire of the recipient for a particular spatial
>       or temporal resolution on a particular video source
>
>    o  To indicate the intent of the sender to send a video source in a
>       particular spatial or temporal resolution
>
>    This document does not mention other requirements.


From ron.even.tlv@gmail.com  Thu Apr 19 01:20:17 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory	resolutions
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Hi Harald,
The reason for the language was that this is really what the receiver wished
to receive. The issue is that at least for H.264 there is no mandatory
requirement for the encoder to support all resolutions within the supported
level. The receiver must be able to crop and scale.
 
Roni Even

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: Thursday, April 19, 2012 10:47 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
> resolutions
> 
> When discussing the issue of resolution negotiation locally, we found a
> quandary:
> 
> We have two different negotiation needs:
> - Maximum resolution that's possible to use in a given context
> - The preferred resolution for a given context
> 
> The current imageattr= spec (RFC 6236) doesn't specify clearly whether
> the result of the negotiation is to be considered mandatory or advisory
> (it uses the term "wishes" a lot), but the description reads most
> consistently if one assumes that it is a "MUST" type request.
> 
> One interpretation is that the highest functionality available in the
> spec is the "limit", while the highest q value is the "preference",
> thus:
> 
> recv [x=320,y=200,q=1.0] [x=200:1024,y=160:768,q=0.1]
> 
> would indicate that the receiver is capable of 200x160 to 1024x768, but
> wants 320x200.
> 
> This maps reasonably to the capabilities spec's talk about "mandatory"
> vs "optional" capabilities.
> 
> Does this interpretation make sense?
> 
>                        Harald
> 
> 
> On 04/12/2012 10:46 AM, Harald Alvestrand wrote:
> > Hello folks,
> >
> > after the IETF, I've attempted to gather together information about
> > resolution negotiation and what we might need to do there.
> >
> > I pulled together the paragraphs below - this may want to be
> > incorporated in an I-D on "SDP usage in RTCWEB", or it may want to go
> > somewhere else. Chairs' guidance is appreciated.
> >
> > At the moment, I have it in xml2rfc format, so it should be easy to
> do
> > cut-and-paste on it, but don't want to push it as a separate I-D
> > unless that's what the chairs desire.
> > Note - there are some QUERY sections in there - I'm unsure what those
> > should be resolved to.
> >
> > Comments?
> >
> >                          Harald
> >
> > 2.  Requirements
> >
> >    The relevant requirement for video resolution negotiation from the
> >    RTCWEB use cases document
> >    [I-D.ietf-rtcweb-use-cases-and-requirements] is:
> >
> >    o  F25 The browser SHOULD use encoding of streams suitable for the
> >       current rendering (e.g. video display size) and SHOULD change
> >       parameters if the rendering changes during the session.
> >
> >    The resulting need is:
> >
> >    o  To negotiate a maximum spatial resolution for all videos at
> call
> >       setup time
> >
> >    o  To negotiate a maximum temporal resolution ("frame rate")
> across
> >       all videos at call setup time
> >
> >    o  To indicate the desire of the recipient for a particular
> spatial
> >       or temporal resolution on a particular video source
> >
> >    o  To indicate the intent of the sender to send a video source in
> a
> >       particular spatial or temporal resolution
> >
> >    This document does not mention other requirements.
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Thu Apr 19 01:47:51 2012
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Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory	resolutions
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On 04/19/2012 10:18 AM, Roni Even wrote:
> Hi Harald,
> The reason for the language was that this is really what the receiver wished
> to receive.
The strength of his wish may be the issue here....
>   The issue is that at least for H.264 there is no mandatory
> requirement for the encoder to support all resolutions within the supported
> level. The receiver must be able to crop and scale.
I couldn't parse that.

Does the sender have to (MUST) send a resolution within the boundaries 
of the "recv" spec, or can he send something outside it? (ie if the 
receiver sends recv [x=320,y=200], can the sender send something with 
x=320,y=240, or is he completely constrained?)

What does the receiver have to be able to do?

                    Harald
>
> Roni Even
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Harald Alvestrand
>> Sent: Thursday, April 19, 2012 10:47 AM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
>> resolutions
>>
>> When discussing the issue of resolution negotiation locally, we found a
>> quandary:
>>
>> We have two different negotiation needs:
>> - Maximum resolution that's possible to use in a given context
>> - The preferred resolution for a given context
>>
>> The current imageattr= spec (RFC 6236) doesn't specify clearly whether
>> the result of the negotiation is to be considered mandatory or advisory
>> (it uses the term "wishes" a lot), but the description reads most
>> consistently if one assumes that it is a "MUST" type request.
>>
>> One interpretation is that the highest functionality available in the
>> spec is the "limit", while the highest q value is the "preference",
>> thus:
>>
>> recv [x=320,y=200,q=1.0] [x=200:1024,y=160:768,q=0.1]
>>
>> would indicate that the receiver is capable of 200x160 to 1024x768, but
>> wants 320x200.
>>
>> This maps reasonably to the capabilities spec's talk about "mandatory"
>> vs "optional" capabilities.
>>
>> Does this interpretation make sense?
>>
>>                         Harald
>>
>>
>> On 04/12/2012 10:46 AM, Harald Alvestrand wrote:
>>> Hello folks,
>>>
>>> after the IETF, I've attempted to gather together information about
>>> resolution negotiation and what we might need to do there.
>>>
>>> I pulled together the paragraphs below - this may want to be
>>> incorporated in an I-D on "SDP usage in RTCWEB", or it may want to go
>>> somewhere else. Chairs' guidance is appreciated.
>>>
>>> At the moment, I have it in xml2rfc format, so it should be easy to
>> do
>>> cut-and-paste on it, but don't want to push it as a separate I-D
>>> unless that's what the chairs desire.
>>> Note - there are some QUERY sections in there - I'm unsure what those
>>> should be resolved to.
>>>
>>> Comments?
>>>
>>>                           Harald
>>>
>>> 2.  Requirements
>>>
>>>     The relevant requirement for video resolution negotiation from the
>>>     RTCWEB use cases document
>>>     [I-D.ietf-rtcweb-use-cases-and-requirements] is:
>>>
>>>     o  F25 The browser SHOULD use encoding of streams suitable for the
>>>        current rendering (e.g. video display size) and SHOULD change
>>>        parameters if the rendering changes during the session.
>>>
>>>     The resulting need is:
>>>
>>>     o  To negotiate a maximum spatial resolution for all videos at
>> call
>>>        setup time
>>>
>>>     o  To negotiate a maximum temporal resolution ("frame rate")
>> across
>>>        all videos at call setup time
>>>
>>>     o  To indicate the desire of the recipient for a particular
>> spatial
>>>        or temporal resolution on a particular video source
>>>
>>>     o  To indicate the intent of the sender to send a video source in
>> a
>>>        particular spatial or temporal resolution
>>>
>>>     This document does not mention other requirements.
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>


From magnus.westerlund@ericsson.com  Thu Apr 19 02:18:55 2012
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Cc: Cullen Jennings <fluffy@cisco.com>, "rai-ads@tools.ietf.org" <rai-ads@tools.ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "clue-chairs@tools.ietf.org" <clue-chairs@tools.ietf.org>
Subject: Re: [rtcweb] URGENT: Dates for upcoming interim ON HOLD
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RTCWEB WG,

The dates for the upcoming interim is off the hold. I can confirm AD
approval, and that we will hold an Face to Face interim meeting on the
12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a
meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for
each of those days.

As a Host I can't confirm the location within Stockholm yet. We are
aiming on either a location in Kista or in the central part of town.
There are some hotels out in Kista, but even if we end up in Kista
staying in central Stockholm allows for a short (17 min) subway ride
from T-centralen plus a no more than 10 min walk to the Kista venue.
Making it a very viable option to book a hotel room in the area around
T-centralen or on Kungsholmen with easy access to the blue-line subway.

Cheers

Magnus Westerlund




On 2012-04-12 19:42, Ted Hardie wrote:
> Please hold off making travel plans based on this timing.   Because
> neither RAI area director will be available during this timing, they
> have asked to consider shifting this timing to the previous week, June
> 7th and 8th.  Until the chairs have had a chance to discuss this,
> please hold off on booking travel plans.
> 
> My apologies for this scramble,
> 
> Ted Hardie
> 
> On Thu, Apr 12, 2012 at 8:20 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
>> After discussion between the WEBRTC and RTCWEB chairs, consulting the
>> doodle poll, and working through the potential for co-location with
>> CLUE, we have decided to set the date for the RTCWEB interim at June
>> 12 & 13th, to be hosted by Ericsson in Stockholm.  We understand that
>> the WEBRTC chairs will hold their meeting on June 11th.  If CLUE
>> wishes to co-locate their meeting, the 14th and 15th will be
>> available.  This would not have been possible on the previous week,
>> given the national holiday in Sweden on the Wednesday of that week.
>>
>> We very much appreciated the work Eric put into his analysis, and in
>> general we will try to prefer hubs; in this particular case, however,
>> a large fraction of the Western European participants are either based
>> in Stockholm or must fly through it to reach one of the larger hubs.
>> Given the availability of a host and this data, we decided to select
>> Stockholm for this meeting.
>>
>> regards,
>>
>> Ted Hardie, for the chairs.
> 


-- 

Magnus Westerlund

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----------------------------------------------------------------------
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RTCWEB WG,

The dates for the upcoming interim is off the hold. I can confirm AD
approval, and that we will hold an Face to Face interim meeting on the
12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a
meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for
each of those days.

As a Host I can't confirm the location within Stockholm yet. We are
aiming on either a location in Kista or in the central part of town.
There are some hotels out in Kista, but even if we end up in Kista
staying in central Stockholm allows for a short (17 min) subway ride
from T-centralen plus a no more than 10 min walk to the Kista venue.
Making it a very viable option to book a hotel room in the area around
T-centralen or on Kungsholmen with easy access to the blue-line subway.

Cheers

Magnus Westerlund
-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From ron.even.tlv@gmail.com  Thu Apr 19 02:58:03 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Harald Alvestrand'" <harald@alvestrand.no>
References: <4F869648.2020605@alvestrand.no> <4F8FC2E0.60701@alvestrand.no> <4f8fcab8.c268b40a.7fe1.ffffb433@mx.google.com> <4F8FD12C.9070703@alvestrand.no>
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Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory	resolutions
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Hi Harald,
Look at example 4.2.1. The sender of the media will reject the recv value in
the offer and may offer other send options. Typically they should be larger
than the rejected value in order to allow cropping and/or scaling down to
the right size. (Cropping may provide better quality than scaling up).

Roni

> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Thursday, April 19, 2012 11:48 AM
> To: Roni Even
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
> resolutions
> 
> On 04/19/2012 10:18 AM, Roni Even wrote:
> > Hi Harald,
> > The reason for the language was that this is really what the receiver
> > wished to receive.
> The strength of his wish may be the issue here....
> >   The issue is that at least for H.264 there is no mandatory
> > requirement for the encoder to support all resolutions within the
> > supported level. The receiver must be able to crop and scale.
> I couldn't parse that.
> 
> Does the sender have to (MUST) send a resolution within the boundaries
> of the "recv" spec, or can he send something outside it? (ie if the
> receiver sends recv [x=320,y=200], can the sender send something with
> x=320,y=240, or is he completely constrained?)
> 
> What does the receiver have to be able to do?
> 
>                     Harald
> >
> > Roni Even
> >
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >> Behalf Of Harald Alvestrand
> >> Sent: Thursday, April 19, 2012 10:47 AM
> >> To: rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
> >> resolutions
> >>
> >> When discussing the issue of resolution negotiation locally, we
> found
> >> a
> >> quandary:
> >>
> >> We have two different negotiation needs:
> >> - Maximum resolution that's possible to use in a given context
> >> - The preferred resolution for a given context
> >>
> >> The current imageattr= spec (RFC 6236) doesn't specify clearly
> >> whether the result of the negotiation is to be considered mandatory
> >> or advisory (it uses the term "wishes" a lot), but the description
> >> reads most consistently if one assumes that it is a "MUST" type
> request.
> >>
> >> One interpretation is that the highest functionality available in
> the
> >> spec is the "limit", while the highest q value is the "preference",
> >> thus:
> >>
> >> recv [x=320,y=200,q=1.0] [x=200:1024,y=160:768,q=0.1]
> >>
> >> would indicate that the receiver is capable of 200x160 to 1024x768,
> >> but wants 320x200.
> >>
> >> This maps reasonably to the capabilities spec's talk about
> "mandatory"
> >> vs "optional" capabilities.
> >>
> >> Does this interpretation make sense?
> >>
> >>                         Harald
> >>
> >>
> >> On 04/12/2012 10:46 AM, Harald Alvestrand wrote:
> >>> Hello folks,
> >>>
> >>> after the IETF, I've attempted to gather together information about
> >>> resolution negotiation and what we might need to do there.
> >>>
> >>> I pulled together the paragraphs below - this may want to be
> >>> incorporated in an I-D on "SDP usage in RTCWEB", or it may want to
> >>> go somewhere else. Chairs' guidance is appreciated.
> >>>
> >>> At the moment, I have it in xml2rfc format, so it should be easy to
> >> do
> >>> cut-and-paste on it, but don't want to push it as a separate I-D
> >>> unless that's what the chairs desire.
> >>> Note - there are some QUERY sections in there - I'm unsure what
> >>> those should be resolved to.
> >>>
> >>> Comments?
> >>>
> >>>                           Harald
> >>>
> >>> 2.  Requirements
> >>>
> >>>     The relevant requirement for video resolution negotiation from
> the
> >>>     RTCWEB use cases document
> >>>     [I-D.ietf-rtcweb-use-cases-and-requirements] is:
> >>>
> >>>     o  F25 The browser SHOULD use encoding of streams suitable for
> the
> >>>        current rendering (e.g. video display size) and SHOULD
> change
> >>>        parameters if the rendering changes during the session.
> >>>
> >>>     The resulting need is:
> >>>
> >>>     o  To negotiate a maximum spatial resolution for all videos at
> >> call
> >>>        setup time
> >>>
> >>>     o  To negotiate a maximum temporal resolution ("frame rate")
> >> across
> >>>        all videos at call setup time
> >>>
> >>>     o  To indicate the desire of the recipient for a particular
> >> spatial
> >>>        or temporal resolution on a particular video source
> >>>
> >>>     o  To indicate the intent of the sender to send a video source
> >>> in
> >> a
> >>>        particular spatial or temporal resolution
> >>>
> >>>     This document does not mention other requirements.
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >


From harald@alvestrand.no  Thu Apr 19 04:17:59 2012
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Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory	resolutions
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On 04/19/2012 11:56 AM, Roni Even wrote:
> Hi Harald,
> Look at example 4.2.1. The sender of the media will reject the recv value in
> the offer and may offer other send options. Typically they should be larger
> than the rejected value in order to allow cropping and/or scaling down to
> the right size. (Cropping may provide better quality than scaling up).

Thank you - that helps a bit.

I interpret the example in 4.2.1 (second alternative) as consisting of 4 
messages; in an offer/answer exchange, this would look like:

1) Alice -> Bob: OFFER
     a=imageattr:97 send [x=800,y=640,sar=1.1,q=0.6] [x=480,y=320] \
                    recv [x=330,y=250]
2) Bob -> Alice: ANSWER
     a=imageattr:97 recv [x=800,y=640,sar=1.1] \
                    send [x=[320:16:640],y=[240:16:480],par=[1.2-1.3]]
3) Alice -> Bob: OFFER (re-invite, probably)
     a=imageattr:97 send [x=800,y=640,sar=1.1] \
                    recv [x=336,y=256]
4) Bob -> Alice: ANSWER
     a=imageattr:97 recv [x=800,y=640,sar=1.1] \
                    send [x=336,y=256]

So after message 2, Bob cannot send anything, since he can't reproduce 
what Alice asked for, right?

And after message 2, Alice can only send 800x640 with a sar of 1.1, 
right? It's forbidden for Alice to send a noncompensated image, or a 
640x480 image?

Again, it's the use of the words "wishes" and "desires" that I want to 
be sure I understand correctly - as used in RFC 6236, they both mean 
"demand" / "require" / "MUST". Right?

> Roni
>
>> -----Original Message-----
>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>> Sent: Thursday, April 19, 2012 11:48 AM
>> To: Roni Even
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
>> resolutions
>>
>> On 04/19/2012 10:18 AM, Roni Even wrote:
>>> Hi Harald,
>>> The reason for the language was that this is really what the receiver
>>> wished to receive.
>> The strength of his wish may be the issue here....
>>>    The issue is that at least for H.264 there is no mandatory
>>> requirement for the encoder to support all resolutions within the
>>> supported level. The receiver must be able to crop and scale.
>> I couldn't parse that.
>>
>> Does the sender have to (MUST) send a resolution within the boundaries
>> of the "recv" spec, or can he send something outside it? (ie if the
>> receiver sends recv [x=320,y=200], can the sender send something with
>> x=320,y=240, or is he completely constrained?)
>>
>> What does the receiver have to be able to do?
>>
>>                      Harald
>>> Roni Even
>>>
>>>> -----Original Message-----
>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>> Behalf Of Harald Alvestrand
>>>> Sent: Thursday, April 19, 2012 10:47 AM
>>>> To: rtcweb@ietf.org
>>>> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
>>>> resolutions
>>>>
>>>> When discussing the issue of resolution negotiation locally, we
>> found
>>>> a
>>>> quandary:
>>>>
>>>> We have two different negotiation needs:
>>>> - Maximum resolution that's possible to use in a given context
>>>> - The preferred resolution for a given context
>>>>
>>>> The current imageattr= spec (RFC 6236) doesn't specify clearly
>>>> whether the result of the negotiation is to be considered mandatory
>>>> or advisory (it uses the term "wishes" a lot), but the description
>>>> reads most consistently if one assumes that it is a "MUST" type
>> request.
>>>> One interpretation is that the highest functionality available in
>> the
>>>> spec is the "limit", while the highest q value is the "preference",
>>>> thus:
>>>>
>>>> recv [x=320,y=200,q=1.0] [x=200:1024,y=160:768,q=0.1]
>>>>
>>>> would indicate that the receiver is capable of 200x160 to 1024x768,
>>>> but wants 320x200.
>>>>
>>>> This maps reasonably to the capabilities spec's talk about
>> "mandatory"
>>>> vs "optional" capabilities.
>>>>
>>>> Does this interpretation make sense?
>>>>
>>>>                          Harald
>>>>
>>>>
>>>> On 04/12/2012 10:46 AM, Harald Alvestrand wrote:
>>>>> Hello folks,
>>>>>
>>>>> after the IETF, I've attempted to gather together information about
>>>>> resolution negotiation and what we might need to do there.
>>>>>
>>>>> I pulled together the paragraphs below - this may want to be
>>>>> incorporated in an I-D on "SDP usage in RTCWEB", or it may want to
>>>>> go somewhere else. Chairs' guidance is appreciated.
>>>>>
>>>>> At the moment, I have it in xml2rfc format, so it should be easy to
>>>> do
>>>>> cut-and-paste on it, but don't want to push it as a separate I-D
>>>>> unless that's what the chairs desire.
>>>>> Note - there are some QUERY sections in there - I'm unsure what
>>>>> those should be resolved to.
>>>>>
>>>>> Comments?
>>>>>
>>>>>                            Harald
>>>>>
>>>>> 2.  Requirements
>>>>>
>>>>>      The relevant requirement for video resolution negotiation from
>> the
>>>>>      RTCWEB use cases document
>>>>>      [I-D.ietf-rtcweb-use-cases-and-requirements] is:
>>>>>
>>>>>      o  F25 The browser SHOULD use encoding of streams suitable for
>> the
>>>>>         current rendering (e.g. video display size) and SHOULD
>> change
>>>>>         parameters if the rendering changes during the session.
>>>>>
>>>>>      The resulting need is:
>>>>>
>>>>>      o  To negotiate a maximum spatial resolution for all videos at
>>>> call
>>>>>         setup time
>>>>>
>>>>>      o  To negotiate a maximum temporal resolution ("frame rate")
>>>> across
>>>>>         all videos at call setup time
>>>>>
>>>>>      o  To indicate the desire of the recipient for a particular
>>>> spatial
>>>>>         or temporal resolution on a particular video source
>>>>>
>>>>>      o  To indicate the intent of the sender to send a video source
>>>>> in
>>>> a
>>>>>         particular spatial or temporal resolution
>>>>>
>>>>>      This document does not mention other requirements.
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>


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On Thu, Apr 19, 2012 at 3:15 AM, Randell Jesup <randell-ietf@jesup.org> wro=
te:
> On 3/29/2012 1:12 PM, Silvia Pfeiffer wrote:
>>
>> Very interesting discussion. Out of personal curiosity, I have some
>> questions inline.
>>
>> On Thu, Mar 29, 2012 at 7:20 AM, Stephan Wenger<stewe@stewe.org> =A0wrot=
e:
>>>
>>>
>>> The most commonly cited timeline for a widely in use technology to be
>>> "save" from a patent viewpoint, based on equitable defenses such as
>>> laches
>>> (in the US) is six years.
>>
>>
>> Very interesting to know the usually used limit on the number of
>> years. IIUC that would make Speex, Vorbis and Theora "save" codecs,
>> since they were released in 2003, 2000, and 2004 respectively?
>
>
> IANAL: =A0No, since the defense (with some notable exceptions) is by a
> specific person against delayed assertion of a suit against them, so some=
one
> using Speex for example could not use that defense for circa 6 years afte=
r
> they began using it. =A0Note that publication of a spec doesn't violate a
> patent (though using it may). =A0In practice the defense might be even mo=
re
> limited, for example if the infringer made very limited use of the patent
> for years such that the owner didn't even know they were infringing; it's
> possible the court might start the clock when the owner knew or could/sho=
uld
> have known of the infringement.
>

I found a fairly detailed analysis of the defense of laches in patent
cases starting at page 27 of this

http://www.fdml.com/defenses.pdf

Reading this dense legal reasoning makes me very nervous about relying
on any of this for any IETF decision. Engineers should not
pretend they are lawyers.

Regards
Marshall

>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Harald Alvestrand'" <harald@alvestrand.no>
References: <4F869648.2020605@alvestrand.no> <4F8FC2E0.60701@alvestrand.no> <4f8fcab8.c268b40a.7fe1.ffffb433@mx.google.com> <4F8FD12C.9070703@alvestrand.no> <4f8fe1a7.2a0db50a.23db.136b@mx.google.com> <4F8FF45C.7090003@alvestrand.no>
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Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory	resolutions
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Hi Harald,
This is a good question, my view will be that since Alice gave a preference
like profile and level (and maybe bandwidth) in the original offer than Bob
can send any video that will be within this level. The image attribute says
that Alice prefers to get something specific to help with processing (note
that Alice can limit the received video by using max-mbps and max-fs for
H.264). It may be good to discuss it with others from MMUSIC.

The usage of the image attribute in my view is more relevant to allow the
receiver to ask for a specific image size since this is not defined in H.264
(in H.263 you can specify the common specific resolutions and frame rate
using the optional parameters in RFC 4629). This was a requirement to allow
a receiver to ask for a specific size.
The usage of "wish" was because H.264 encoder is not required to support
scale to a specific resolution and a compliant decoder must support
everything within the negotiated profile and level. So when a receiver asks
for a size it can happen only if the encoder can do it but if the encoder
cannot it is still compliant.

I think that one reason for no scaling requirements on encoder is to support
simple video cameras that support one resolution and send an H.264 stream
(no scaling/cropping capabilities) and the decoder must be able to display
it on any output device (monitor).

Roni


> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Thursday, April 19, 2012 2:18 PM
> To: Roni Even
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
> resolutions
> 
> On 04/19/2012 11:56 AM, Roni Even wrote:
> > Hi Harald,
> > Look at example 4.2.1. The sender of the media will reject the recv
> > value in the offer and may offer other send options. Typically they
> > should be larger than the rejected value in order to allow cropping
> > and/or scaling down to the right size. (Cropping may provide better
> quality than scaling up).
> 
> Thank you - that helps a bit.
> 
> I interpret the example in 4.2.1 (second alternative) as consisting of
> 4 messages; in an offer/answer exchange, this would look like:
> 
> 1) Alice -> Bob: OFFER
>      a=imageattr:97 send [x=800,y=640,sar=1.1,q=0.6] [x=480,y=320] \
>                     recv [x=330,y=250]
> 2) Bob -> Alice: ANSWER
>      a=imageattr:97 recv [x=800,y=640,sar=1.1] \
>                     send [x=[320:16:640],y=[240:16:480],par=[1.2-1.3]]
> 3) Alice -> Bob: OFFER (re-invite, probably)
>      a=imageattr:97 send [x=800,y=640,sar=1.1] \
>                     recv [x=336,y=256]
> 4) Bob -> Alice: ANSWER
>      a=imageattr:97 recv [x=800,y=640,sar=1.1] \
>                     send [x=336,y=256]
> 
> So after message 2, Bob cannot send anything, since he can't reproduce
> what Alice asked for, right?
> 
> And after message 2, Alice can only send 800x640 with a sar of 1.1,
> right? It's forbidden for Alice to send a noncompensated image, or a
> 640x480 image?
> 
> Again, it's the use of the words "wishes" and "desires" that I want to
> be sure I understand correctly - as used in RFC 6236, they both mean
> "demand" / "require" / "MUST". Right?
> 
> > Roni
> >
> >> -----Original Message-----
> >> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> >> Sent: Thursday, April 19, 2012 11:48 AM
> >> To: Roni Even
> >> Cc: rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Resolution negotiation - mandatory / advisory
> >> resolutions
> >>
> >> On 04/19/2012 10:18 AM, Roni Even wrote:
> >>> Hi Harald,
> >>> The reason for the language was that this is really what the
> >>> receiver wished to receive.
> >> The strength of his wish may be the issue here....
> >>>    The issue is that at least for H.264 there is no mandatory
> >>> requirement for the encoder to support all resolutions within the
> >>> supported level. The receiver must be able to crop and scale.
> >> I couldn't parse that.
> >>
> >> Does the sender have to (MUST) send a resolution within the
> >> boundaries of the "recv" spec, or can he send something outside it?
> >> (ie if the receiver sends recv [x=320,y=200], can the sender send
> >> something with x=320,y=240, or is he completely constrained?)
> >>
> >> What does the receiver have to be able to do?
> >>
> >>                      Harald
> >>> Roni Even
> >>>
> >>>> -----Original Message-----
> >>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >>>> Behalf Of Harald Alvestrand
> >>>> Sent: Thursday, April 19, 2012 10:47 AM
> >>>> To: rtcweb@ietf.org
> >>>> Subject: Re: [rtcweb] Resolution negotiation - mandatory /
> advisory
> >>>> resolutions
> >>>>
> >>>> When discussing the issue of resolution negotiation locally, we
> >> found
> >>>> a
> >>>> quandary:
> >>>>
> >>>> We have two different negotiation needs:
> >>>> - Maximum resolution that's possible to use in a given context
> >>>> - The preferred resolution for a given context
> >>>>
> >>>> The current imageattr= spec (RFC 6236) doesn't specify clearly
> >>>> whether the result of the negotiation is to be considered
> mandatory
> >>>> or advisory (it uses the term "wishes" a lot), but the description
> >>>> reads most consistently if one assumes that it is a "MUST" type
> >> request.
> >>>> One interpretation is that the highest functionality available in
> >> the
> >>>> spec is the "limit", while the highest q value is the
> "preference",
> >>>> thus:
> >>>>
> >>>> recv [x=320,y=200,q=1.0] [x=200:1024,y=160:768,q=0.1]
> >>>>
> >>>> would indicate that the receiver is capable of 200x160 to
> 1024x768,
> >>>> but wants 320x200.
> >>>>
> >>>> This maps reasonably to the capabilities spec's talk about
> >> "mandatory"
> >>>> vs "optional" capabilities.
> >>>>
> >>>> Does this interpretation make sense?
> >>>>
> >>>>                          Harald
> >>>>
> >>>>
> >>>> On 04/12/2012 10:46 AM, Harald Alvestrand wrote:
> >>>>> Hello folks,
> >>>>>
> >>>>> after the IETF, I've attempted to gather together information
> >>>>> about resolution negotiation and what we might need to do there.
> >>>>>
> >>>>> I pulled together the paragraphs below - this may want to be
> >>>>> incorporated in an I-D on "SDP usage in RTCWEB", or it may want
> to
> >>>>> go somewhere else. Chairs' guidance is appreciated.
> >>>>>
> >>>>> At the moment, I have it in xml2rfc format, so it should be easy
> >>>>> to
> >>>> do
> >>>>> cut-and-paste on it, but don't want to push it as a separate I-D
> >>>>> unless that's what the chairs desire.
> >>>>> Note - there are some QUERY sections in there - I'm unsure what
> >>>>> those should be resolved to.
> >>>>>
> >>>>> Comments?
> >>>>>
> >>>>>                            Harald
> >>>>>
> >>>>> 2.  Requirements
> >>>>>
> >>>>>      The relevant requirement for video resolution negotiation
> >>>>> from
> >> the
> >>>>>      RTCWEB use cases document
> >>>>>      [I-D.ietf-rtcweb-use-cases-and-requirements] is:
> >>>>>
> >>>>>      o  F25 The browser SHOULD use encoding of streams suitable
> >>>>> for
> >> the
> >>>>>         current rendering (e.g. video display size) and SHOULD
> >> change
> >>>>>         parameters if the rendering changes during the session.
> >>>>>
> >>>>>      The resulting need is:
> >>>>>
> >>>>>      o  To negotiate a maximum spatial resolution for all videos
> >>>>> at
> >>>> call
> >>>>>         setup time
> >>>>>
> >>>>>      o  To negotiate a maximum temporal resolution ("frame rate")
> >>>> across
> >>>>>         all videos at call setup time
> >>>>>
> >>>>>      o  To indicate the desire of the recipient for a particular
> >>>> spatial
> >>>>>         or temporal resolution on a particular video source
> >>>>>
> >>>>>      o  To indicate the intent of the sender to send a video
> >>>>> source in
> >>>> a
> >>>>>         particular spatial or temporal resolution
> >>>>>
> >>>>>      This document does not mention other requirements.
> >>>> _______________________________________________
> >>>> rtcweb mailing list
> >>>> rtcweb@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/rtcweb
> >


From kpfleming@digium.com  Thu Apr 19 07:20:49 2012
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Subject: Re: [rtcweb] Confirmation of F2F Interim meeting
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On 04/19/2012 04:21 AM, Magnus Westerlund wrote:
> RTCWEB WG,
>
> The dates for the upcoming interim is off the hold. I can confirm AD
> approval, and that we will hold an Face to Face interim meeting on the
> 12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a
> meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for
> each of those days.

Is the W3C meeting 'open attendance' like the IETF meeting will be, or 
does it require membership or sponsorship?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From harald@alvestrand.no  Thu Apr 19 08:00:43 2012
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Subject: Re: [rtcweb] Confirmation of F2F Interim meeting
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On 04/19/2012 04:20 PM, Kevin P. Fleming wrote:
> On 04/19/2012 04:21 AM, Magnus Westerlund wrote:
>> RTCWEB WG,
>>
>> The dates for the upcoming interim is off the hold. I can confirm AD
>> approval, and that we will hold an Face to Face interim meeting on the
>> 12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a
>> meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for
>> each of those days.
>
> Is the W3C meeting 'open attendance' like the IETF meeting will be, or 
> does it require membership or sponsorship?
The WG chairs can welcome people into the meeting as "observers with 
speaking privilleges".
All we want as formality is that you send us a note beforehand asking to 
be an observer.
We will accept the request.

              Harald



From randell-ietf@jesup.org  Thu Apr 19 08:33:09 2012
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Subject: Re: [rtcweb] Proposal for H.263 baseline codec
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On 4/19/2012 7:32 AM, Marshall Eubanks wrote:
> On Thu, Apr 19, 2012 at 3:15 AM, Randell Jesup<randell-ietf@jesup.org>  wrote:
>> On 3/29/2012 1:12 PM, Silvia Pfeiffer wrote:
>>>
>>> Very interesting discussion. Out of personal curiosity, I have some
>>> questions inline.
>>>
>>> On Thu, Mar 29, 2012 at 7:20 AM, Stephan Wenger<stewe@stewe.org>    wrote:
>>>>
>>>>
>>>> The most commonly cited timeline for a widely in use technology to be
>>>> "save" from a patent viewpoint, based on equitable defenses such as
>>>> laches
>>>> (in the US) is six years.
>>>
>>>
>>> Very interesting to know the usually used limit on the number of
>>> years. IIUC that would make Speex, Vorbis and Theora "save" codecs,
>>> since they were released in 2003, 2000, and 2004 respectively?
>>
>>
>> IANAL:  No, since the defense (with some notable exceptions) is by a
>> specific person against delayed assertion of a suit against them, so someone
>> using Speex for example could not use that defense for circa 6 years after
>> they began using it.  Note that publication of a spec doesn't violate a
>> patent (though using it may).  In practice the defense might be even more
>> limited, for example if the infringer made very limited use of the patent
>> for years such that the owner didn't even know they were infringing; it's
>> possible the court might start the clock when the owner knew or could/should
>> have known of the infringement.
>>
>
> I found a fairly detailed analysis of the defense of laches in patent
> cases starting at page 27 of this
>
> http://www.fdml.com/defenses.pdf
>
> Reading this dense legal reasoning makes me very nervous about relying
> on any of this for any IETF decision. Engineers should not
> pretend they are lawyers.

Quite true (though I've played one on TV^H^H^H^H^H^H^H^H^H amateur 
lawyer quite a bit in a former job).  That document agrees with my 
comments (IANAL!!) so far as I can tell, and the basic answer is still 
"No, do not assume you're safe because you're using a codec published 
more than 6 years ago."  Not assuming you're safe is always the safe 
thing to do.  ;-)


-- 
Randell Jesup
randell-ietf@jesup.org

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--20cf3071c7e0bb8d0b04be0bcd97
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For folks that are not on the CLUE WG mailing list, it is highly likely
that we will be able to hold our interim meeting in Stockholm on June
7th-8th.  If you would like to also attend the CLUE meeting, then if you
could please indicate Stockholm as your location preference in our poll.

Thanks,
Mary
CLUE WG co-chair

---------- Forwarded message ----------
From: Mary Barnes <mary.ietf.barnes@gmail.com>
Date: Thu, Apr 19, 2012 at 12:46 PM
Subject: Re: [clue] Fwd: [rtcweb] URGENT: Dates for upcoming interim ON HOL=
D
To: Marshall Eubanks <marshall.eubanks@gmail.com>
Cc: clue <clue@ietf.org>


As I've highlighted in previous emails, we have a doodle for location
selection for the CLUE meeting:
http://www.doodle.com/wwrdycma9ymrdn3r
This poll closes at noon Pacific today - i.e., in 1.5 hours.  So, we will
make the decision for the location at the end of today.

Note that our date has been set for June 7-8 based on our original poll:
http://doodle.com/nm3pp69znr3286cy

Please note, that there are significantly fewer people that have responded
to the location doodle poll than people that indicated they could attend
the meeting on the 7th-8th, per the poll for meeting dates.  So we'll
assume whomever doesn't respond to the location poll is okay with any of
the locations.  Folks that want to attend both meetings are strongly
encouraged to indicate their preference for Stockholm unless you want to
spend the weekend traveling from Boston or San Jose.

Thanks,
Mary.


On Thu, Apr 19, 2012 at 6:35 AM, Marshall Eubanks <
marshall.eubanks@gmail.com> wrote:

> FYI
>
>
> ---------- Forwarded message ----------
> From: Magnus Westerlund <magnus.westerlund@ericsson.com>
> Date: Thu, Apr 19, 2012 at 5:18 AM
> Subject: Re: [rtcweb] URGENT: Dates for upcoming interim ON HOLD
> To: Ted Hardie <ted.ietf@gmail.com>
> Cc: Cullen Jennings <fluffy@cisco.com>, "rai-ads@tools.ietf.org"
> <rai-ads@tools.ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,
> "clue-chairs@tools.ietf.org" <clue-chairs@tools.ietf.org>
>
>
> RTCWEB WG,
>
> The dates for the upcoming interim is off the hold. I can confirm AD
> approval, and that we will hold an Face to Face interim meeting on the
> 12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a
> meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for
> each of those days.
>
> As a Host I can't confirm the location within Stockholm yet. We are
> aiming on either a location in Kista or in the central part of town.
> There are some hotels out in Kista, but even if we end up in Kista
> staying in central Stockholm allows for a short (17 min) subway ride
> from T-centralen plus a no more than 10 min walk to the Kista venue.
> Making it a very viable option to book a hotel room in the area around
> T-centralen or on Kungsholmen with easy access to the blue-line subway.
>
> Cheers
>
> Magnus Westerlund
>
>
>
>
> On 2012-04-12 19:42, Ted Hardie wrote:
> > Please hold off making travel plans based on this timing.   Because
> > neither RAI area director will be available during this timing, they
> > have asked to consider shifting this timing to the previous week, June
> > 7th and 8th.  Until the chairs have had a chance to discuss this,
> > please hold off on booking travel plans.
> >
> > My apologies for this scramble,
> >
> > Ted Hardie
> >
> > On Thu, Apr 12, 2012 at 8:20 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
> >> After discussion between the WEBRTC and RTCWEB chairs, consulting the
> >> doodle poll, and working through the potential for co-location with
> >> CLUE, we have decided to set the date for the RTCWEB interim at June
> >> 12 & 13th, to be hosted by Ericsson in Stockholm.  We understand that
> >> the WEBRTC chairs will hold their meeting on June 11th.  If CLUE
> >> wishes to co-locate their meeting, the 14th and 15th will be
> >> available.  This would not have been possible on the previous week,
> >> given the national holiday in Sweden on the Wednesday of that week.
> >>
> >> We very much appreciated the work Eric put into his analysis, and in
> >> general we will try to prefer hubs; in this particular case, however,
> >> a large fraction of the Western European participants are either based
> >> in Stockholm or must fly through it to reach one of the larger hubs.
> >> Given the availability of a host and this data, we decided to select
> >> Stockholm for this meeting.
> >>
> >> regards,
> >>
> >> Ted Hardie, for the chairs.
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> 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
> _______________________________________________
> clue mailing list
> clue@ietf.org
> https://www.ietf.org/mailman/listinfo/clue
>

--20cf3071c7e0bb8d0b04be0bcd97
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For folks that are not on the CLUE WG mailing list, it is highly likely tha=
t we will be able to hold our interim meeting in Stockholm on June 7th-8th.=
 =A0If you would like to also attend the CLUE meeting, then if you could pl=
ease indicate Stockholm as your location preference in our poll.=A0<div>
<br></div><div>Thanks,</div><div>Mary</div><div>CLUE WG co-chair<br><br><di=
v class=3D"gmail_quote">---------- Forwarded message ----------<br>From: <b=
 class=3D"gmail_sendername">Mary Barnes</b> <span dir=3D"ltr">&lt;<a href=
=3D"mailto:mary.ietf.barnes@gmail.com">mary.ietf.barnes@gmail.com</a>&gt;</=
span><br>
Date: Thu, Apr 19, 2012 at 12:46 PM<br>Subject: Re: [clue] Fwd: [rtcweb] UR=
GENT: Dates for upcoming interim ON HOLD<br>To: Marshall Eubanks &lt;<a hre=
f=3D"mailto:marshall.eubanks@gmail.com">marshall.eubanks@gmail.com</a>&gt;<=
br>
Cc: clue &lt;<a href=3D"mailto:clue@ietf.org">clue@ietf.org</a>&gt;<br><br>=
<br>As I&#39;ve highlighted in previous emails, we have a doodle for locati=
on selection for the CLUE meeting:=A0<br><div><a href=3D"http://www.doodle.=
com/wwrdycma9ymrdn3r" target=3D"_blank">http://www.doodle.com/wwrdycma9ymrd=
n3r</a></div>
<div>This poll closes at noon Pacific today - i.e., in 1.5 hours. =A0So, we=
 will make the decision for the location at the end of today.</div>
<div><br></div><div>Note that our date has been set for June 7-8 based on o=
ur original poll: =A0=A0<a href=3D"http://doodle.com/nm3pp69znr3286cy" targ=
et=3D"_blank">http://doodle.com/nm3pp69znr3286cy</a></div><div><br></div><d=
iv>
Please note, that there are significantly fewer people that have responded =
to the location doodle poll than people that indicated they could attend th=
e meeting on the 7th-8th, per the poll for meeting dates. =A0So we&#39;ll a=
ssume whomever doesn&#39;t respond to the location poll is okay with any of=
 the locations. =A0Folks that want to attend both meetings are strongly enc=
ouraged to indicate their preference for Stockholm unless you want to spend=
 the weekend traveling from Boston or San Jose.=A0</div>

<div><br></div><div>Thanks,</div><div>Mary.=A0<div><div class=3D"h5"><br><b=
r><div class=3D"gmail_quote">On Thu, Apr 19, 2012 at 6:35 AM, Marshall Euba=
nks <span dir=3D"ltr">&lt;<a href=3D"mailto:marshall.eubanks@gmail.com" tar=
get=3D"_blank">marshall.eubanks@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">FYI<br>
<div><div><br>
<br>
---------- Forwarded message ----------<br>
From: Magnus Westerlund &lt;<a href=3D"mailto:magnus.westerlund@ericsson.co=
m" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;<br>
Date: Thu, Apr 19, 2012 at 5:18 AM<br>
Subject: Re: [rtcweb] URGENT: Dates for upcoming interim ON HOLD<br>
To: Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">=
ted.ietf@gmail.com</a>&gt;<br>
Cc: Cullen Jennings &lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blan=
k">fluffy@cisco.com</a>&gt;, &quot;<a href=3D"mailto:rai-ads@tools.ietf.org=
" target=3D"_blank">rai-ads@tools.ietf.org</a>&quot;<br>
&lt;<a href=3D"mailto:rai-ads@tools.ietf.org" target=3D"_blank">rai-ads@too=
ls.ietf.org</a>&gt;, &quot;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_bl=
ank">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org" targe=
t=3D"_blank">rtcweb@ietf.org</a>&gt;,<br>

&quot;<a href=3D"mailto:clue-chairs@tools.ietf.org" target=3D"_blank">clue-=
chairs@tools.ietf.org</a>&quot; &lt;<a href=3D"mailto:clue-chairs@tools.iet=
f.org" target=3D"_blank">clue-chairs@tools.ietf.org</a>&gt;<br>
<br>
<br>
RTCWEB WG,<br>
<br>
The dates for the upcoming interim is off the hold. I can confirm AD<br>
approval, and that we will hold an Face to Face interim meeting on the<br>
12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a<br>
meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for<br>
each of those days.<br>
<br>
As a Host I can&#39;t confirm the location within Stockholm yet. We are<br>
aiming on either a location in Kista or in the central part of town.<br>
There are some hotels out in Kista, but even if we end up in Kista<br>
staying in central Stockholm allows for a short (17 min) subway ride<br>
from T-centralen plus a no more than 10 min walk to the Kista venue.<br>
Making it a very viable option to book a hotel room in the area around<br>
T-centralen or on Kungsholmen with easy access to the blue-line subway.<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
<br>
<br>
<br>
On 2012-04-12 19:42, Ted Hardie wrote:<br>
&gt; Please hold off making travel plans based on this timing. =A0 Because<=
br>
&gt; neither RAI area director will be available during this timing, they<b=
r>
&gt; have asked to consider shifting this timing to the previous week, June=
<br>
&gt; 7th and 8th. =A0Until the chairs have had a chance to discuss this,<br=
>
&gt; please hold off on booking travel plans.<br>
&gt;<br>
&gt; My apologies for this scramble,<br>
&gt;<br>
&gt; Ted Hardie<br>
&gt;<br>
&gt; On Thu, Apr 12, 2012 at 8:20 AM, Ted Hardie &lt;<a href=3D"mailto:ted.=
ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt; wrote:<br>
&gt;&gt; After discussion between the WEBRTC and RTCWEB chairs, consulting =
the<br>
&gt;&gt; doodle poll, and working through the potential for co-location wit=
h<br>
&gt;&gt; CLUE, we have decided to set the date for the RTCWEB interim at Ju=
ne<br>
&gt;&gt; 12 &amp; 13th, to be hosted by Ericsson in Stockholm. =A0We unders=
tand that<br>
&gt;&gt; the WEBRTC chairs will hold their meeting on June 11th. =A0If CLUE=
<br>
&gt;&gt; wishes to co-locate their meeting, the 14th and 15th will be<br>
&gt;&gt; available. =A0This would not have been possible on the previous we=
ek,<br>
&gt;&gt; given the national holiday in Sweden on the Wednesday of that week=
.<br>
&gt;&gt;<br>
&gt;&gt; We very much appreciated the work Eric put into his analysis, and =
in<br>
&gt;&gt; general we will try to prefer hubs; in this particular case, howev=
er,<br>
&gt;&gt; a large fraction of the Western European participants are either b=
ased<br>
&gt;&gt; in Stockholm or must fly through it to reach one of the larger hub=
s.<br>
&gt;&gt; Given the availability of a host and this data, we decided to sele=
ct<br>
&gt;&gt; Stockholm for this meeting.<br>
&gt;&gt;<br>
&gt;&gt; regards,<br>
&gt;&gt;<br>
&gt;&gt; Ted Hardie, for the chairs.<br>
&gt;<br>
<br>
<br>
--<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287" target=3D"_blank">+46 10 7148287</a>=
<br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079" target=3D"_blank">+46 73 0949079</=
a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><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>
</div></div><div><div>_______________________________________________<br>
clue mailing list<br>
<a href=3D"mailto:clue@ietf.org" target=3D"_blank">clue@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/clue" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/clue</a><br>
</div></div></blockquote></div><br></div></div></div>
</div><br></div>

--20cf3071c7e0bb8d0b04be0bcd97--

From mary.ietf.barnes@gmail.com  Thu Apr 19 15:52:05 2012
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References: <4F8FD90B.5040409@ericsson.com>
Date: Thu, 19 Apr 2012 17:52:03 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Content-Type: multipart/alternative; boundary=bcaec54ee6be384b8a04be1005ce
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Confirmation of F2F Interim meeting
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--bcaec54ee6be384b8a04be1005ce
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I looked on Google maps and it indicates an 18 minute drive to Kista, but
it looks like a 45 min trip as opposed to 30.  Of course, I just did the
calculation from Central Kista to Central Stockholm, so that doesn't
account for the walking to/from meeting location (which perhaps is closer
than the route I measured) nor to/from hotels in the City center.

Also, just to confirm, Ericsson is also still willing to host the CLUE
meeting?

Thanks,
Mary.

On Thu, Apr 19, 2012 at 4:21 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> RTCWEB WG,
>
> The dates for the upcoming interim is off the hold. I can confirm AD
> approval, and that we will hold an Face to Face interim meeting on the
> 12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a
> meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for
> each of those days.
>
> As a Host I can't confirm the location within Stockholm yet. We are
> aiming on either a location in Kista or in the central part of town.
> There are some hotels out in Kista, but even if we end up in Kista
> staying in central Stockholm allows for a short (17 min) subway ride
> from T-centralen plus a no more than 10 min walk to the Kista venue.
> Making it a very viable option to book a hotel room in the area around
> T-centralen or on Kungsholmen with easy access to the blue-line subway.
>
> Cheers
>
> Magnus Westerlund
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> 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
>

--bcaec54ee6be384b8a04be1005ce
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I looked on Google maps and it indicates an 18 minute drive to Kista, but i=
t looks like a 45 min trip as opposed to 30. =A0Of course, I just did the c=
alculation from Central Kista to Central Stockholm, so that doesn&#39;t acc=
ount for the walking to/from meeting location (which perhaps is closer than=
 the route I measured) nor to/from hotels in the City center.=A0<div>
<br></div><div>Also, just to confirm, Ericsson is also still willing to hos=
t the CLUE meeting? =A0=A0</div><div><br></div><div>Thanks,<br><div>Mary. =
=A0=A0</div><div><br><div class=3D"gmail_quote">On Thu, Apr 19, 2012 at 4:2=
1 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.west=
erlund@ericsson.com">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<b=
r>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">RTCWEB WG,<br>
<br>
The dates for the upcoming interim is off the hold. I can confirm AD<br>
approval, and that we will hold an Face to Face interim meeting on the<br>
12th and 13th of June in Stockholm. The W3C WebRTC WG will hold a<br>
meeting in Stockholm on the 11th of June. Assume meeting 9.00-18.00 for<br>
each of those days.<br>
<br>
As a Host I can&#39;t confirm the location within Stockholm yet. We are<br>
aiming on either a location in Kista or in the central part of town.<br>
There are some hotels out in Kista, but even if we end up in Kista<br>
staying in central Stockholm allows for a short (17 min) subway ride<br>
from T-centralen plus a no more than 10 min walk to the Kista venue.<br>
Making it a very viable option to book a hotel room in the area around<br>
T-centralen or on Kungsholmen with easy access to the blue-line subway.<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
--<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<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:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.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></div>

--bcaec54ee6be384b8a04be1005ce--

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Date: Thu, 19 Apr 2012 23:42:39 -0500
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On Apr 19, 2012, at 10:32 AM, Randell Jesup wrote:

>>=20
>=20
> Quite true (though I've played one on TV^H^H^H^H^H^H^H^H^H amateur =
lawyer quite a bit in a former job).  That document agrees with my =
comments (IANAL!!) so far as I can tell, and the basic answer is still =
"No, do not assume you're safe because you're using a codec published =
more than 6 years ago."  Not assuming you're safe is always the safe =
thing to do.  ;-)
>=20

To paraphrase Whedon: You're never truly safe; the ancient ones are down =
there waiting. Miss a blood sacrifice, and there''s hell to pay.

Or,  Lubarsky's alternative cognate: There's always one more patent.

There is, however, a margin of safety in numbers. If many little red =
riding hoods have walked the path before you and others walk beside you, =
the chance of meeting a wolf are reduced. Or you might just meet a VERY =
HUNGRY wolf. But you have a chance of hearing the screams and being able =
to run away.

On the other hand, if you KNOW there's  a wolf on the hunt, stay home.

--
Dean=

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On 2012-04-20 00:52, Mary Barnes wrote:
> I looked on Google maps and it indicates an 18 minute drive to Kista,
> but it looks like a 45 min trip as opposed to 30.  Of course, I just did
> the calculation from Central Kista to Central Stockholm, so that doesn't
> account for the walking to/from meeting location (which perhaps is
> closer than the route I measured) nor to/from hotels in the City center. 

Well, why would you be driving? The subway system takes approx 17
minutes between T-centralen and Kista station. If you would driving to
and from central Stockholm in rush hour, which is what you will face
arriving between 8-9 in the morning and going back around 18, it can
definitely takes 30-40 minutes. Plus you have to pay congestion charges
when entering or leaving downtown. Plus the expense of parking.

As we haven't managed to get a solid booking on a meeting room yet I
can't say exactly where the meeting will be held. I still think staying
in the city center will be a good option for most as it gives easy
access to a very large selection of restaurants.

My recommendation is to not rent a car. Use the subway system or other
trains to get around. A 72 hour (3 day) ticket for the subway, local
trains and buses are 230 sek for unlimited travel. Anyone attending both
the CLUE and WebRTC meetings a 7-day unlimited travel ticket might be a
good buy, it cost 300 sek. Of course single tickets or strips for a
limited number of trips are available (For more information on Stockholm
local travel (http://sl.se/). The unlimited tickets even work on some
boat lines.

To get to and from the airport if you would staying in Kista you take a
Taxi approx 400 sek. If one goes downtown and have a hotel in the
vicinity of the central station then taking Arlanda Express
(http://www.arlandaexpress.com) is a good option. 260 sek for single or
490 for a return ticket. This is an express train that takes 20 min from
Arlanda to Centralen. Taxi Arlanda to central Stockholm is available for
a fixed price from most Taxi companies which is around 520 sek.

Important about Taxi in Stockholm. This is a de-regulated market. The
driver or their companies can basically set the fares in the meters
freely. The only requirement is that they have a yellow sticker visible
that gives you a comparison price for a trip. They also need to display
the drivers Taxi license. I strongly recommend that you use one of the
three big companies:

- Taxi 020 		http://www.taxi020.se/
- Taxi Stockholm  	http://www.taxistockholm.se/
- Taxi Kurir 		http://www.taxikurir.se/stockholm

The highest comparison price for a regular sized car shouldn't be more
than around 310 Sek. If it is more than 400 you are definitely getting
ripped off.

Taxi is quite expensive in Sweden. A plus side is that all the big
companies do take credit cards.

> 
> Also, just to confirm, Ericsson is also still willing to host the CLUE
> meeting?   

Yes, we are.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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: "Eggert, Lars" <lars@netapp.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] Confirmation of F2F Interim meeting
Thread-Index: AQHNHg3O+ERf37knqEGpA75TkxtX8ZajN0+AgACHaoCAABxBAA==
Date: Fri, 20 Apr 2012 08:37:48 +0000
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On Apr 20, 2012, at 8:56, Magnus Westerlund wrote:
> As we haven't managed to get a solid booking on a meeting room yet I
> can't say exactly where the meeting will be held. I still think =
staying
> in the city center will be a good option for most as it gives easy
> access to a very large selection of restaurants.

+1

Take it from someone who travelled to Kista many times on project work: =
you do want to stay downtown (on the blue line).

Lars=

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Date: Fri, 20 Apr 2012 09:42:04 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Confirmation of F2F Interim meeting
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I had not intention to drive, I used the train symbol option on Google
maps.  That's why my question as to perhaps Google maps calculations are
off for that route.   That calculation included walking time from what
Google maps has as city centers to the train stations, but that's probably
a reasonable buffer given the figures you note, which don't include the
wait times for trains.

Thanks,
Mary.

On Fri, Apr 20, 2012 at 1:56 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2012-04-20 00:52, Mary Barnes wrote:
> > I looked on Google maps and it indicates an 18 minute drive to Kista,
> > but it looks like a 45 min trip as opposed to 30.  Of course, I just di=
d
> > the calculation from Central Kista to Central Stockholm, so that doesn'=
t
> > account for the walking to/from meeting location (which perhaps is
> > closer than the route I measured) nor to/from hotels in the City center=
.
>
> Well, why would you be driving? The subway system takes approx 17
> minutes between T-centralen and Kista station. If you would driving to
> and from central Stockholm in rush hour, which is what you will face
> arriving between 8-9 in the morning and going back around 18, it can
> definitely takes 30-40 minutes. Plus you have to pay congestion charges
> when entering or leaving downtown. Plus the expense of parking.
>
> As we haven't managed to get a solid booking on a meeting room yet I
> can't say exactly where the meeting will be held. I still think staying
> in the city center will be a good option for most as it gives easy
> access to a very large selection of restaurants.
>
> My recommendation is to not rent a car. Use the subway system or other
> trains to get around. A 72 hour (3 day) ticket for the subway, local
> trains and buses are 230 sek for unlimited travel. Anyone attending both
> the CLUE and WebRTC meetings a 7-day unlimited travel ticket might be a
> good buy, it cost 300 sek. Of course single tickets or strips for a
> limited number of trips are available (For more information on Stockholm
> local travel (http://sl.se/). The unlimited tickets even work on some
> boat lines.
>
> To get to and from the airport if you would staying in Kista you take a
> Taxi approx 400 sek. If one goes downtown and have a hotel in the
> vicinity of the central station then taking Arlanda Express
> (http://www.arlandaexpress.com) is a good option. 260 sek for single or
> 490 for a return ticket. This is an express train that takes 20 min from
> Arlanda to Centralen. Taxi Arlanda to central Stockholm is available for
> a fixed price from most Taxi companies which is around 520 sek.
>
> Important about Taxi in Stockholm. This is a de-regulated market. The
> driver or their companies can basically set the fares in the meters
> freely. The only requirement is that they have a yellow sticker visible
> that gives you a comparison price for a trip. They also need to display
> the drivers Taxi license. I strongly recommend that you use one of the
> three big companies:
>
> - Taxi 020              http://www.taxi020.se/
> - Taxi Stockholm        http://www.taxistockholm.se/
> - Taxi Kurir            http://www.taxikurir.se/stockholm
>
> The highest comparison price for a regular sized car shouldn't be more
> than around 310 Sek. If it is more than 400 you are definitely getting
> ripped off.
>
> Taxi is quite expensive in Sweden. A plus side is that all the big
> companies do take credit cards.
>
> >
> > Also, just to confirm, Ericsson is also still willing to host the CLUE
> > meeting?
>
> Yes, we are.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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I had not intention to drive, I used the train symbol option on Google maps=
. =A0That&#39;s why my question as to perhaps Google maps calculations are =
off for that route. =A0 That calculation included walking time from what Go=
ogle maps has as city centers to the train stations, but that&#39;s probabl=
y a reasonable buffer given the figures you note, which don&#39;t include t=
he wait times for trains.=A0<div>
<br></div><div>Thanks,</div><div>Mary.=A0<div><br><div class=3D"gmail_quote=
">On Fri, Apr 20, 2012 at 1:56 AM, Magnus Westerlund <span dir=3D"ltr">&lt;=
<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsso=
n.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"im">On 2012-04-20 00:52, Mary =
Barnes wrote:<br>
&gt; I looked on Google maps and it indicates an 18 minute drive to Kista,<=
br>
&gt; but it looks like a 45 min trip as opposed to 30. =A0Of course, I just=
 did<br>
&gt; the calculation from Central Kista to Central Stockholm, so that doesn=
&#39;t<br>
&gt; account for the walking to/from meeting location (which perhaps is<br>
&gt; closer than the route I measured) nor to/from hotels in the City cente=
r.<br>
<br>
</div>Well, why would you be driving? The subway system takes approx 17<br>
minutes between T-centralen and Kista station. If you would driving to<br>
and from central Stockholm in rush hour, which is what you will face<br>
arriving between 8-9 in the morning and going back around 18, it can<br>
definitely takes 30-40 minutes. Plus you have to pay congestion charges<br>
when entering or leaving downtown. Plus the expense of parking.<br>
<br>
As we haven&#39;t managed to get a solid booking on a meeting room yet I<br=
>
can&#39;t say exactly where the meeting will be held. I still think staying=
<br>
in the city center will be a good option for most as it gives easy<br>
access to a very large selection of restaurants.<br>
<br>
My recommendation is to not rent a car. Use the subway system or other<br>
trains to get around. A 72 hour (3 day) ticket for the subway, local<br>
trains and buses are 230 sek for unlimited travel. Anyone attending both<br=
>
the CLUE and WebRTC meetings a 7-day unlimited travel ticket might be a<br>
good buy, it cost 300 sek. Of course single tickets or strips for a<br>
limited number of trips are available (For more information on Stockholm<br=
>
local travel (<a href=3D"http://sl.se/" target=3D"_blank">http://sl.se/</a>=
). The unlimited tickets even work on some<br>
boat lines.<br>
<br>
To get to and from the airport if you would staying in Kista you take a<br>
Taxi approx 400 sek. If one goes downtown and have a hotel in the<br>
vicinity of the central station then taking Arlanda Express<br>
(<a href=3D"http://www.arlandaexpress.com" target=3D"_blank">http://www.arl=
andaexpress.com</a>) is a good option. 260 sek for single or<br>
490 for a return ticket. This is an express train that takes 20 min from<br=
>
Arlanda to Centralen. Taxi Arlanda to central Stockholm is available for<br=
>
a fixed price from most Taxi companies which is around 520 sek.<br>
<br>
Important about Taxi in Stockholm. This is a de-regulated market. The<br>
driver or their companies can basically set the fares in the meters<br>
freely. The only requirement is that they have a yellow sticker visible<br>
that gives you a comparison price for a trip. They also need to display<br>
the drivers Taxi license. I strongly recommend that you use one of the<br>
three big companies:<br>
<br>
- Taxi 020 =A0 =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.taxi020.se/" ta=
rget=3D"_blank">http://www.taxi020.se/</a><br>
- Taxi Stockholm =A0 =A0 =A0 =A0<a href=3D"http://www.taxistockholm.se/" ta=
rget=3D"_blank">http://www.taxistockholm.se/</a><br>
- Taxi Kurir =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.taxikurir.se/stoc=
kholm" target=3D"_blank">http://www.taxikurir.se/stockholm</a><br>
<br>
The highest comparison price for a regular sized car shouldn&#39;t be more<=
br>
than around 310 Sek. If it is more than 400 you are definitely getting<br>
ripped off.<br>
<br>
Taxi is quite expensive in Sweden. A plus side is that all the big<br>
companies do take credit cards.<br>
<div class=3D"im"><br>
&gt;<br>
&gt; Also, just to confirm, Ericsson is also still willing to host the CLUE=
<br>
&gt; meeting?<br>
<br>
</div>Yes, we are.<br>
<br>
Cheers<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<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:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br></div></div>

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On 03/29/2012 09:53 AM, Stephan Wenger wrote:

> The other issue, though (the fact that the license grant extends only to
> the VP8 implementation as provided by google, and does not extent to
> derivative works such as hardware implementations) should be moderately
> alarming even for an open source person.  With respect to this clause, I
<snip>

This is concerning, even for us open source software distributors. In 
fact, a similar situation existed some time ago with iLBC; the license 
that GIPS offered covered only the code as distributed as part of the 
RFC (although the language stating this was quite poorly constructed), 
and that code had (and still has) fundamental issues (it has at least 
two places where it invokes 'undefined behavior' according to the C 
standard). When we fixed these problems in our distribution, and then I 
re-read the license, I realized that the license did not allow us to 
distribute this modified version. We later managed to get explicit 
permission from Google to distribute a modified version, as they have 
changed the license to no longer have this restriction.

The WebM patent license has much of the same problem, though: it only 
applies to distributions of the code made by Google, without changes (no 
derivative works). At least, this is my reading of it, being an engineer 
married to a lawyer who handles such things, but not a lawyer myself :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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On 04/20/2012 04:24 PM, Kevin P. Fleming wrote:
> On 03/29/2012 09:53 AM, Stephan Wenger wrote:
>
>> The other issue, though (the fact that the license grant extends only to
>> the VP8 implementation as provided by google, and does not extent to
>> derivative works such as hardware implementations) should be moderately
>> alarming even for an open source person.  With respect to this clause, I
> <snip>
>
> This is concerning, even for us open source software distributors. In
> fact, a similar situation existed some time ago with iLBC; the license
> that GIPS offered covered only the code as distributed as part of the
> RFC (although the language stating this was quite poorly constructed),
> and that code had (and still has) fundamental issues (it has at least
> two places where it invokes 'undefined behavior' according to the C
> standard). When we fixed these problems in our distribution, and then
> I re-read the license, I realized that the license did not allow us to
> distribute this modified version. We later managed to get explicit
> permission from Google to distribute a modified version, as they have
> changed the license to no longer have this restriction.
>
> The WebM patent license has much of the same problem, though: it only
> applies to distributions of the code made by Google, without changes
> (no derivative works). At least, this is my reading of it, being an
> engineer married to a lawyer who handles such things, but not a lawyer
> myself :-)
>
The answer that I have seen mentioned is simply that Google cannot know
what patents someone is infringing on if they extend or modify the code
or make an independent implementation.  One thing that I do know is a
completely independent implementation is being done by some of the
developers of x264 (and naming their project xvp8 as a result), and this
is happening with the implicit blessing of Google, because development
is being tracked in the #vp8 channel on FreeNode.

So, Google is fine with this kind of development happening.  However,
having explicit license for this kind of behavior would be much better,
and is sorely needed, considering all the other challenges facing VP8
adoption.

-- 
Libre Video
http://librevideo.org


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On 04/20/2012 10:24 PM, Kevin P. Fleming wrote:
> On 03/29/2012 09:53 AM, Stephan Wenger wrote:
>
>> The other issue, though (the fact that the license grant extends only to
>> the VP8 implementation as provided by google, and does not extent to
>> derivative works such as hardware implementations) should be moderately
>> alarming even for an open source person.  With respect to this clause, I
> <snip>
>
> This is concerning, even for us open source software distributors. In 
> fact, a similar situation existed some time ago with iLBC; the license 
> that GIPS offered covered only the code as distributed as part of the 
> RFC (although the language stating this was quite poorly constructed), 
> and that code had (and still has) fundamental issues (it has at least 
> two places where it invokes 'undefined behavior' according to the C 
> standard). When we fixed these problems in our distribution, and then 
> I re-read the license, I realized that the license did not allow us to 
> distribute this modified version. We later managed to get explicit 
> permission from Google to distribute a modified version, as they have 
> changed the license to no longer have this restriction.
>
> The WebM patent license has much of the same problem, though: it only 
> applies to distributions of the code made by Google, without changes 
> (no derivative works). At least, this is my reading of it, being an 
> engineer married to a lawyer who handles such things, but not a lawyer 
> myself :-)
>
This was already addressed by Serge' mail of April 3. See also the FAQ:

http://www.webmproject.org/about/faq/#does-the-vp8-patent-grant-also-extend-to-hardware-implementations

If this isn't clear enough, please explain why it is not, so that we can 
make it even clearer.




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From: Gregory Maxwell <gmaxwell@juniper.net>
To: "EXT - kpfleming@digium.com" <kpfleming@digium.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 20 Apr 2012 14:19:11 -0700
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pfleming@digium.com wrote:
>On 03/29/2012 09:53 AM, Stephan Wenger wrote:
>> The other issue, though (the fact that the license grant extends only to
>> the VP8 implementation as provided by google, and does not extent to
>> derivative works such as hardware implementations) should be moderately
> This is concerning, even for us open source software distributors. In
> fact, a similar situation existed some time ago with iLBC; the license
> that GIPS offered covered only the code as distributed as part of the
> RFC (although the language stating this was quite poorly constructed),

Stop. All of you are confused.

Maybe I can help=97

There are _two_  VP8 patent license grants:

One for the specification: http://www.webmproject.org/license/bitstream/
One for the software: http://www.webmproject.org/license/additional/

These grants are subtly different in the scope of what they cover.

There are important licensing drafting reasons for a construction
which distinguishes "an implementation" and "any implementation".
In particular, the license drafter writing this kind of a royalty free
license has a tricky time making sure that the permission is broad
enough to include all the correct coverage,  but not so over-broad that
a malicious implementer could (e.g.) add a search engine to their VP8
encoder and claim that the VP8 license entitles them to search patents.

So, the drafter says something like=97 "those patent claims [...] that
are necessarily infringed by implementation of this specification",
which achieves the desired limited scope=97 and that is _all_ you get
with the MPEG-LA pool patent licenses. This is what the Google bitstream
spec license provides.

But, if we're to be serious and honest about making a royalty free format
and implementation that permission isn't quite sufficient: The reference
implementation might practice some VP8 related techniques which are not
"necessarily infringed"=97 they're helpful performance enhancements,
they could simply be one option out of many ways of doing the same thing,
or could just be something that was incidentally included (or even a
malicious patent trap snake in the grass)=97 and so it's important that
everything in the reference software also be licensed=97 even if its not
"necessarily infringed" by the specification.   But since it's not a
goal to also license unrelated things that a downstream user may cram in
(e.g. a search engine), that license must be limited in scope to things
which are actually in the reference implementation: "This grant does not
include claims that would be infringed only as a consequence of further
modification of this implementation." (this language is actually almost
identical to the language for this purpose in the GPL, FWIW)

So, in summary:

You are licensed, via the bitstream license, for anything Google controls
which is necessary to implement the specification=97 in hardware,
software, home grown, or reference implementations.  This is pretty
much as strong a grant as you can find anyone offering for any other
format specification.

You are also licensed, via the additional rights license, for any
use of the VP8 reference implementation including modified versions,
limited to the extent that the modifications don't expand the scope of
the coverage. (Exactly as the GPL explicit patent grant does)

This has already been clarified by Google on this list:
"Google confirms that the VP8 patent grant applies to both
third-party hardware and software implementations of VP8."
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04006.html

I admit that the fact that there are two similar but distinct licenses
is a little confusing, but I don't understand why we have to go over
it again and again on this list.  I would think that the prior Google
comment would have been sufficiently definitive.

From kpfleming@digium.com  Fri Apr 20 14:50:48 2012
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On 04/20/2012 04:19 PM, Gregory Maxwell wrote:
> pfleming@digium.com wrote:
>> On 03/29/2012 09:53 AM, Stephan Wenger wrote:
>>> The other issue, though (the fact that the license grant extends only=
 to
>>> the VP8 implementation as provided by google, and does not extent to
>>> derivative works such as hardware implementations) should be moderate=
ly
>> This is concerning, even for us open source software distributors. In
>> fact, a similar situation existed some time ago with iLBC; the license
>> that GIPS offered covered only the code as distributed as part of the
>> RFC (although the language stating this was quite poorly constructed),
>
> Stop. All of you are confused.
>
> Maybe I can help=97
>
> There are _two_  VP8 patent license grants:
>
> One for the specification: http://www.webmproject.org/license/bitstream=
/
> One for the software: http://www.webmproject.org/license/additional/
>
> These grants are subtly different in the scope of what they cover.
>
> There are important licensing drafting reasons for a construction
> which distinguishes "an implementation" and "any implementation".
> In particular, the license drafter writing this kind of a royalty free
> license has a tricky time making sure that the permission is broad
> enough to include all the correct coverage,  but not so over-broad that
> a malicious implementer could (e.g.) add a search engine to their VP8
> encoder and claim that the VP8 license entitles them to search patents.
>
> So, the drafter says something like=97 "those patent claims [...] that
> are necessarily infringed by implementation of this specification",
> which achieves the desired limited scope=97 and that is _all_ you get
> with the MPEG-LA pool patent licenses. This is what the Google bitstrea=
m
> spec license provides.
>
> But, if we're to be serious and honest about making a royalty free form=
at
> and implementation that permission isn't quite sufficient: The referenc=
e
> implementation might practice some VP8 related techniques which are not
> "necessarily infringed"=97 they're helpful performance enhancements,
> they could simply be one option out of many ways of doing the same thin=
g,
> or could just be something that was incidentally included (or even a
> malicious patent trap snake in the grass)=97 and so it's important that
> everything in the reference software also be licensed=97 even if its no=
t
> "necessarily infringed" by the specification.   But since it's not a
> goal to also license unrelated things that a downstream user may cram i=
n
> (e.g. a search engine), that license must be limited in scope to things
> which are actually in the reference implementation: "This grant does no=
t
> include claims that would be infringed only as a consequence of further
> modification of this implementation." (this language is actually almost
> identical to the language for this purpose in the GPL, FWIW)
>
> So, in summary:
>
> You are licensed, via the bitstream license, for anything Google contro=
ls
> which is necessary to implement the specification=97 in hardware,
> software, home grown, or reference implementations.  This is pretty
> much as strong a grant as you can find anyone offering for any other
> format specification.
>
> You are also licensed, via the additional rights license, for any
> use of the VP8 reference implementation including modified versions,
> limited to the extent that the modifications don't expand the scope of
> the coverage. (Exactly as the GPL explicit patent grant does)
>
> This has already been clarified by Google on this list:
> "Google confirms that the VP8 patent grant applies to both
> third-party hardware and software implementations of VP8."
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04006.html
>
> I admit that the fact that there are two similar but distinct licenses
> is a little confusing, but I don't understand why we have to go over
> it again and again on this list.  I would think that the prior Google
> comment would have been sufficiently definitive.

Well, I apologize, somehow I missed that while catching up on this list.=20
Your explanation certainly covers my concerns; the patent grant allows=20
us to distribute modified versions, as long as they don't infringe on=20
any patents that aren't necessarily infringed by the reference=20
implementation. Thanks for the clue-bat :-)

--=20
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpflemin=
g
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sun, 22 Apr 2012 19:12:16 +0200
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Subject: [rtcweb] Add, remove, add, remove, add, remove a media stream (lines in SDP)
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Hi, a stream in an SDP cannot be removed but just "disabled" (by
setting its port to 0). In the same media session a participant could
add a video stream and "disable" it later multiple times.

Let's assume a media session initiated with just an audio stream:

1) A participant adds a video stream and the remote accepts it. This
involves a new media section in the SDP.
2) Later the same participant removes the video stream which means
setting its port to 0 in the SDP.
3) Later he adds video again.
4) ...and so on.

AFAIK there are two choices for step 2 and 3:

a) Releasing video resources and the UDP socket so when video is
enabled again a new line should be added to the SDP and new resources
(ICE, UDP socket) required.

b) Not releasing all the resources (keep the UDP socket but stop
sending/receiving data on it), and re-enabling it when video is added
again, so the same SDP line would be reused by setting its port to the
previous value.


Choice a) means more and more lines in the SDP for each video
activation. Option b) means not releasing resources after disabling a
media stream.

If I'm not wrong in the above text, which would be the WebRTC approach
for this topic?

Thanks a lot.

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

From christer.holmberg@ericsson.com  Sun Apr 22 23:41:45 2012
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Subject: Re: [rtcweb] Add, remove, add, remove, add, remove a media stream (lines in SDP)
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Hi,

I see no reason why you couldn't release resources when you disable media (=
read: set port to zero), but still re-use the associated m- line later.

Regards,

Christer

________________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of I=F1ak=
i Baz Castillo [ibc@aliax.net]
Sent: Sunday, April 22, 2012 8:12 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Add, remove, add, remove, add,        remove a media stre=
am (lines in SDP)

Hi, a stream in an SDP cannot be removed but just "disabled" (by
setting its port to 0). In the same media session a participant could
add a video stream and "disable" it later multiple times.

Let's assume a media session initiated with just an audio stream:

1) A participant adds a video stream and the remote accepts it. This
involves a new media section in the SDP.
2) Later the same participant removes the video stream which means
setting its port to 0 in the SDP.
3) Later he adds video again.
4) ...and so on.

AFAIK there are two choices for step 2 and 3:

a) Releasing video resources and the UDP socket so when video is
enabled again a new line should be added to the SDP and new resources
(ICE, UDP socket) required.

b) Not releasing all the resources (keep the UDP socket but stop
sending/receiving data on it), and re-enabling it when video is added
again, so the same SDP line would be reused by setting its port to the
previous value.


Choice a) means more and more lines in the SDP for each video
activation. Option b) means not releasing resources after disabling a
media stream.

If I'm not wrong in the above text, which would be the WebRTC approach
for this topic?

Thanks a lot.

--
I=F1aki Baz Castillo
<ibc@aliax.net>
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb=

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Subject: Re: [rtcweb] Add, remove, add, remove, add, remove a media stream (lines in SDP)
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2012/4/23 Christer Holmberg <christer.holmberg@ericsson.com>:
> Hi,
>
> I see no reason why you couldn't release resources when you disable media=
 (read: set port to zero), but still re-use the associated m- line later.

Maybe the previosly used UDP port is not available later. Probably
there are no more reasons :)


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

From saul@ag-projects.com  Mon Apr 23 00:41:32 2012
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Subject: Re: [rtcweb] Add, remove, add, remove, add, remove a media stream (lines in SDP)
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Hi,

On Apr 23, 2012, at 9:29 AM, I=F1aki Baz Castillo wrote:

> 2012/4/23 Christer Holmberg <christer.holmberg@ericsson.com>:
>> Hi,
>>=20
>> I see no reason why you couldn't release resources when you disable =
media (read: set port to zero), but still re-use the associated m- line =
later.
>=20
> Maybe the previosly used UDP port is not available later. Probably
> there are no more reasons :)
>=20

But when you disabled the stream you did set the port to zero, so when =
you resurrect the stream you'd allocate a new port. Were you thinking =
about setting the stream as inactive (a=3Dinactive in SDP terms) =
instead?

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [rtcweb] Add, remove, add, remove, add, remove a media stream (lines in SDP)
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2012/4/23 Sa=C3=BAl Ibarra Corretg=C3=A9 <saul@ag-projects.com>:
> On Apr 23, 2012, at 9:29 AM, I=C3=B1aki Baz Castillo wrote:
>> Maybe the previosly used UDP port is not available later. Probably
>> there are no more reasons :)
>>
>
> But when you disabled the stream you did set the port to zero, so when yo=
u resurrect the stream you'd allocate a new port.

The stack could try to allocate the same port as default bahavior.



> Were you thinking about setting the stream as inactive (a=3Dinactive in S=
DP terms) instead?

But setting a=3Dinactive just pauses the stream in a single direction
(according to RFC 4566). You could be interested in stopping sending
and receiving video.





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

From saul@ag-projects.com  Mon Apr 23 01:12:20 2012
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On Apr 23, 2012, at 10:03 AM, I=F1aki Baz Castillo wrote:

> 2012/4/23 Sa=FAl Ibarra Corretg=E9 <saul@ag-projects.com>:
>> On Apr 23, 2012, at 9:29 AM, I=F1aki Baz Castillo wrote:
>>> Maybe the previosly used UDP port is not available later. Probably
>>> there are no more reasons :)
>>>=20
>>=20
>> But when you disabled the stream you did set the port to zero, so =
when you resurrect the stream you'd allocate a new port.
>=20
> The stack could try to allocate the same port as default bahavior.
>=20

Well, then the stack needs to be fixed :-) The stack should 'forget' the =
port as soon as it sets it as zero in the stream, and allocate a new =
one, the same way it would do for a brand new stream.

>=20
>=20
>> Were you thinking about setting the stream as inactive (a=3Dinactive =
in SDP terms) instead?
>=20
> But setting a=3Dinactive just pauses the stream in a single direction
> (according to RFC 4566). You could be interested in stopping sending
> and receiving video.
>=20

IIRC inactive pauses the stream in both directions. For pausing it in a =
single direction you can use sendonly / recvonly.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [rtcweb] Add, remove, add, remove, add, remove a media stream (lines in SDP)
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2012/4/23 Sa=C3=BAl Ibarra Corretg=C3=A9 <saul@ag-projects.com>:
>>> But when you disabled the stream you did set the port to zero, so when =
you resurrect the stream you'd allocate a new port.
>>
>> The stack could try to allocate the same port as default behavior.
>>
>
> Well, then the stack needs to be fixed :-) The stack should 'forget' the =
port as soon as it sets it as zero in the stream, and allocate a new one, t=
he same way it would do for a brand new stream.

The WebRTCP stack is being designed right now :)



>>> Were you thinking about setting the stream as inactive (a=3Dinactive in=
 SDP terms) instead?
>>
>> But setting a=3Dinactive just pauses the stream in a single direction
>> (according to RFC 4566). You could be interested in stopping sending
>> and receiving video.
>>
>
> IIRC inactive pauses the stream in both directions. For pausing it in a s=
ingle direction you can use sendonly / recvonly.

You are totally right, my fault. Anyhow setting a=3Dinactive means that
resources are not freed (for example, RTCP must be still sent). Not
sure if this is good or bad.


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

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Subject: Re: [rtcweb] Add, remove, add, remove, add, remove a media stream (lines in SDP)
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On 4/23/2012 2:40 AM, Christer Holmberg wrote:
> Hi,
>
> I see no reason why you couldn't release resources when you disable media (read:
> set port to zero), but still re-use the associated m- line later.

Exactly - m-lines can't be removed, but they can be re-used - and the 
critical point is that the port doesn't have to match the previous port. 
  So generally they should be no worse than a high-water-mark of the 
number of audio and video streams in use at any one time.

Of course with bundle the port issue is simpler.

> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Iñaki Baz Castillo [ibc@aliax.net]
>
> Hi, a stream in an SDP cannot be removed but just "disabled" (by
> setting its port to 0). In the same media session a participant could
> add a video stream and "disable" it later multiple times.
>
> Let's assume a media session initiated with just an audio stream:
>
> 1) A participant adds a video stream and the remote accepts it. This
> involves a new media section in the SDP.
> 2) Later the same participant removes the video stream which means
> setting its port to 0 in the SDP.
> 3) Later he adds video again.
> 4) ...and so on.
>
> AFAIK there are two choices for step 2 and 3:
>
> a) Releasing video resources and the UDP socket so when video is
> enabled again a new line should be added to the SDP and new resources
> (ICE, UDP socket) required.
>
> b) Not releasing all the resources (keep the UDP socket but stop
> sending/receiving data on it), and re-enabling it when video is added
> again, so the same SDP line would be reused by setting its port to the
> previous value.
>
>
> Choice a) means more and more lines in the SDP for each video
> activation. Option b) means not releasing resources after disabling a
> media stream.
>
> If I'm not wrong in the above text, which would be the WebRTC approach
> for this topic?

-- 
Randell Jesup
randell-ietf@jesup.org

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2012/4/23 Randell Jesup <randell-ietf@jesup.org>:
> Exactly - m-lines can't be removed, but they can be re-used -

> and the critical point is that the port doesn't have to match the previou=
s port.

Definitely I need a coffee. No idea why I assumed that the same port
should be re-used. So things become easier:

- Reuse the same m-line.
- Set port to zero when disabling the stream and free resources.
- Allocate a new port when re-enabling it and write the new port in the m-l=
ine.

Sorry, I was creating a non existing issue.



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

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--485b397dd77560654904be63fcd1
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It's needed to indicate to the offerer that the answer is not final; any
resources associated with the offer should not yet be released.

One example where this may be useful is when using RTCP mux; the initial
answer may indicate that mux is to be used, but the final answer may come
from a different endpoint and choose not to use mux. Using SDP_PRANSWER
allows the offerer to honor the second answer, since the RTCP transport is
not discarded by the first answer.

On Mon, Apr 16, 2012 at 11:52 AM, Henry Lum <Henry.Lum@genesyslab.com>wrote=
:

>  It=E2=80=99s unclear to me what the purpose of SDP_PRANSWER is really fo=
r and
> why it needs to be there. Originally I thought it was meant for early
> media, but section 6.1.4 paragraph 3 says that it should not result in
> starting of media transmission. Is there a reason to provide an answer
> before media is ready to be transmitted? There are separate handles for
> processing ICE candidates so I don=E2=80=99t see a need to have provision=
al
> answers. ****
>
> ** **
>
> I am not proposing PRANSWER for handling early media though, since I
> believe early media is more of an application level issue (it=E2=80=99s m=
ore like
> late-billing on the application side) and should be considered as a
> separate offer/answer if needed.****
>
> ** **
>
> Regards,****
>
> Henry ****
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--485b397dd77560654904be63fcd1
Content-Type: text/html; charset=UTF-8
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<div class=3D"gmail_extra">It&#39;s needed to indicate to the offerer that =
the answer is not final; any resources associated with the offer should not=
 yet be released.</div><div class=3D"gmail_extra"><br></div><div class=3D"g=
mail_extra">


One example where this may be useful is when using RTCP mux; the initial an=
swer may indicate that mux is to be used, but the final answer may come fro=
m a different endpoint and choose not to use mux. Using SDP_PRANSWER allows=
 the offerer to honor the second answer, since the RTCP transport is not di=
scarded by the first answer.<br>


<br><div class=3D"gmail_quote">On Mon, Apr 16, 2012 at 11:52 AM, Henry Lum =
<span dir=3D"ltr">&lt;<a href=3D"mailto:Henry.Lum@genesyslab.com" target=3D=
"_blank">Henry.Lum@genesyslab.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">











<div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple">

<div>

<p class=3D"MsoNormal">It=E2=80=99s unclear to me what the purpose of SDP_P=
RANSWER is
really for and why it needs to be there. Originally I thought it was meant =
for early
media, but section 6.1.4 paragraph 3 says that it should not result in star=
ting
of media transmission. Is there a reason to provide an answer before media =
is
ready to be transmitted? There are separate handles for processing ICE cand=
idates
so I don=E2=80=99t see a need to have provisional answers. <u></u><u></u></=
p>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>

<p class=3D"MsoNormal">I am not proposing PRANSWER for handling early media=
 though,
since I believe early media is more of an application level issue (it=E2=80=
=99s
more like late-billing on the application side) and should be considered as=
 a
separate offer/answer if needed.<u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>

<p class=3D"MsoNormal">Regards,<u></u><u></u></p>

<p class=3D"MsoNormal">Henry <u></u><u></u></p>

</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>

--485b397dd77560654904be63fcd1--

From magnus.westerlund@ericsson.com  Tue Apr 24 00:19:42 2012
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On 2012-04-20 16:42, Mary Barnes wrote:
> I had not intention to drive, I used the train symbol option on Google
> maps.  That's why my question as to perhaps Google maps calculations are
> off for that route.   That calculation included walking time from what
> Google maps has as city centers to the train stations, but that's
> probably a reasonable buffer given the figures you note, which don't
> include the wait times for trains. 

Yes, I would expect that door to door it does take approximately 30
minutes. Yes, if you during rush hour just go down to the station you
will have to wait some few minutes on the next train. The train interval
is either 7,8 or 10 minutes around the times when the meeting is planned
to start and stop.

Staying downtown (if we have the meeting in Kista) makes it easier in
the evenings returning from dinner. Staying in Kista saves you some time
in the morning. But you most likely are having dinner downtown anyway so
you get the same amount of subway riding to do. I do want to point out
that the subway is safe and runs until 1.00 AM weekdays. If we will have
the meeting downtown staying in Kista is just adding travel with no
benefit.

cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Tue Apr 24 07:25:32 2012
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Subject: [rtcweb] Alternative Proposal for Dynamic Codec Parameter Change (Was: Re: Resolution negotiation - a contribution)
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Harald, WG,

This is posted as individual. When it comes to this topic I will not
engage in any chair activities because I do have an alternative proposal
based on
https://datatracker.ietf.org/doc/draft-westerlund-avtext-codec-operation-point/.


My proposal has some significant differences in how it works compared to
Harald's. I will start with a discussion of requirements, in addition to
Harald's, then an overview of my proposal, and ending with a discussion
of the differences between the proposals. This is quite long but I do
hope you will read it to the end.

Requirements
------------

Let's start with the requirements that Harald have written. I think they
should be formulated differently. First of all I think the requirements
are negotiated, indicated etc in the context of a peer connection.

Secondly, when it comes to maximum resolution and maximum frame-rate,
there are actually three different limits that I think are important;
Highest spatial resolution, highest frame-rate, and maximum complexity
for a given video stream. The maximum complexity is often expressed as
the number of macroblocks a video codec can process per second. This is
a well-established complexity measure, used as part of standardized
video codec's "level" definitions since the introduction of H.263 Annex
X in 2004. As this is basically a joint maximum requirement on the
maximum amount of pixels per frame times the maximum frame-rate, there
exist cases where this complexity figure actually is the constraining
factor forcing a sender or receiver to either request higher frame rate
and lower resolution or higher resolution and a lower frame rate.

The requirements should also be clearer in that one needs to handle
multiple SSRCs per RTP session, including multiple cameras or audio
streams (microphones) from a single end-point, where each stream can be
encoded using different parameter values.

I also think it is important that we consider what additional encoding
property parameters that would make sense for WebRTC to have a
possibility to dynamically negotiate, request and indicate.

Some additional use cases that should be considered in the requirements:

1) We have use cases including multi-party communication. One way to
realize these is to use a central node and I believe everyone agrees
that we should have good support for getting this usage to work well.
Thus in a basic star topology of different participants there is going
to be different path characteristics between the central node and each
participant.

1A) This makes it necessary to consider how one can ensure that the
central node can deliver appropriate rates. One way is to de-couple the
links by having the central node perform individual transcodings to
every participant. A simpler non-transcoding central node, only
forwarding streams between end-points, would have to enforce the lowest
path characteristics to all. If one don't want to transcode at the
central node and not use the lowest path characteristics to all, one
need to consider either simulcast or scalable video coding. Both
simulcast and scalable video coding result in that at least in the
direction from a participant to the central node one needs to use
multiple codec operation points. Either one per peer connection, which
is how I see simulcast being realized with today's API and
functionality, or using an encoding format supporting scalable coding
within a single peer connection.

1B) In cases where the central node has minimal functionality and
basically is a relay and an ICE plus DTLS-SRTP termination point (I
assume EKT to avoid having to do re-encryption), there is a need to be
able to handle sources from different participants. This puts extra
requirements on how to successfully negotiate the parameters. For
example changing the values for one media source should not force one to
renegotiate with everyone.

2) The non-centralized multiparty use case appears to equally or more
stress the need for having timely dynamic control. If each sender has a
number of peer-connections to its peers, it may use local audio levels
to determine if its media stream is to be sent or not. Thus the amount
of bit-rate and screen estate needed to display will rapidly change as
different users speaks. Thus the need for minimal delay when changing
preferences are important.

3) We also have speech and audio including audio-only use cases. For
audio there could also exist desire to request or indicate changes in
the audio bandwidth required, or usage of multi-channel.

4) Adaptation to legacy node from a central node in a multi-party
conference. In some use cases legacy nodes might have special needs that
are within the profiles a WebRTC end-point is capable of producing. Thus
the central node might request the nodes to constrain themselves to
particular payload types, audio bandwidth etc to meet a joining session
participant.

5) There appear to exist a need for expressing dynamic requests for
target bit-rate as one parameter. This can be supported by TMMBR
(RFC5104) but there exist additional transport related parameters could
help with the adaptation. These include MTU, limits on packet rate, and
amount of aggregation of audio frames in the payload.

Overview
--------

The basic idea in this proposal is to use JSEP to establish the outer
limits for behavior and then use Codec Operation Point (COP) proposal as
detailed in draft-westerlund-avtext-codec-operation-point to handle
dynamic changes during the session.

So highest resolution, frame-rate and maximum complexity are expressed
in JSEP SDP. Complexity is in several video codecs expressed by profile
and level. I know that VP8 currently doesn't have this but it is under
discussion when it comes to these parameters.

During the session the browser implementation detects when there is need
to use COP to do any of the following things.

A) Request new target values for codec operation, for example due to
that the GUI element displaying a video has changed due to window resize.

B) Indicate when the end-point in its role as sender change parameters.

In addition to just spatial resolution and video frame rate, I propose
that the following parameters are considered as parameters that could be
dynamically possible to indicate and request.

Spatial resolution (as x and y resolution), Frame-rate, Picture Aspect
ratio, Sample Aspect Ratio, Payload Type, Bit-rate, Token Bucket Size
(To control burstiness of sender), Channels, Sampling Rate, Maximum RTP
Packet Size, Maximum RTP Packet Rate, and Application Data Unit
Aggregation (to control amount of audio frames in the same RTP packet).


Differences
-----------

A) Using COP and using SDP based signaling for the dynamic changes are
two quite different models in relation to how the interaction happens.

For COP this all happens in the browser, normally initiated by the
browser's own determination that a COP request or notification is
needed. Harald's proposal appears to require that the JS initiate a
renegotiation. This puts a requirement on the implementation to listen
to the correct callbacks to know when changes happens, such as window
resize. To my knowledge there are not yet any proposals for how the
browser can initiate a JSEP renegotiation.

Thus COP has the advantage that there is no API changes to get browser
triggered parameter changes. W3C can select too but are not required to
add API methods to allow JS to make codec parameter requests.

The next thing is that COP does not require the JS application to have
code to detect and handle re-negotiation. This makes it simpler for the
basic application to get good behavior and they are not interrupted nor
do they need to handle JSEP & Offer/Answer state machine lock-out
effects due to dynamic changes.

How big impact these API issues have are unclear as W3C currently appear
not to have included any discussion of how the browser can initiate a
offer/answer exchange towards the JS when it determines a need to change
parameters.

But I am worried that using SDP and with an API that requires the
application to listen for triggers that could benefit from a codec
parameter renegotiation. This will likely only result in good behavior
for the JS application implementors that are really good and find out
what listeners and what signaling tricks are needed with JSEP to get
good performance. I would much rather prefer good behavior by default in
simple applications, i.e. using the default behavior that the browser
implementor have put in.

B) Using the media plane, i.e. RTCP for this signaling lets it in most
case go directly between the encoding and the decoding entity in the
code. There is no need to involve the JS nor the signaling server. One
issue of using JSEP and SDP is that the state machine lock-out effects
that can occur if one has sent an Offer. Then that browser may not be
able to send a new updated Offer reflecting the latest change until the
answer has been properly processed. COP doesn't have these limitations.
It can send a new parameter request immediately, only limited by RTCP
bandwidth restrictions. Using the media plane in my view guarantees that
COP is never worse than what the signaling plane can perform at its best.

C) As the general restrictions are determined in the initial
negotiation, COP doesn't have the issue that in-flight media streams can
become out of bounds. Thus no need for a two phase change of signaling
parameters.

D) Relying on draft-lennox-mmusic-sdp-source-selection-04 has several
additional implications that should be discussed separately. The draft
currently includes the following functionalities.

  D1) It contains a media stream pause proposal. This makes it subject
to the architectural debate currently ongoing in dispatch around
draft-westerlund-avtext-rtp-stream-pause-00 which is a competing
proposal for the same functionality.

  D2) The inclusion of max desired frame rate in a SSRC specific way

  D3) Extending the image attribute to be per SSRC specific expression
of desired resolutions.

  D4) Expressing relative priority in receiving different SSRCs.

  D5) Providing an "information" on a sent SSRC

  D6) Indication if the media sender is actively sending media using the
given SSRC.

Is it a correct observation that only D2 and D3 are required for the
functionality of Resolution negotiation?

E) The standardization situation is similar for both proposals. They are
both relying on Internet drafts that are currently individual
submissions. Both are partially caught in the architectural discussion
which was held in Paris in the DISPATCH WG around Media pause/resume
(draft-westerlund-avtext-rtp-stream-pause-00) and Media Stream Selection
(draft-westerlund-dispatch-stream-selection-00) on what the most
appropriate level of discussion are. This discussion will continue on
the RAI-Area mailing list.

F) As seen by the discussion on the mailing list the imageattr
definitions may not be a 100% match to what is desired with Harald's
proposal. I believe that COP's are more appropriate especially the
"target" values possibility. In addition these are still open for
adjustment and if they don't match WebRTC's requirements.

I would also like to point out that I believe this functionality is also
highly desirable for CLUE and that their requirements should be taken
into account. I do think that this is one of the aspects where having
matching functionality will make it much easier to have WebRTC to CLUE
interworking.

Thanks for reading all the way here!

Cheers

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------



From harald@alvestrand.no  Tue Apr 24 09:43:48 2012
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I'm happy to see this - there are a number of quibbles I want to make, 
but overall, I like it.

My big worry is the IPR issue - I think that negotiation of video size 
during a call is likely to be "MUST implement", and I don't want to 
require implementation of a protocol where people have filed non-RF IPR 
claims against it if there are viable alternatives.

               Harald

On 04/24/2012 04:25 PM, Magnus Westerlund wrote:
> Harald, WG,
>
> This is posted as individual. When it comes to this topic I will not
> engage in any chair activities because I do have an alternative proposal
> based on
> https://datatracker.ietf.org/doc/draft-westerlund-avtext-codec-operation-point/.
>
>
> My proposal has some significant differences in how it works compared to
> Harald's. I will start with a discussion of requirements, in addition to
> Harald's, then an overview of my proposal, and ending with a discussion
> of the differences between the proposals. This is quite long but I do
> hope you will read it to the end.
>
> Requirements
> ------------
>
> Let's start with the requirements that Harald have written. I think they
> should be formulated differently. First of all I think the requirements
> are negotiated, indicated etc in the context of a peer connection.
>
> Secondly, when it comes to maximum resolution and maximum frame-rate,
> there are actually three different limits that I think are important;
> Highest spatial resolution, highest frame-rate, and maximum complexity
> for a given video stream. The maximum complexity is often expressed as
> the number of macroblocks a video codec can process per second. This is
> a well-established complexity measure, used as part of standardized
> video codec's "level" definitions since the introduction of H.263 Annex
> X in 2004. As this is basically a joint maximum requirement on the
> maximum amount of pixels per frame times the maximum frame-rate, there
> exist cases where this complexity figure actually is the constraining
> factor forcing a sender or receiver to either request higher frame rate
> and lower resolution or higher resolution and a lower frame rate.
>
> The requirements should also be clearer in that one needs to handle
> multiple SSRCs per RTP session, including multiple cameras or audio
> streams (microphones) from a single end-point, where each stream can be
> encoded using different parameter values.
>
> I also think it is important that we consider what additional encoding
> property parameters that would make sense for WebRTC to have a
> possibility to dynamically negotiate, request and indicate.
>
> Some additional use cases that should be considered in the requirements:
>
> 1) We have use cases including multi-party communication. One way to
> realize these is to use a central node and I believe everyone agrees
> that we should have good support for getting this usage to work well.
> Thus in a basic star topology of different participants there is going
> to be different path characteristics between the central node and each
> participant.
>
> 1A) This makes it necessary to consider how one can ensure that the
> central node can deliver appropriate rates. One way is to de-couple the
> links by having the central node perform individual transcodings to
> every participant. A simpler non-transcoding central node, only
> forwarding streams between end-points, would have to enforce the lowest
> path characteristics to all. If one don't want to transcode at the
> central node and not use the lowest path characteristics to all, one
> need to consider either simulcast or scalable video coding. Both
> simulcast and scalable video coding result in that at least in the
> direction from a participant to the central node one needs to use
> multiple codec operation points. Either one per peer connection, which
> is how I see simulcast being realized with today's API and
> functionality, or using an encoding format supporting scalable coding
> within a single peer connection.
>
> 1B) In cases where the central node has minimal functionality and
> basically is a relay and an ICE plus DTLS-SRTP termination point (I
> assume EKT to avoid having to do re-encryption), there is a need to be
> able to handle sources from different participants. This puts extra
> requirements on how to successfully negotiate the parameters. For
> example changing the values for one media source should not force one to
> renegotiate with everyone.
>
> 2) The non-centralized multiparty use case appears to equally or more
> stress the need for having timely dynamic control. If each sender has a
> number of peer-connections to its peers, it may use local audio levels
> to determine if its media stream is to be sent or not. Thus the amount
> of bit-rate and screen estate needed to display will rapidly change as
> different users speaks. Thus the need for minimal delay when changing
> preferences are important.
>
> 3) We also have speech and audio including audio-only use cases. For
> audio there could also exist desire to request or indicate changes in
> the audio bandwidth required, or usage of multi-channel.
>
> 4) Adaptation to legacy node from a central node in a multi-party
> conference. In some use cases legacy nodes might have special needs that
> are within the profiles a WebRTC end-point is capable of producing. Thus
> the central node might request the nodes to constrain themselves to
> particular payload types, audio bandwidth etc to meet a joining session
> participant.
>
> 5) There appear to exist a need for expressing dynamic requests for
> target bit-rate as one parameter. This can be supported by TMMBR
> (RFC5104) but there exist additional transport related parameters could
> help with the adaptation. These include MTU, limits on packet rate, and
> amount of aggregation of audio frames in the payload.
>
> Overview
> --------
>
> The basic idea in this proposal is to use JSEP to establish the outer
> limits for behavior and then use Codec Operation Point (COP) proposal as
> detailed in draft-westerlund-avtext-codec-operation-point to handle
> dynamic changes during the session.
>
> So highest resolution, frame-rate and maximum complexity are expressed
> in JSEP SDP. Complexity is in several video codecs expressed by profile
> and level. I know that VP8 currently doesn't have this but it is under
> discussion when it comes to these parameters.
>
> During the session the browser implementation detects when there is need
> to use COP to do any of the following things.
>
> A) Request new target values for codec operation, for example due to
> that the GUI element displaying a video has changed due to window resize.
>
> B) Indicate when the end-point in its role as sender change parameters.
>
> In addition to just spatial resolution and video frame rate, I propose
> that the following parameters are considered as parameters that could be
> dynamically possible to indicate and request.
>
> Spatial resolution (as x and y resolution), Frame-rate, Picture Aspect
> ratio, Sample Aspect Ratio, Payload Type, Bit-rate, Token Bucket Size
> (To control burstiness of sender), Channels, Sampling Rate, Maximum RTP
> Packet Size, Maximum RTP Packet Rate, and Application Data Unit
> Aggregation (to control amount of audio frames in the same RTP packet).
>
>
> Differences
> -----------
>
> A) Using COP and using SDP based signaling for the dynamic changes are
> two quite different models in relation to how the interaction happens.
>
> For COP this all happens in the browser, normally initiated by the
> browser's own determination that a COP request or notification is
> needed. Harald's proposal appears to require that the JS initiate a
> renegotiation. This puts a requirement on the implementation to listen
> to the correct callbacks to know when changes happens, such as window
> resize. To my knowledge there are not yet any proposals for how the
> browser can initiate a JSEP renegotiation.
>
> Thus COP has the advantage that there is no API changes to get browser
> triggered parameter changes. W3C can select too but are not required to
> add API methods to allow JS to make codec parameter requests.
>
> The next thing is that COP does not require the JS application to have
> code to detect and handle re-negotiation. This makes it simpler for the
> basic application to get good behavior and they are not interrupted nor
> do they need to handle JSEP&  Offer/Answer state machine lock-out
> effects due to dynamic changes.
>
> How big impact these API issues have are unclear as W3C currently appear
> not to have included any discussion of how the browser can initiate a
> offer/answer exchange towards the JS when it determines a need to change
> parameters.
>
> But I am worried that using SDP and with an API that requires the
> application to listen for triggers that could benefit from a codec
> parameter renegotiation. This will likely only result in good behavior
> for the JS application implementors that are really good and find out
> what listeners and what signaling tricks are needed with JSEP to get
> good performance. I would much rather prefer good behavior by default in
> simple applications, i.e. using the default behavior that the browser
> implementor have put in.
>
> B) Using the media plane, i.e. RTCP for this signaling lets it in most
> case go directly between the encoding and the decoding entity in the
> code. There is no need to involve the JS nor the signaling server. One
> issue of using JSEP and SDP is that the state machine lock-out effects
> that can occur if one has sent an Offer. Then that browser may not be
> able to send a new updated Offer reflecting the latest change until the
> answer has been properly processed. COP doesn't have these limitations.
> It can send a new parameter request immediately, only limited by RTCP
> bandwidth restrictions. Using the media plane in my view guarantees that
> COP is never worse than what the signaling plane can perform at its best.
>
> C) As the general restrictions are determined in the initial
> negotiation, COP doesn't have the issue that in-flight media streams can
> become out of bounds. Thus no need for a two phase change of signaling
> parameters.
>
> D) Relying on draft-lennox-mmusic-sdp-source-selection-04 has several
> additional implications that should be discussed separately. The draft
> currently includes the following functionalities.
>
>    D1) It contains a media stream pause proposal. This makes it subject
> to the architectural debate currently ongoing in dispatch around
> draft-westerlund-avtext-rtp-stream-pause-00 which is a competing
> proposal for the same functionality.
>
>    D2) The inclusion of max desired frame rate in a SSRC specific way
>
>    D3) Extending the image attribute to be per SSRC specific expression
> of desired resolutions.
>
>    D4) Expressing relative priority in receiving different SSRCs.
>
>    D5) Providing an "information" on a sent SSRC
>
>    D6) Indication if the media sender is actively sending media using the
> given SSRC.
>
> Is it a correct observation that only D2 and D3 are required for the
> functionality of Resolution negotiation?
>
> E) The standardization situation is similar for both proposals. They are
> both relying on Internet drafts that are currently individual
> submissions. Both are partially caught in the architectural discussion
> which was held in Paris in the DISPATCH WG around Media pause/resume
> (draft-westerlund-avtext-rtp-stream-pause-00) and Media Stream Selection
> (draft-westerlund-dispatch-stream-selection-00) on what the most
> appropriate level of discussion are. This discussion will continue on
> the RAI-Area mailing list.
>
> F) As seen by the discussion on the mailing list the imageattr
> definitions may not be a 100% match to what is desired with Harald's
> proposal. I believe that COP's are more appropriate especially the
> "target" values possibility. In addition these are still open for
> adjustment and if they don't match WebRTC's requirements.
>
> I would also like to point out that I believe this functionality is also
> highly desirable for CLUE and that their requirements should be taken
> into account. I do think that this is one of the aspects where having
> matching functionality will make it much easier to have WebRTC to CLUE
> interworking.
>
> Thanks for reading all the way here!
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> 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] Alternative Proposal for Dynamic Codec Parameter Change (Was: Re: Resolution negotiation - a contribution)
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Harald Alvestrand wrote:
> My big worry is the IPR issue - I think that negotiation of video size
> during a call is likely to be "MUST implement", and I don't want to
> require implementation of a protocol where people have filed non-RF IPR
> claims against it if there are viable alternatives.

Particularly when those claims come from the author's employer.

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Magnus' proposal for only using SDP to set the limits of behavior makes
sense to me.  Clearly SDP is not potentially viable for rapid adjustment.

The question is really about what information it is useful
for a receiver or sender to provide.  For example=2C I would question=20
whether an SVC sender necessarily needs to send a message saying=20
it is adjusting its sending rate=2C assuming that the new rate is within=20
the range negotiated. =20

In any case=2C this really is not an RTCWEB-specific problem=2C and comimg
up with different solutions in different IETF WGs seems highly undesirable.




> > This is posted as individual. When it comes to this topic I will not
> > engage in any chair activities because I do have an alternative proposa=
l
> > based on
> > https://datatracker.ietf.org/doc/draft-westerlund-avtext-codec-operatio=
n-point/.
> >
> >
> > My proposal has some significant differences in how it works compared t=
o
> > Harald's. I will start with a discussion of requirements=2C in addition=
 to
> > Harald's=2C then an overview of my proposal=2C and ending with a discus=
sion
> > of the differences between the proposals. This is quite long but I do
> > hope you will read it to the end.
> >
> > Requirements
> > ------------
> >
> > Let's start with the requirements that Harald have written. I think the=
y
> > should be formulated differently. First of all I think the requirements
> > are negotiated=2C indicated etc in the context of a peer connection.
> >
> > Secondly=2C when it comes to maximum resolution and maximum frame-rate=
=2C
> > there are actually three different limits that I think are important=3B
> > Highest spatial resolution=2C highest frame-rate=2C and maximum complex=
ity
> > for a given video stream. The maximum complexity is often expressed as
> > the number of macroblocks a video codec can process per second. This is
> > a well-established complexity measure=2C used as part of standardized
> > video codec's "level" definitions since the introduction of H.263 Annex
> > X in 2004. As this is basically a joint maximum requirement on the
> > maximum amount of pixels per frame times the maximum frame-rate=2C ther=
e
> > exist cases where this complexity figure actually is the constraining
> > factor forcing a sender or receiver to either request higher frame rate
> > and lower resolution or higher resolution and a lower frame rate.
> >
> > The requirements should also be clearer in that one needs to handle
> > multiple SSRCs per RTP session=2C including multiple cameras or audio
> > streams (microphones) from a single end-point=2C where each stream can =
be
> > encoded using different parameter values.
> >
> > I also think it is important that we consider what additional encoding
> > property parameters that would make sense for WebRTC to have a
> > possibility to dynamically negotiate=2C request and indicate.
> >
> > Some additional use cases that should be considered in the requirements=
:
> >
> > 1) We have use cases including multi-party communication. One way to
> > realize these is to use a central node and I believe everyone agrees
> > that we should have good support for getting this usage to work well.
> > Thus in a basic star topology of different participants there is going
> > to be different path characteristics between the central node and each
> > participant.
> >
> > 1A) This makes it necessary to consider how one can ensure that the
> > central node can deliver appropriate rates. One way is to de-couple the
> > links by having the central node perform individual transcodings to
> > every participant. A simpler non-transcoding central node=2C only
> > forwarding streams between end-points=2C would have to enforce the lowe=
st
> > path characteristics to all. If one don't want to transcode at the
> > central node and not use the lowest path characteristics to all=2C one
> > need to consider either simulcast or scalable video coding. Both
> > simulcast and scalable video coding result in that at least in the
> > direction from a participant to the central node one needs to use
> > multiple codec operation points. Either one per peer connection=2C whic=
h
> > is how I see simulcast being realized with today's API and
> > functionality=2C or using an encoding format supporting scalable coding
> > within a single peer connection.
> >
> > 1B) In cases where the central node has minimal functionality and
> > basically is a relay and an ICE plus DTLS-SRTP termination point (I
> > assume EKT to avoid having to do re-encryption)=2C there is a need to b=
e
> > able to handle sources from different participants. This puts extra
> > requirements on how to successfully negotiate the parameters. For
> > example changing the values for one media source should not force one t=
o
> > renegotiate with everyone.
> >
> > 2) The non-centralized multiparty use case appears to equally or more
> > stress the need for having timely dynamic control. If each sender has a
> > number of peer-connections to its peers=2C it may use local audio level=
s
> > to determine if its media stream is to be sent or not. Thus the amount
> > of bit-rate and screen estate needed to display will rapidly change as
> > different users speaks. Thus the need for minimal delay when changing
> > preferences are important.
> >
> > 3) We also have speech and audio including audio-only use cases. For
> > audio there could also exist desire to request or indicate changes in
> > the audio bandwidth required=2C or usage of multi-channel.
> >
> > 4) Adaptation to legacy node from a central node in a multi-party
> > conference. In some use cases legacy nodes might have special needs tha=
t
> > are within the profiles a WebRTC end-point is capable of producing. Thu=
s
> > the central node might request the nodes to constrain themselves to
> > particular payload types=2C audio bandwidth etc to meet a joining sessi=
on
> > participant.
> >
> > 5) There appear to exist a need for expressing dynamic requests for
> > target bit-rate as one parameter. This can be supported by TMMBR
> > (RFC5104) but there exist additional transport related parameters could
> > help with the adaptation. These include MTU=2C limits on packet rate=2C=
 and
> > amount of aggregation of audio frames in the payload.
> >
> > Overview
> > --------
> >
> > The basic idea in this proposal is to use JSEP to establish the outer
> > limits for behavior and then use Codec Operation Point (COP) proposal a=
s
> > detailed in draft-westerlund-avtext-codec-operation-point to handle
> > dynamic changes during the session.
> >
> > So highest resolution=2C frame-rate and maximum complexity are expresse=
d
> > in JSEP SDP. Complexity is in several video codecs expressed by profile
> > and level. I know that VP8 currently doesn't have this but it is under
> > discussion when it comes to these parameters.
> >
> > During the session the browser implementation detects when there is nee=
d
> > to use COP to do any of the following things.
> >
> > A) Request new target values for codec operation=2C for example due to
> > that the GUI element displaying a video has changed due to window resiz=
e.
> >
> > B) Indicate when the end-point in its role as sender change parameters.
> >
> > In addition to just spatial resolution and video frame rate=2C I propos=
e
> > that the following parameters are considered as parameters that could b=
e
> > dynamically possible to indicate and request.
> >
> > Spatial resolution (as x and y resolution)=2C Frame-rate=2C Picture Asp=
ect
> > ratio=2C Sample Aspect Ratio=2C Payload Type=2C Bit-rate=2C Token Bucke=
t Size
> > (To control burstiness of sender)=2C Channels=2C Sampling Rate=2C Maxim=
um RTP
> > Packet Size=2C Maximum RTP Packet Rate=2C and Application Data Unit
> > Aggregation (to control amount of audio frames in the same RTP packet).
> >
> >
> > Differences
> > -----------
> >
> > A) Using COP and using SDP based signaling for the dynamic changes are
> > two quite different models in relation to how the interaction happens.
> >
> > For COP this all happens in the browser=2C normally initiated by the
> > browser's own determination that a COP request or notification is
> > needed. Harald's proposal appears to require that the JS initiate a
> > renegotiation. This puts a requirement on the implementation to listen
> > to the correct callbacks to know when changes happens=2C such as window
> > resize. To my knowledge there are not yet any proposals for how the
> > browser can initiate a JSEP renegotiation.
> >
> > Thus COP has the advantage that there is no API changes to get browser
> > triggered parameter changes. W3C can select too but are not required to
> > add API methods to allow JS to make codec parameter requests.
> >
> > The next thing is that COP does not require the JS application to have
> > code to detect and handle re-negotiation. This makes it simpler for the
> > basic application to get good behavior and they are not interrupted nor
> > do they need to handle JSEP&  Offer/Answer state machine lock-out
> > effects due to dynamic changes.
> >
> > How big impact these API issues have are unclear as W3C currently appea=
r
> > not to have included any discussion of how the browser can initiate a
> > offer/answer exchange towards the JS when it determines a need to chang=
e
> > parameters.
> >
> > But I am worried that using SDP and with an API that requires the
> > application to listen for triggers that could benefit from a codec
> > parameter renegotiation. This will likely only result in good behavior
> > for the JS application implementors that are really good and find out
> > what listeners and what signaling tricks are needed with JSEP to get
> > good performance. I would much rather prefer good behavior by default i=
n
> > simple applications=2C i.e. using the default behavior that the browser
> > implementor have put in.
> >
> > B) Using the media plane=2C i.e. RTCP for this signaling lets it in mos=
t
> > case go directly between the encoding and the decoding entity in the
> > code. There is no need to involve the JS nor the signaling server. One
> > issue of using JSEP and SDP is that the state machine lock-out effects
> > that can occur if one has sent an Offer. Then that browser may not be
> > able to send a new updated Offer reflecting the latest change until the
> > answer has been properly processed. COP doesn't have these limitations.
> > It can send a new parameter request immediately=2C only limited by RTCP
> > bandwidth restrictions. Using the media plane in my view guarantees tha=
t
> > COP is never worse than what the signaling plane can perform at its bes=
t.
> >
> > C) As the general restrictions are determined in the initial
> > negotiation=2C COP doesn't have the issue that in-flight media streams =
can
> > become out of bounds. Thus no need for a two phase change of signaling
> > parameters.
> >
> > D) Relying on draft-lennox-mmusic-sdp-source-selection-04 has several
> > additional implications that should be discussed separately. The draft
> > currently includes the following functionalities.
> >
> >    D1) It contains a media stream pause proposal. This makes it subject
> > to the architectural debate currently ongoing in dispatch around
> > draft-westerlund-avtext-rtp-stream-pause-00 which is a competing
> > proposal for the same functionality.
> >
> >    D2) The inclusion of max desired frame rate in a SSRC specific way
> >
> >    D3) Extending the image attribute to be per SSRC specific expression
> > of desired resolutions.
> >
> >    D4) Expressing relative priority in receiving different SSRCs.
> >
> >    D5) Providing an "information" on a sent SSRC
> >
> >    D6) Indication if the media sender is actively sending media using t=
he
> > given SSRC.
> >
> > Is it a correct observation that only D2 and D3 are required for the
> > functionality of Resolution negotiation?
> >
> > E) The standardization situation is similar for both proposals. They ar=
e
> > both relying on Internet drafts that are currently individual
> > submissions. Both are partially caught in the architectural discussion
> > which was held in Paris in the DISPATCH WG around Media pause/resume
> > (draft-westerlund-avtext-rtp-stream-pause-00) and Media Stream Selectio=
n
> > (draft-westerlund-dispatch-stream-selection-00) on what the most
> > appropriate level of discussion are. This discussion will continue on
> > the RAI-Area mailing list.
> >
> > F) As seen by the discussion on the mailing list the imageattr
> > definitions may not be a 100% match to what is desired with Harald's
> > proposal. I believe that COP's are more appropriate especially the
> > "target" values possibility. In addition these are still open for
> > adjustment and if they don't match WebRTC's requirements.
> >
> > I would also like to point out that I believe this functionality is als=
o
> > highly desirable for CLUE and that their requirements should be taken
> > into account. I do think that this is one of the aspects where having
> > matching functionality will make it much easier to have WebRTC to CLUE
> > interworking.
> >
> > Thanks for reading all the way here!
> >
> > Cheers
> >
> > Magnus Westerlund
> >
> > ----------------------------------------------------------------------
> > Multimedia Technologies=2C Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm=2C Sweden| mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> >
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Magnus' proposal for only using SDP to set the limits of behavior makes<br>=
sense to me.&nbsp=3B Clearly SDP is not potentially viable for rapid adjust=
ment.<br><br>The question is really about what information it is useful<br>=
for a receiver or sender to provide.&nbsp=3B For example=2C I would questio=
n <br>whether an SVC sender necessarily needs to send a message saying <br>=
it is adjusting its sending rate=2C assuming that the new rate is within <b=
r>the range negotiated.&nbsp=3B <br><br>In any case=2C this really is not a=
n RTCWEB-specific problem=2C and comimg<br>up with different solutions in d=
ifferent IETF WGs seems highly undesirable.<br><br><br><br><br><div>&gt=3B =
&gt=3B This is posted as individual. When it comes to this topic I will not=
<br>&gt=3B &gt=3B engage in any chair activities because I do have an alter=
native proposal<br>&gt=3B &gt=3B based on<br>&gt=3B &gt=3B https://datatrac=
ker.ietf.org/doc/draft-westerlund-avtext-codec-operation-point/.<br>&gt=3B =
&gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B My proposal has some significant d=
ifferences in how it works compared to<br>&gt=3B &gt=3B Harald's. I will st=
art with a discussion of requirements=2C in addition to<br>&gt=3B &gt=3B Ha=
rald's=2C then an overview of my proposal=2C and ending with a discussion<b=
r>&gt=3B &gt=3B of the differences between the proposals. This is quite lon=
g but I do<br>&gt=3B &gt=3B hope you will read it to the end.<br>&gt=3B &gt=
=3B<br>&gt=3B &gt=3B Requirements<br>&gt=3B &gt=3B ------------<br>&gt=3B &=
gt=3B<br>&gt=3B &gt=3B Let's start with the requirements that Harald have w=
ritten. I think they<br>&gt=3B &gt=3B should be formulated differently. Fir=
st of all I think the requirements<br>&gt=3B &gt=3B are negotiated=2C indic=
ated etc in the context of a peer connection.<br>&gt=3B &gt=3B<br>&gt=3B &g=
t=3B Secondly=2C when it comes to maximum resolution and maximum frame-rate=
=2C<br>&gt=3B &gt=3B there are actually three different limits that I think=
 are important=3B<br>&gt=3B &gt=3B Highest spatial resolution=2C highest fr=
ame-rate=2C and maximum complexity<br>&gt=3B &gt=3B for a given video strea=
m. The maximum complexity is often expressed as<br>&gt=3B &gt=3B the number=
 of macroblocks a video codec can process per second. This is<br>&gt=3B &gt=
=3B a well-established complexity measure=2C used as part of standardized<b=
r>&gt=3B &gt=3B video codec's "level" definitions since the introduction of=
 H.263 Annex<br>&gt=3B &gt=3B X in 2004. As this is basically a joint maxim=
um requirement on the<br>&gt=3B &gt=3B maximum amount of pixels per frame t=
imes the maximum frame-rate=2C there<br>&gt=3B &gt=3B exist cases where thi=
s complexity figure actually is the constraining<br>&gt=3B &gt=3B factor fo=
rcing a sender or receiver to either request higher frame rate<br>&gt=3B &g=
t=3B and lower resolution or higher resolution and a lower frame rate.<br>&=
gt=3B &gt=3B<br>&gt=3B &gt=3B The requirements should also be clearer in th=
at one needs to handle<br>&gt=3B &gt=3B multiple SSRCs per RTP session=2C i=
ncluding multiple cameras or audio<br>&gt=3B &gt=3B streams (microphones) f=
rom a single end-point=2C where each stream can be<br>&gt=3B &gt=3B encoded=
 using different parameter values.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B I also=
 think it is important that we consider what additional encoding<br>&gt=3B =
&gt=3B property parameters that would make sense for WebRTC to have a<br>&g=
t=3B &gt=3B possibility to dynamically negotiate=2C request and indicate.<b=
r>&gt=3B &gt=3B<br>&gt=3B &gt=3B Some additional use cases that should be c=
onsidered in the requirements:<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B 1) We have=
 use cases including multi-party communication. One way to<br>&gt=3B &gt=3B=
 realize these is to use a central node and I believe everyone agrees<br>&g=
t=3B &gt=3B that we should have good support for getting this usage to work=
 well.<br>&gt=3B &gt=3B Thus in a basic star topology of different particip=
ants there is going<br>&gt=3B &gt=3B to be different path characteristics b=
etween the central node and each<br>&gt=3B &gt=3B participant.<br>&gt=3B &g=
t=3B<br>&gt=3B &gt=3B 1A) This makes it necessary to consider how one can e=
nsure that the<br>&gt=3B &gt=3B central node can deliver appropriate rates.=
 One way is to de-couple the<br>&gt=3B &gt=3B links by having the central n=
ode perform individual transcodings to<br>&gt=3B &gt=3B every participant. =
A simpler non-transcoding central node=2C only<br>&gt=3B &gt=3B forwarding =
streams between end-points=2C would have to enforce the lowest<br>&gt=3B &g=
t=3B path characteristics to all. If one don't want to transcode at the<br>=
&gt=3B &gt=3B central node and not use the lowest path characteristics to a=
ll=2C one<br>&gt=3B &gt=3B need to consider either simulcast or scalable vi=
deo coding. Both<br>&gt=3B &gt=3B simulcast and scalable video coding resul=
t in that at least in the<br>&gt=3B &gt=3B direction from a participant to =
the central node one needs to use<br>&gt=3B &gt=3B multiple codec operation=
 points. Either one per peer connection=2C which<br>&gt=3B &gt=3B is how I =
see simulcast being realized with today's API and<br>&gt=3B &gt=3B function=
ality=2C or using an encoding format supporting scalable coding<br>&gt=3B &=
gt=3B within a single peer connection.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B 1B=
) In cases where the central node has minimal functionality and<br>&gt=3B &=
gt=3B basically is a relay and an ICE plus DTLS-SRTP termination point (I<b=
r>&gt=3B &gt=3B assume EKT to avoid having to do re-encryption)=2C there is=
 a need to be<br>&gt=3B &gt=3B able to handle sources from different partic=
ipants. This puts extra<br>&gt=3B &gt=3B requirements on how to successfull=
y negotiate the parameters. For<br>&gt=3B &gt=3B example changing the value=
s for one media source should not force one to<br>&gt=3B &gt=3B renegotiate=
 with everyone.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B 2) The non-centralized mu=
ltiparty use case appears to equally or more<br>&gt=3B &gt=3B stress the ne=
ed for having timely dynamic control. If each sender has a<br>&gt=3B &gt=3B=
 number of peer-connections to its peers=2C it may use local audio levels<b=
r>&gt=3B &gt=3B to determine if its media stream is to be sent or not. Thus=
 the amount<br>&gt=3B &gt=3B of bit-rate and screen estate needed to displa=
y will rapidly change as<br>&gt=3B &gt=3B different users speaks. Thus the =
need for minimal delay when changing<br>&gt=3B &gt=3B preferences are impor=
tant.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B 3) We also have speech and audio in=
cluding audio-only use cases. For<br>&gt=3B &gt=3B audio there could also e=
xist desire to request or indicate changes in<br>&gt=3B &gt=3B the audio ba=
ndwidth required=2C or usage of multi-channel.<br>&gt=3B &gt=3B<br>&gt=3B &=
gt=3B 4) Adaptation to legacy node from a central node in a multi-party<br>=
&gt=3B &gt=3B conference. In some use cases legacy nodes might have special=
 needs that<br>&gt=3B &gt=3B are within the profiles a WebRTC end-point is =
capable of producing. Thus<br>&gt=3B &gt=3B the central node might request =
the nodes to constrain themselves to<br>&gt=3B &gt=3B particular payload ty=
pes=2C audio bandwidth etc to meet a joining session<br>&gt=3B &gt=3B parti=
cipant.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B 5) There appear to exist a need f=
or expressing dynamic requests for<br>&gt=3B &gt=3B target bit-rate as one =
parameter. This can be supported by TMMBR<br>&gt=3B &gt=3B (RFC5104) but th=
ere exist additional transport related parameters could<br>&gt=3B &gt=3B he=
lp with the adaptation. These include MTU=2C limits on packet rate=2C and<b=
r>&gt=3B &gt=3B amount of aggregation of audio frames in the payload.<br>&g=
t=3B &gt=3B<br>&gt=3B &gt=3B Overview<br>&gt=3B &gt=3B --------<br>&gt=3B &=
gt=3B<br>&gt=3B &gt=3B The basic idea in this proposal is to use JSEP to es=
tablish the outer<br>&gt=3B &gt=3B limits for behavior and then use Codec O=
peration Point (COP) proposal as<br>&gt=3B &gt=3B detailed in draft-westerl=
und-avtext-codec-operation-point to handle<br>&gt=3B &gt=3B dynamic changes=
 during the session.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B So highest resolutio=
n=2C frame-rate and maximum complexity are expressed<br>&gt=3B &gt=3B in JS=
EP SDP. Complexity is in several video codecs expressed by profile<br>&gt=
=3B &gt=3B and level. I know that VP8 currently doesn't have this but it is=
 under<br>&gt=3B &gt=3B discussion when it comes to these parameters.<br>&g=
t=3B &gt=3B<br>&gt=3B &gt=3B During the session the browser implementation =
detects when there is need<br>&gt=3B &gt=3B to use COP to do any of the fol=
lowing things.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B A) Request new target valu=
es for codec operation=2C for example due to<br>&gt=3B &gt=3B that the GUI =
element displaying a video has changed due to window resize.<br>&gt=3B &gt=
=3B<br>&gt=3B &gt=3B B) Indicate when the end-point in its role as sender c=
hange parameters.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B In addition to just spa=
tial resolution and video frame rate=2C I propose<br>&gt=3B &gt=3B that the=
 following parameters are considered as parameters that could be<br>&gt=3B =
&gt=3B dynamically possible to indicate and request.<br>&gt=3B &gt=3B<br>&g=
t=3B &gt=3B Spatial resolution (as x and y resolution)=2C Frame-rate=2C Pic=
ture Aspect<br>&gt=3B &gt=3B ratio=2C Sample Aspect Ratio=2C Payload Type=
=2C Bit-rate=2C Token Bucket Size<br>&gt=3B &gt=3B (To control burstiness o=
f sender)=2C Channels=2C Sampling Rate=2C Maximum RTP<br>&gt=3B &gt=3B Pack=
et Size=2C Maximum RTP Packet Rate=2C and Application Data Unit<br>&gt=3B &=
gt=3B Aggregation (to control amount of audio frames in the same RTP packet=
).<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B Differences<br>&gt=3B=
 &gt=3B -----------<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B A) Using COP and usin=
g SDP based signaling for the dynamic changes are<br>&gt=3B &gt=3B two quit=
e different models in relation to how the interaction happens.<br>&gt=3B &g=
t=3B<br>&gt=3B &gt=3B For COP this all happens in the browser=2C normally i=
nitiated by the<br>&gt=3B &gt=3B browser's own determination that a COP req=
uest or notification is<br>&gt=3B &gt=3B needed. Harald's proposal appears =
to require that the JS initiate a<br>&gt=3B &gt=3B renegotiation. This puts=
 a requirement on the implementation to listen<br>&gt=3B &gt=3B to the corr=
ect callbacks to know when changes happens=2C such as window<br>&gt=3B &gt=
=3B resize. To my knowledge there are not yet any proposals for how the<br>=
&gt=3B &gt=3B browser can initiate a JSEP renegotiation.<br>&gt=3B &gt=3B<b=
r>&gt=3B &gt=3B Thus COP has the advantage that there is no API changes to =
get browser<br>&gt=3B &gt=3B triggered parameter changes. W3C can select to=
o but are not required to<br>&gt=3B &gt=3B add API methods to allow JS to m=
ake codec parameter requests.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B The next th=
ing is that COP does not require the JS application to have<br>&gt=3B &gt=
=3B code to detect and handle re-negotiation. This makes it simpler for the=
<br>&gt=3B &gt=3B basic application to get good behavior and they are not i=
nterrupted nor<br>&gt=3B &gt=3B do they need to handle JSEP&amp=3B  Offer/A=
nswer state machine lock-out<br>&gt=3B &gt=3B effects due to dynamic change=
s.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B How big impact these API issues have a=
re unclear as W3C currently appear<br>&gt=3B &gt=3B not to have included an=
y discussion of how the browser can initiate a<br>&gt=3B &gt=3B offer/answe=
r exchange towards the JS when it determines a need to change<br>&gt=3B &gt=
=3B parameters.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B But I am worried that usi=
ng SDP and with an API that requires the<br>&gt=3B &gt=3B application to li=
sten for triggers that could benefit from a codec<br>&gt=3B &gt=3B paramete=
r renegotiation. This will likely only result in good behavior<br>&gt=3B &g=
t=3B for the JS application implementors that are really good and find out<=
br>&gt=3B &gt=3B what listeners and what signaling tricks are needed with J=
SEP to get<br>&gt=3B &gt=3B good performance. I would much rather prefer go=
od behavior by default in<br>&gt=3B &gt=3B simple applications=2C i.e. usin=
g the default behavior that the browser<br>&gt=3B &gt=3B implementor have p=
ut in.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B B) Using the media plane=2C i.e. R=
TCP for this signaling lets it in most<br>&gt=3B &gt=3B case go directly be=
tween the encoding and the decoding entity in the<br>&gt=3B &gt=3B code. Th=
ere is no need to involve the JS nor the signaling server. One<br>&gt=3B &g=
t=3B issue of using JSEP and SDP is that the state machine lock-out effects=
<br>&gt=3B &gt=3B that can occur if one has sent an Offer. Then that browse=
r may not be<br>&gt=3B &gt=3B able to send a new updated Offer reflecting t=
he latest change until the<br>&gt=3B &gt=3B answer has been properly proces=
sed. COP doesn't have these limitations.<br>&gt=3B &gt=3B It can send a new=
 parameter request immediately=2C only limited by RTCP<br>&gt=3B &gt=3B ban=
dwidth restrictions. Using the media plane in my view guarantees that<br>&g=
t=3B &gt=3B COP is never worse than what the signaling plane can perform at=
 its best.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B C) As the general restrictions=
 are determined in the initial<br>&gt=3B &gt=3B negotiation=2C COP doesn't =
have the issue that in-flight media streams can<br>&gt=3B &gt=3B become out=
 of bounds. Thus no need for a two phase change of signaling<br>&gt=3B &gt=
=3B parameters.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B D) Relying on draft-lenno=
x-mmusic-sdp-source-selection-04 has several<br>&gt=3B &gt=3B additional im=
plications that should be discussed separately. The draft<br>&gt=3B &gt=3B =
currently includes the following functionalities.<br>&gt=3B &gt=3B<br>&gt=
=3B &gt=3B    D1) It contains a media stream pause proposal. This makes it =
subject<br>&gt=3B &gt=3B to the architectural debate currently ongoing in d=
ispatch around<br>&gt=3B &gt=3B draft-westerlund-avtext-rtp-stream-pause-00=
 which is a competing<br>&gt=3B &gt=3B proposal for the same functionality.=
<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B    D2) The inclusion of max desired fram=
e rate in a SSRC specific way<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B    D3) Exte=
nding the image attribute to be per SSRC specific expression<br>&gt=3B &gt=
=3B of desired resolutions.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B    D4) Expres=
sing relative priority in receiving different SSRCs.<br>&gt=3B &gt=3B<br>&g=
t=3B &gt=3B    D5) Providing an "information" on a sent SSRC<br>&gt=3B &gt=
=3B<br>&gt=3B &gt=3B    D6) Indication if the media sender is actively send=
ing media using the<br>&gt=3B &gt=3B given SSRC.<br>&gt=3B &gt=3B<br>&gt=3B=
 &gt=3B Is it a correct observation that only D2 and D3 are required for th=
e<br>&gt=3B &gt=3B functionality of Resolution negotiation?<br>&gt=3B &gt=
=3B<br>&gt=3B &gt=3B E) The standardization situation is similar for both p=
roposals. They are<br>&gt=3B &gt=3B both relying on Internet drafts that ar=
e currently individual<br>&gt=3B &gt=3B submissions. Both are partially cau=
ght in the architectural discussion<br>&gt=3B &gt=3B which was held in Pari=
s in the DISPATCH WG around Media pause/resume<br>&gt=3B &gt=3B (draft-west=
erlund-avtext-rtp-stream-pause-00) and Media Stream Selection<br>&gt=3B &gt=
=3B (draft-westerlund-dispatch-stream-selection-00) on what the most<br>&gt=
=3B &gt=3B appropriate level of discussion are. This discussion will contin=
ue on<br>&gt=3B &gt=3B the RAI-Area mailing list.<br>&gt=3B &gt=3B<br>&gt=
=3B &gt=3B F) As seen by the discussion on the mailing list the imageattr<b=
r>&gt=3B &gt=3B definitions may not be a 100% match to what is desired with=
 Harald's<br>&gt=3B &gt=3B proposal. I believe that COP's are more appropri=
ate especially the<br>&gt=3B &gt=3B "target" values possibility. In additio=
n these are still open for<br>&gt=3B &gt=3B adjustment and if they don't ma=
tch WebRTC's requirements.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B I would also l=
ike to point out that I believe this functionality is also<br>&gt=3B &gt=3B=
 highly desirable for CLUE and that their requirements should be taken<br>&=
gt=3B &gt=3B into account. I do think that this is one of the aspects where=
 having<br>&gt=3B &gt=3B matching functionality will make it much easier to=
 have WebRTC to CLUE<br>&gt=3B &gt=3B interworking.<br>&gt=3B &gt=3B<br>&gt=
=3B &gt=3B Thanks for reading all the way here!<br>&gt=3B &gt=3B<br>&gt=3B =
&gt=3B Cheers<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B Magnus Westerlund<br>&gt=3B=
 &gt=3B<br>&gt=3B &gt=3B --------------------------------------------------=
--------------------<br>&gt=3B &gt=3B Multimedia Technologies=2C Ericsson R=
esearch EAB/TVM<br>&gt=3B &gt=3B ------------------------------------------=
----------------------------<br>&gt=3B &gt=3B Ericsson AB                | =
Phone  +46 10 7148287<br>&gt=3B &gt=3B F=E4r=F6gatan 6                | Mob=
ile +46 73 0949079<br>&gt=3B &gt=3B SE-164 80 Stockholm=2C Sweden| mailto: =
magnus.westerlund@ericsson.com<br>&gt=3B &gt=3B ---------------------------=
-------------------------------------------<br>&gt=3B &gt=3B<br>&gt=3B &gt=
=3B<br>&gt=3B &gt=3B<br>&gt=3B <br>&gt=3B _________________________________=
______________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb@ietf.org<br>&=
gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br></div> 		 	   		  </d=
iv></body>
</html>=

--_788a7258-0d1c-4de8-89f5-7844106da596_--

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Subject: Re: [rtcweb] Alternative Proposal for Dynamic Codec Parameter Change (Was: Re: Resolution negotiation - a contribution)
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On 04/24/2012 08:13 PM, Bernard Aboba wrote:
> Magnus' proposal for only using SDP to set the limits of behavior makes
> sense to me.  Clearly SDP is not potentially viable for rapid adjustment.
I would make that "potentially not viable" - if the RTT between clients 
and server is very low, and the server processing time is very short, 
there can be scenarios where sub-500-ms renegotiates are reasonable to 
expect even with server-mediated renegotiation, and there's always the 
potential of SDP exchange over a data channel, once that's commonly 
implemented. But it will be trivially easy for something to get 
implemented in a nonoptimal fashion, and for this to take quite a bit 
longer.
>
> The question is really about what information it is useful
> for a receiver or sender to provide.  For example, I would question
> whether an SVC sender necessarily needs to send a message saying
> it is adjusting its sending rate, assuming that the new rate is within
> the range negotiated.
>
> In any case, this really is not an RTCWEB-specific problem, and comimg
> up with different solutions in different IETF WGs seems highly 
> undesirable.
I would love to not invent mechanisms. All the mechanisms so far 
suggested have been previously proposed elsewhere, but only the 
m-line-level a=imageattr spec (RFC 6236) has been published as a 
standars-track RFC.


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    On 04/24/2012 08:13 PM, Bernard Aboba wrote:
    <blockquote cite="mid:BLU169-W83F897D215C15DC22F510E93260@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">
        Magnus' proposal for only using SDP to set the limits of
        behavior makes<br>
        sense to me.&nbsp; Clearly SDP is not potentially viable for rapid
        adjustment.<br>
      </div>
    </blockquote>
    I would make that "potentially not viable" - if the RTT between
    clients and server is very low, and the server processing time is
    very short, there can be scenarios where sub-500-ms renegotiates are
    reasonable to expect even with server-mediated renegotiation, and
    there's always the potential of SDP exchange over a data channel,
    once that's commonly implemented. But it will be trivially easy for
    something to get implemented in a nonoptimal fashion, and for this
    to take quite a bit longer.<br>
    <blockquote cite="mid:BLU169-W83F897D215C15DC22F510E93260@phx.gbl"
      type="cite">
      <div dir="ltr"><br>
        The question is really about what information it is useful<br>
        for a receiver or sender to provide.&nbsp; For example, I would
        question <br>
        whether an SVC sender necessarily needs to send a message saying
        <br>
        it is adjusting its sending rate, assuming that the new rate is
        within <br>
        the range negotiated.&nbsp; <br>
        <br>
        In any case, this really is not an RTCWEB-specific problem, and
        comimg<br>
        up with different solutions in different IETF WGs seems highly
        undesirable.<br>
      </div>
    </blockquote>
    I would love to not invent mechanisms. All the mechanisms so far
    suggested have been previously proposed elsewhere, but only the
    m-line-level a=imageattr spec (RFC 6236) has been published as a
    standars-track RFC.<br>
    <br>
  </body>
</html>

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Subject: Re: [rtcweb] Alternative Proposal for Dynamic Codec Parameter Change (Was: Re: Resolution negotiation - a contribution)
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On Tue, Apr 24, 2012 at 10:25 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

>
> Harald, WG,
>
> This is posted as individual. When it comes to this topic I will not
> engage in any chair activities because I do have an alternative proposal
> based on
>
> https://datatracker.ietf.org/doc/draft-westerlund-avtext-codec-operation-=
point/
> .
>
>
> My proposal has some significant differences in how it works compared to
> Harald's. I will start with a discussion of requirements, in addition to
> Harald's, then an overview of my proposal, and ending with a discussion
> of the differences between the proposals. This is quite long but I do
> hope you will read it to the end.
>
> Requirements
> ------------
>
> Let's start with the requirements that Harald have written. I think they
> should be formulated differently. First of all I think the requirements
> are negotiated, indicated etc in the context of a peer connection.
>
> Secondly, when it comes to maximum resolution and maximum frame-rate,
> there are actually three different limits that I think are important;
> Highest spatial resolution, highest frame-rate, and maximum complexity
> for a given video stream. The maximum complexity is often expressed as
> the number of macroblocks a video codec can process per second. This is
> a well-established complexity measure, used as part of standardized
> video codec's "level" definitions since the introduction of H.263 Annex
> X in 2004. As this is basically a joint maximum requirement on the
> maximum amount of pixels per frame times the maximum frame-rate, there
> exist cases where this complexity figure actually is the constraining
> factor forcing a sender or receiver to either request higher frame rate
> and lower resolution or higher resolution and a lower frame rate.
>
> The requirements should also be clearer in that one needs to handle
> multiple SSRCs per RTP session, including multiple cameras or audio
> streams (microphones) from a single end-point, where each stream can be
> encoded using different parameter values.
>
> I also think it is important that we consider what additional encoding
> property parameters that would make sense for WebRTC to have a
> possibility to dynamically negotiate, request and indicate.
>
> Some additional use cases that should be considered in the requirements:
>
> 1) We have use cases including multi-party communication. One way to
> realize these is to use a central node and I believe everyone agrees
> that we should have good support for getting this usage to work well.
> Thus in a basic star topology of different participants there is going
> to be different path characteristics between the central node and each
> participant.
>
> 1A) This makes it necessary to consider how one can ensure that the
> central node can deliver appropriate rates. One way is to de-couple the
> links by having the central node perform individual transcodings to
> every participant. A simpler non-transcoding central node, only
> forwarding streams between end-points, would have to enforce the lowest
> path characteristics to all. If one don't want to transcode at the
> central node and not use the lowest path characteristics to all, one
> need to consider either simulcast or scalable video coding. Both
> simulcast and scalable video coding result in that at least in the
> direction from a participant to the central node one needs to use
> multiple codec operation points. Either one per peer connection, which
> is how I see simulcast being realized with today's API and
> functionality, or using an encoding format supporting scalable coding
> within a single peer connection.
>
> 1B) In cases where the central node has minimal functionality and
> basically is a relay and an ICE plus DTLS-SRTP termination point (I
> assume EKT to avoid having to do re-encryption), there is a need to be
> able to handle sources from different participants. This puts extra
> requirements on how to successfully negotiate the parameters. For
> example changing the values for one media source should not force one to
> renegotiate with everyone.
>
> 2) The non-centralized multiparty use case appears to equally or more
> stress the need for having timely dynamic control. If each sender has a
> number of peer-connections to its peers, it may use local audio levels
> to determine if its media stream is to be sent or not. Thus the amount
> of bit-rate and screen estate needed to display will rapidly change as
> different users speaks. Thus the need for minimal delay when changing
> preferences are important.
>
> 3) We also have speech and audio including audio-only use cases. For
> audio there could also exist desire to request or indicate changes in
> the audio bandwidth required, or usage of multi-channel.
>
> 4) Adaptation to legacy node from a central node in a multi-party
> conference. In some use cases legacy nodes might have special needs that
> are within the profiles a WebRTC end-point is capable of producing. Thus
> the central node might request the nodes to constrain themselves to
> particular payload types, audio bandwidth etc to meet a joining session
> participant.
>
> 5) There appear to exist a need for expressing dynamic requests for
> target bit-rate as one parameter. This can be supported by TMMBR
> (RFC5104) but there exist additional transport related parameters could
> help with the adaptation. These include MTU, limits on packet rate, and
> amount of aggregation of audio frames in the payload.
>
> Overview
> --------
>
> The basic idea in this proposal is to use JSEP to establish the outer
> limits for behavior and then use Codec Operation Point (COP) proposal as
> detailed in draft-westerlund-avtext-codec-operation-point to handle
> dynamic changes during the session.
>
> So highest resolution, frame-rate and maximum complexity are expressed
> in JSEP SDP. Complexity is in several video codecs expressed by profile
> and level. I know that VP8 currently doesn't have this but it is under
> discussion when it comes to these parameters.
>
> During the session the browser implementation detects when there is need
> to use COP to do any of the following things.
>
> A) Request new target values for codec operation, for example due to
> that the GUI element displaying a video has changed due to window resize.
>
> B) Indicate when the end-point in its role as sender change parameters.
>
> In addition to just spatial resolution and video frame rate, I propose
> that the following parameters are considered as parameters that could be
> dynamically possible to indicate and request.
>
> Spatial resolution (as x and y resolution), Frame-rate, Picture Aspect
> ratio, Sample Aspect Ratio, Payload Type, Bit-rate, Token Bucket Size
> (To control burstiness of sender), Channels, Sampling Rate, Maximum RTP
> Packet Size, Maximum RTP Packet Rate, and Application Data Unit
> Aggregation (to control amount of audio frames in the same RTP packet).
>
>
> Differences
> -----------
>
> A) Using COP and using SDP based signaling for the dynamic changes are
> two quite different models in relation to how the interaction happens.
>
> For COP this all happens in the browser, normally initiated by the
> browser's own determination that a COP request or notification is
> needed. Harald's proposal appears to require that the JS initiate a
> renegotiation. This puts a requirement on the implementation to listen
> to the correct callbacks to know when changes happens, such as window
> resize. To my knowledge there are not yet any proposals for how the
> browser can initiate a JSEP renegotiation.
>

Maybe I misunderstand what you are saying, but the application will
definitely know when the browser size changes, and can then trigger a JSEP
renegotiation by calling createOffer and shipping it off.

I have the opposite concern - how can the browser know when the application
makes a change, for instance to display a participant in a large view
versus a small view? This may be possible if using a <video/> for display,
but I don't think it will be possible when using WebGL for rendering, where
the size of the view will be dictated only by the geometry of the scene.



> Thus COP has the advantage that there is no API changes to get browser
> triggered parameter changes. W3C can select too but are not required to
> add API methods to allow JS to make codec parameter requests.
>
> The next thing is that COP does not require the JS application to have
> code to detect and handle re-negotiation. This makes it simpler for the
> basic application to get good behavior and they are not interrupted nor
> do they need to handle JSEP & Offer/Answer state machine lock-out
> effects due to dynamic changes.
>
> How big impact these API issues have are unclear as W3C currently appear
> not to have included any discussion of how the browser can initiate a
> offer/answer exchange towards the JS when it determines a need to change
> parameters.
>
> But I am worried that using SDP and with an API that requires the
> application to listen for triggers that could benefit from a codec
> parameter renegotiation. This will likely only result in good behavior
> for the JS application implementors that are really good and find out
> what listeners and what signaling tricks are needed with JSEP to get
> good performance. I would much rather prefer good behavior by default in
> simple applications, i.e. using the default behavior that the browser
> implementor have put in.
>

The issue here is that this behavior needs to be sufficiently specified, or
else we will have many different behaviors, none of which can be fully
overridden by the app. I worry this will make life hard for people trying
to develop sophisticated apps.


> B) Using the media plane, i.e. RTCP for this signaling lets it in most
> case go directly between the encoding and the decoding entity in the
> code. There is no need to involve the JS nor the signaling server. One
> issue of using JSEP and SDP is that the state machine lock-out effects
> that can occur if one has sent an Offer. Then that browser may not be
> able to send a new updated Offer reflecting the latest change until the
> answer has been properly processed. COP doesn't have these limitations.
> It can send a new parameter request immediately, only limited by RTCP
> bandwidth restrictions. Using the media plane in my view guarantees that
> COP is never worse than what the signaling plane can perform at its best.
>
> C) As the general restrictions are determined in the initial
> negotiation, COP doesn't have the issue that in-flight media streams can
> become out of bounds. Thus no need for a two phase change of signaling
> parameters.
>
> D) Relying on draft-lennox-mmusic-sdp-source-selection-04 has several
> additional implications that should be discussed separately. The draft
> currently includes the following functionalities.
>
>  D1) It contains a media stream pause proposal. This makes it subject
> to the architectural debate currently ongoing in dispatch around
> draft-westerlund-avtext-rtp-stream-pause-00 which is a competing
> proposal for the same functionality.
>
>  D2) The inclusion of max desired frame rate in a SSRC specific way
>
>  D3) Extending the image attribute to be per SSRC specific expression
> of desired resolutions.
>
>  D4) Expressing relative priority in receiving different SSRCs.
>
>  D5) Providing an "information" on a sent SSRC
>
>  D6) Indication if the media sender is actively sending media using the
> given SSRC.
>
> Is it a correct observation that only D2 and D3 are required for the
> functionality of Resolution negotiation?
>
> E) The standardization situation is similar for both proposals. They are
> both relying on Internet drafts that are currently individual
> submissions. Both are partially caught in the architectural discussion
> which was held in Paris in the DISPATCH WG around Media pause/resume
> (draft-westerlund-avtext-rtp-stream-pause-00) and Media Stream Selection
> (draft-westerlund-dispatch-stream-selection-00) on what the most
> appropriate level of discussion are. This discussion will continue on
> the RAI-Area mailing list.
>
> F) As seen by the discussion on the mailing list the imageattr
> definitions may not be a 100% match to what is desired with Harald's
> proposal. I believe that COP's are more appropriate especially the
> "target" values possibility. In addition these are still open for
> adjustment and if they don't match WebRTC's requirements.
>
> I would also like to point out that I believe this functionality is also
> highly desirable for CLUE and that their requirements should be taken
> into account. I do think that this is one of the aspects where having
> matching functionality will make it much easier to have WebRTC to CLUE
> interworking.
>
> Thanks for reading all the way here!
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> 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
> ----------------------------------------------------------------------
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue, Apr 2=
4, 2012 at 10:25 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerlund@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"><br>
Harald, WG,<br>
<br>
This is posted as individual. When it comes to this topic I will not<br>
engage in any chair activities because I do have an alternative proposal<br=
>
based on<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-westerlund-avtext-codec-o=
peration-point/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-w=
esterlund-avtext-codec-operation-point/</a>.<br>
<br>
<br>
My proposal has some significant differences in how it works compared to<br=
>
Harald&#39;s. I will start with a discussion of requirements, in addition t=
o<br>
Harald&#39;s, then an overview of my proposal, and ending with a discussion=
<br>
of the differences between the proposals. This is quite long but I do<br>
hope you will read it to the end.<br>
<br>
Requirements<br>
------------<br>
<br>
Let&#39;s start with the requirements that Harald have written. I think the=
y<br>
should be formulated differently. First of all I think the requirements<br>
are negotiated, indicated etc in the context of a peer connection.<br>
<br>
Secondly, when it comes to maximum resolution and maximum frame-rate,<br>
there are actually three different limits that I think are important;<br>
Highest spatial resolution, highest frame-rate, and maximum complexity<br>
for a given video stream. The maximum complexity is often expressed as<br>
the number of macroblocks a video codec can process per second. This is<br>
a well-established complexity measure, used as part of standardized<br>
video codec&#39;s &quot;level&quot; definitions since the introduction of H=
.263 Annex<br>
X in 2004. As this is basically a joint maximum requirement on the<br>
maximum amount of pixels per frame times the maximum frame-rate, there<br>
exist cases where this complexity figure actually is the constraining<br>
factor forcing a sender or receiver to either request higher frame rate<br>
and lower resolution or higher resolution and a lower frame rate.<br>
<br>
The requirements should also be clearer in that one needs to handle<br>
multiple SSRCs per RTP session, including multiple cameras or audio<br>
streams (microphones) from a single end-point, where each stream can be<br>
encoded using different parameter values.<br>
<br>
I also think it is important that we consider what additional encoding<br>
property parameters that would make sense for WebRTC to have a<br>
possibility to dynamically negotiate, request and indicate.<br>
<br>
Some additional use cases that should be considered in the requirements:<br=
>
<br>
1) We have use cases including multi-party communication. One way to<br>
realize these is to use a central node and I believe everyone agrees<br>
that we should have good support for getting this usage to work well.<br>
Thus in a basic star topology of different participants there is going<br>
to be different path characteristics between the central node and each<br>
participant.<br>
<br>
1A) This makes it necessary to consider how one can ensure that the<br>
central node can deliver appropriate rates. One way is to de-couple the<br>
links by having the central node perform individual transcodings to<br>
every participant. A simpler non-transcoding central node, only<br>
forwarding streams between end-points, would have to enforce the lowest<br>
path characteristics to all. If one don&#39;t want to transcode at the<br>
central node and not use the lowest path characteristics to all, one<br>
need to consider either simulcast or scalable video coding. Both<br>
simulcast and scalable video coding result in that at least in the<br>
direction from a participant to the central node one needs to use<br>
multiple codec operation points. Either one per peer connection, which<br>
is how I see simulcast being realized with today&#39;s API and<br>
functionality, or using an encoding format supporting scalable coding<br>
within a single peer connection.<br>
<br>
1B) In cases where the central node has minimal functionality and<br>
basically is a relay and an ICE plus DTLS-SRTP termination point (I<br>
assume EKT to avoid having to do re-encryption), there is a need to be<br>
able to handle sources from different participants. This puts extra<br>
requirements on how to successfully negotiate the parameters. For<br>
example changing the values for one media source should not force one to<br=
>
renegotiate with everyone.<br>
<br>
2) The non-centralized multiparty use case appears to equally or more<br>
stress the need for having timely dynamic control. If each sender has a<br>
number of peer-connections to its peers, it may use local audio levels<br>
to determine if its media stream is to be sent or not. Thus the amount<br>
of bit-rate and screen estate needed to display will rapidly change as<br>
different users speaks. Thus the need for minimal delay when changing<br>
preferences are important.<br>
<br>
3) We also have speech and audio including audio-only use cases. For<br>
audio there could also exist desire to request or indicate changes in<br>
the audio bandwidth required, or usage of multi-channel.<br>
<br>
4) Adaptation to legacy node from a central node in a multi-party<br>
conference. In some use cases legacy nodes might have special needs that<br=
>
are within the profiles a WebRTC end-point is capable of producing. Thus<br=
>
the central node might request the nodes to constrain themselves to<br>
particular payload types, audio bandwidth etc to meet a joining session<br>
participant.<br>
<br>
5) There appear to exist a need for expressing dynamic requests for<br>
target bit-rate as one parameter. This can be supported by TMMBR<br>
(RFC5104) but there exist additional transport related parameters could<br>
help with the adaptation. These include MTU, limits on packet rate, and<br>
amount of aggregation of audio frames in the payload.<br>
<br>
Overview<br>
--------<br>
<br>
The basic idea in this proposal is to use JSEP to establish the outer<br>
limits for behavior and then use Codec Operation Point (COP) proposal as<br=
>
detailed in draft-westerlund-avtext-codec-operation-point to handle<br>
dynamic changes during the session.<br>
<br>
So highest resolution, frame-rate and maximum complexity are expressed<br>
in JSEP SDP. Complexity is in several video codecs expressed by profile<br>
and level. I know that VP8 currently doesn&#39;t have this but it is under<=
br>
discussion when it comes to these parameters.<br>
<br>
During the session the browser implementation detects when there is need<br=
>
to use COP to do any of the following things.<br>
<br>
A) Request new target values for codec operation, for example due to<br>
that the GUI element displaying a video has changed due to window resize.<b=
r>
<br>
B) Indicate when the end-point in its role as sender change parameters.<br>
<br>
In addition to just spatial resolution and video frame rate, I propose<br>
that the following parameters are considered as parameters that could be<br=
>
dynamically possible to indicate and request.<br>
<br>
Spatial resolution (as x and y resolution), Frame-rate, Picture Aspect<br>
ratio, Sample Aspect Ratio, Payload Type, Bit-rate, Token Bucket Size<br>
(To control burstiness of sender), Channels, Sampling Rate, Maximum RTP<br>
Packet Size, Maximum RTP Packet Rate, and Application Data Unit<br>
Aggregation (to control amount of audio frames in the same RTP packet).<br>
<br>
<br>
Differences<br>
-----------<br>
<br>
A) Using COP and using SDP based signaling for the dynamic changes are<br>
two quite different models in relation to how the interaction happens.<br>
<br>
For COP this all happens in the browser, normally initiated by the<br>
browser&#39;s own determination that a COP request or notification is<br>
needed. Harald&#39;s proposal appears to require that the JS initiate a<br>
renegotiation. This puts a requirement on the implementation to listen<br>
to the correct callbacks to know when changes happens, such as window<br>
resize. To my knowledge there are not yet any proposals for how the<br>
browser can initiate a JSEP renegotiation.<br></blockquote><div><br></div><=
div>Maybe I misunderstand what you are saying, but the application will def=
initely know when the browser size changes, and can then trigger a JSEP ren=
egotiation by calling createOffer and shipping it off.</div>

<div><br></div><div>I have the opposite concern - how can the browser know =
when the application makes a change, for instance to display a participant =
in a large view versus a small view? This may be possible if using a &lt;vi=
deo/&gt; for display, but I don&#39;t think it will be possible when using =
WebGL for rendering, where the size of the view will be dictated only by th=
e geometry of the scene.</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">
<br>
Thus COP has the advantage that there is no API changes to get browser<br>
triggered parameter changes. W3C can select too but are not required to<br>
add API methods to allow JS to make codec parameter requests.<br>
<br>
The next thing is that COP does not require the JS application to have<br>
code to detect and handle re-negotiation. This makes it simpler for the<br>
basic application to get good behavior and they are not interrupted nor<br>
do they need to handle JSEP &amp; Offer/Answer state machine lock-out<br>
effects due to dynamic changes.<br>
<br>
How big impact these API issues have are unclear as W3C currently appear<br=
>
not to have included any discussion of how the browser can initiate a<br>
offer/answer exchange towards the JS when it determines a need to change<br=
>
parameters.<br>
<br>
But I am worried that using SDP and with an API that requires the<br>
application to listen for triggers that could benefit from a codec<br>
parameter renegotiation. This will likely only result in good behavior<br>
for the JS application implementors that are really good and find out<br>
what listeners and what signaling tricks are needed with JSEP to get<br>
good performance. I would much rather prefer good behavior by default in<br=
>
simple applications, i.e. using the default behavior that the browser<br>
implementor have put in.<br></blockquote><div><br></div><div>The issue here=
 is that this behavior needs to be sufficiently specified, or else we will =
have many different behaviors, none of which can be fully overridden by the=
 app. I worry this will make life hard for people trying to develop sophist=
icated apps.</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<br>
B) Using the media plane, i.e. RTCP for this signaling lets it in most<br>
case go directly between the encoding and the decoding entity in the<br>
code. There is no need to involve the JS nor the signaling server. One<br>
issue of using JSEP and SDP is that the state machine lock-out effects<br>
that can occur if one has sent an Offer. Then that browser may not be<br>
able to send a new updated Offer reflecting the latest change until the<br>
answer has been properly processed. COP doesn&#39;t have these limitations.=
<br>
It can send a new parameter request immediately, only limited by RTCP<br>
bandwidth restrictions. Using the media plane in my view guarantees that<br=
>
COP is never worse than what the signaling plane can perform at its best.<b=
r>
<br>
C) As the general restrictions are determined in the initial<br>
negotiation, COP doesn&#39;t have the issue that in-flight media streams ca=
n<br>
become out of bounds. Thus no need for a two phase change of signaling<br>
parameters.<br>
<br>
D) Relying on draft-lennox-mmusic-sdp-source-selection-04 has several<br>
additional implications that should be discussed separately. The draft<br>
currently includes the following functionalities.<br>
<br>
 =C2=A0D1) It contains a media stream pause proposal. This makes it subject=
<br>
to the architectural debate currently ongoing in dispatch around<br>
draft-westerlund-avtext-rtp-stream-pause-00 which is a competing<br>
proposal for the same functionality.<br>
<br>
 =C2=A0D2) The inclusion of max desired frame rate in a SSRC specific way<b=
r>
<br>
 =C2=A0D3) Extending the image attribute to be per SSRC specific expression=
<br>
of desired resolutions.<br>
<br>
 =C2=A0D4) Expressing relative priority in receiving different SSRCs.<br>
<br>
 =C2=A0D5) Providing an &quot;information&quot; on a sent SSRC<br>
<br>
 =C2=A0D6) Indication if the media sender is actively sending media using t=
he<br>
given SSRC.<br>
<br>
Is it a correct observation that only D2 and D3 are required for the<br>
functionality of Resolution negotiation?<br>
<br>
E) The standardization situation is similar for both proposals. They are<br=
>
both relying on Internet drafts that are currently individual<br>
submissions. Both are partially caught in the architectural discussion<br>
which was held in Paris in the DISPATCH WG around Media pause/resume<br>
(draft-westerlund-avtext-rtp-stream-pause-00) and Media Stream Selection<br=
>
(draft-westerlund-dispatch-stream-selection-00) on what the most<br>
appropriate level of discussion are. This discussion will continue on<br>
the RAI-Area mailing list.<br>
<br>
F) As seen by the discussion on the mailing list the imageattr<br>
definitions may not be a 100% match to what is desired with Harald&#39;s<br=
>
proposal. I believe that COP&#39;s are more appropriate especially the<br>
&quot;target&quot; values possibility. In addition these are still open for=
<br>
adjustment and if they don&#39;t match WebRTC&#39;s requirements.<br>
<br>
I would also like to point out that I believe this functionality is also<br=
>
highly desirable for CLUE and that their requirements should be taken<br>
into account. I do think that this is one of the aspects where having<br>
matching functionality will make it much easier to have WebRTC to CLUE<br>
interworking.<br>
<br>
Thanks for reading all the way here!<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<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 71=
48287</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">+46=
 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<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>

--20cf303b42db224db704be785bda--

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Subject: Re: [rtcweb] Alternative Proposal for Dynamic Codec Parameter Change (Was: Re: Resolution negotiation - a contribution)
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On 2012-04-25 05:20, Justin Uberti wrote:
> On Tue, Apr 24, 2012 at 10:25 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     A) Using COP and using SDP based signaling for the dynamic changes are
>     two quite different models in relation to how the interaction happens.
> 
>     For COP this all happens in the browser, normally initiated by the
>     browser's own determination that a COP request or notification is
>     needed. Harald's proposal appears to require that the JS initiate a
>     renegotiation. This puts a requirement on the implementation to listen
>     to the correct callbacks to know when changes happens, such as window
>     resize. To my knowledge there are not yet any proposals for how the
>     browser can initiate a JSEP renegotiation.
> 
> 
> Maybe I misunderstand what you are saying, but the application will
> definitely know when the browser size changes, and can then trigger a
> JSEP renegotiation by calling createOffer and shipping it off.

Yes, they can but that requires them to write code to do this. With COP
the browser can do it and it works well for all applications, not only
the ones that have implementors that are aware of the need to do this.

I am proposing a mechanism that makes the default behavior if you care
nothing about controlling or optimizing this behavior in your JS to be
good. In addition JS that wants to control it can affect the behavior
through generic mechanisms, such as controlling the video element. And
if clear need is shown and some functionality is missing, W3C can create
an API to provide explicit control.

> 
> I have the opposite concern - how can the browser know when the
> application makes a change, for instance to display a participant in a
> large view versus a small view? This may be possible if using a <video/>
> for display, but I don't think it will be possible when using WebGL for
> rendering, where the size of the view will be dictated only by the
> geometry of the scene.

What I understand from my colleagues is that all rendering of video have
to go through a video element, even if not displayed directly. Thus the
application can control the size of the video element used and possibly
consumed by WebGL to be included in rendering. Thus the application will
have control over this even when COP is being used.

Can you be more specific on any particular usage that you think would
have issues?



> 
>     But I am worried that using SDP and with an API that requires the
>     application to listen for triggers that could benefit from a codec
>     parameter renegotiation. This will likely only result in good behavior
>     for the JS application implementors that are really good and find out
>     what listeners and what signaling tricks are needed with JSEP to get
>     good performance. I would much rather prefer good behavior by default in
>     simple applications, i.e. using the default behavior that the browser
>     implementor have put in.
> 
> 
> The issue here is that this behavior needs to be sufficiently specified,
> or else we will have many different behaviors, none of which can be
> fully overridden by the app. I worry this will make life hard for people
> trying to develop sophisticated apps.


I agree that it needs to be clearly specified. But COP appears to be
less prone to complete failures by using target values. Then at least it
will receive video that can be used, even if not optimal. imageattr can
result in failures in the negotiation requiring at minimal a second
round potentially no agreed on acceptable resolution then that is
clearly sub-optimal. That is even without considering involving JS
mainipulating this.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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 harald@alvestrand.no  Wed Apr 25 09:13:59 2012
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Date: Wed, 25 Apr 2012 18:13:54 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Alternative Proposal for Dynamic Codec Parameter Change (Was: Re: Resolution negotiation - a contribution)
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On 04/25/2012 04:35 PM, Magnus Westerlund wrote:
> On 2012-04-25 05:20, Justin Uberti wrote:
>> On Tue, Apr 24, 2012 at 10:25 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com<mailto:magnus.westerlund@ericsson.com>>
>> wrote:
>>
>>      A) Using COP and using SDP based signaling for the dynamic changes are
>>      two quite different models in relation to how the interaction happens.
>>
>>      For COP this all happens in the browser, normally initiated by the
>>      browser's own determination that a COP request or notification is
>>      needed. Harald's proposal appears to require that the JS initiate a
>>      renegotiation. This puts a requirement on the implementation to listen
>>      to the correct callbacks to know when changes happens, such as window
>>      resize. To my knowledge there are not yet any proposals for how the
>>      browser can initiate a JSEP renegotiation.
>>
>>
>> Maybe I misunderstand what you are saying, but the application will
>> definitely know when the browser size changes, and can then trigger a
>> JSEP renegotiation by calling createOffer and shipping it off.
> Yes, they can but that requires them to write code to do this. With COP
> the browser can do it and it works well for all applications, not only
> the ones that have implementors that are aware of the need to do this.
>
> I am proposing a mechanism that makes the default behavior if you care
> nothing about controlling or optimizing this behavior in your JS to be
> good. In addition JS that wants to control it can affect the behavior
> through generic mechanisms, such as controlling the video element. And
> if clear need is shown and some functionality is missing, W3C can create
> an API to provide explicit control.
It's not at all clear to me that the browser being in control is right 
for all cases.
Renegotiating has a cost, and some of the costs are known only to the 
application.

We may possibly handle this via constraints like 
"VideoEnumRenegotiationTendency" or something like that - when you get 
the proposal together more, I'd like to see the controls you're proposing.
>> I have the opposite concern - how can the browser know when the
>> application makes a change, for instance to display a participant in a
>> large view versus a small view? This may be possible if using a<video/>
>> for display, but I don't think it will be possible when using WebGL for
>> rendering, where the size of the view will be dictated only by the
>> geometry of the scene.
> What I understand from my colleagues is that all rendering of video have
> to go through a video element, even if not displayed directly. Thus the
> application can control the size of the video element used and possibly
> consumed by WebGL to be included in rendering. Thus the application will
> have control over this even when COP is being used.
We have on the "list of things that need doing" the connection of a 
mediastream directly to a canvas, a WebGL element or being captured to a 
file, so don't gamble too much on the video element to stay in the path.

Example code for capturing video to a canvas (taken from a real app):

        video.src = webkitURL.createObjectURL(s);
        canvas.getContext("2d").drawImage(video, 0, 0, 300, 300, 0, 0, 
300, 300);

I am not sure where the size controls go. I'm not even sure the 
developer knows.

> Can you be more specific on any particular usage that you think would
> have issues?
>
>
>
>>      But I am worried that using SDP and with an API that requires the
>>      application to listen for triggers that could benefit from a codec
>>      parameter renegotiation. This will likely only result in good behavior
>>      for the JS application implementors that are really good and find out
>>      what listeners and what signaling tricks are needed with JSEP to get
>>      good performance. I would much rather prefer good behavior by default in
>>      simple applications, i.e. using the default behavior that the browser
>>      implementor have put in.
>>
>>
>> The issue here is that this behavior needs to be sufficiently specified,
>> or else we will have many different behaviors, none of which can be
>> fully overridden by the app. I worry this will make life hard for people
>> trying to develop sophisticated apps.
>
> I agree that it needs to be clearly specified. But COP appears to be
> less prone to complete failures by using target values. Then at least it
> will receive video that can be used, even if not optimal. imageattr can
> result in failures in the negotiation requiring at minimal a second
> round potentially no agreed on acceptable resolution then that is
> clearly sub-optimal. That is even without considering involving JS
> mainipulating this.
I certainly hope that we can avoid the need for SDP manipulation of JS 
in all of these proposals.


From dbenham@cisco.com  Wed Apr 25 17:32:23 2012
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Thread-Topic: [rtcweb] Proposal for H.263 baseline codec
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From: "David Benham (dbenham)" <dbenham@cisco.com>
To: "Gregory Maxwell" <gmaxwell@juniper.net>, <sergel@google.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Proposal for H.263 baseline codec
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Great summary & clarification, Gregory.

Serge
When one receives/signs the MPEG-LA license for H264 pool, all of the
patents (~8 pages worth) are listed so that the licensee knows what they
are covered for.  =20

Wasn't able to find a list of patents currently licensable/owned by
Google that are included in the royalty free VP8 bitstream spec or
implementation license.  With that info, presumably one's experts could
make a risk assessment (yes, easier said than done) as to what might
exist but *not* be covered by the VP8 license.   Has Google published
the list of patents minimally covered by your VP8 licenses?
Independently searching  through ON2 patents awarded/filed might perhaps
come close, but a definitive list from Google would be much more
reliable.


> -----Original Message-----
> From: Gregory Maxwell [mailto:gmaxwell@juniper.net]
> Sent: Friday, April 20, 2012 2:19 PM
> To: EXT - kpfleming@digium.com; rtcweb@ietf.org
> Subject: Re: [rtcweb] Proposal for H.263 baseline codec
>=20
> pfleming@digium.com wrote:
> >On 03/29/2012 09:53 AM, Stephan Wenger wrote:
> >> The other issue, though (the fact that the license grant extends
only to
> >> the VP8 implementation as provided by google, and does not extent
to
> >> derivative works such as hardware implementations) should be
> moderately
> > This is concerning, even for us open source software distributors.
In
> > fact, a similar situation existed some time ago with iLBC; the
license
> > that GIPS offered covered only the code as distributed as part of
the
> > RFC (although the language stating this was quite poorly
constructed),
>=20
> Stop. All of you are confused.
>=20
> Maybe I can help-
>=20
> There are _two_  VP8 patent license grants:
>=20
> One for the specification:
http://www.webmproject.org/license/bitstream/
> One for the software: http://www.webmproject.org/license/additional/
>=20
> These grants are subtly different in the scope of what they cover.
>=20
> There are important licensing drafting reasons for a construction
> which distinguishes "an implementation" and "any implementation".
> In particular, the license drafter writing this kind of a royalty free
> license has a tricky time making sure that the permission is broad
> enough to include all the correct coverage,  but not so over-broad
that
> a malicious implementer could (e.g.) add a search engine to their VP8
> encoder and claim that the VP8 license entitles them to search
patents.
>=20
> So, the drafter says something like- "those patent claims [...] that
> are necessarily infringed by implementation of this specification",
> which achieves the desired limited scope- and that is _all_ you get
> with the MPEG-LA pool patent licenses. This is what the Google
bitstream
> spec license provides.
>=20
> But, if we're to be serious and honest about making a royalty free
format
> and implementation that permission isn't quite sufficient: The
reference
> implementation might practice some VP8 related techniques which are
not
> "necessarily infringed"- they're helpful performance enhancements,
> they could simply be one option out of many ways of doing the same
thing,
> or could just be something that was incidentally included (or even a
> malicious patent trap snake in the grass)- and so it's important that
> everything in the reference software also be licensed- even if its not
> "necessarily infringed" by the specification.   But since it's not a
> goal to also license unrelated things that a downstream user may cram
in
> (e.g. a search engine), that license must be limited in scope to
things
> which are actually in the reference implementation: "This grant does
not
> include claims that would be infringed only as a consequence of
further
> modification of this implementation." (this language is actually
almost
> identical to the language for this purpose in the GPL, FWIW)
>=20
> So, in summary:
>=20
> You are licensed, via the bitstream license, for anything Google
controls
> which is necessary to implement the specification- in hardware,
> software, home grown, or reference implementations.  This is pretty
> much as strong a grant as you can find anyone offering for any other
> format specification.
>=20
> You are also licensed, via the additional rights license, for any
> use of the VP8 reference implementation including modified versions,
> limited to the extent that the modifications don't expand the scope of
> the coverage. (Exactly as the GPL explicit patent grant does)
>=20
> This has already been clarified by Google on this list:
> "Google confirms that the VP8 patent grant applies to both
> third-party hardware and software implementations of VP8."
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04006.html
>=20
> I admit that the fact that there are two similar but distinct licenses
> is a little confusing, but I don't understand why we have to go over
> it again and again on this list.  I would think that the prior Google
> comment would have been sufficiently definitive.


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The chairs would like to ask the working group to focus on the use
case draft.  If you have use cases that need to be added to the
document or text changes you'd like to suggest, please send them in
for discussion before May 15th.  After this round, we will look toward
having a working group last call on the document (hopefully before the
interim meeting).

regards,

Ted, Magnus, Cullen

From Jim.Barnett@genesyslab.com  Fri Apr 27 09:36:02 2012
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I would like to see a corporate call center use case.  Specifically, a
user downloads a web page from a corporate web site, clicks a 'call us'
button and is connected to a gateway server that is controlled by the
corporation.  The communication up to the corporate boundary cannot be
eavesdropped, but, inside the corporate boundary:  1) the corporation
can route the call to whoever it wants (meaning that the caller can
verify that he is connected to the corporation, but is not necessarily
assured of the identity of the person he is speaking to within the
corporation) 2) the corporation can eavesdrop/record the call (n.b. this
is mandatory in financial institutions, and common in most others).=20

This corresponds to a very common current PSTN use case (except that
with webRTC the call is more secure up to the corporate boundary).  I
think that corporations will be eager to add webRTC support to their
call centers - as long as it doesn't mess up their existing operations
(call routing, recording, etc.)  They will most likely want to put in a
gateway, and treat it as the webRTC endpoint.  Inside the gateway the
call should look just like one that came in from the PSTN (via a SIP
trunk or PSTN/SIP gateway.)

I think others have suggested use cases involving outbound calls from
corporations, but I think that those should probably be treated
separately. =20

- Jim=20

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Ted Hardie
Sent: Friday, April 27, 2012 12:15 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Use Case draft

The chairs would like to ask the working group to focus on the use case
draft.  If you have use cases that need to be added to the document or
text changes you'd like to suggest, please send them in for discussion
before May 15th.  After this round, we will look toward having a working
group last call on the document (hopefully before the interim meeting).

regards,

Ted, Magnus, Cullen
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From tterriberry@mozilla.com  Fri Apr 27 09:43:55 2012
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Ted Hardie wrote:
> The chairs would like to ask the working group to focus on the use
> case draft.  If you have use cases that need to be added to the
> document or text changes you'd like to suggest, please send them in

I proposed the following use-case back in February, but there wasn't 
much discussion on actually adding it to the document: 
http://www.ietf.org/mail-archive/web/rtcweb/current/msg03357.html

Let me know if the WG would like to proceed with something like this.

From dwing@cisco.com  Fri Apr 27 09:48:59 2012
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From: "Dan Wing" <dwing@cisco.com>
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Subject: [rtcweb] use cases, F20 and encryption, SCTP - comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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> The chairs would like to ask the working group to focus on the use
> case draft.  If you have use cases that need to be added to the
> document or text changes you'd like to suggest, please send them in
> for discussion before May 15th.  After this round, we will look
> toward having a working group last call on the document (hopefully
> before the interim meeting).

A few comments on draft-ietf-rtcweb-use-cases-and-requirements-07:


1. Requirement F20 states:

   F20  It MUST be possible to protect streams from eavesdropping.

Consensus in the room during my presentation to RTCWEB at IETF83 was that we
don't need to support un-encrypted media (RTP) at all, and that all media
would be SRTP.  Can that be captured in F20 by re-wording, or perhaps in a
new requirement if we can't reword F20?  If there is a need or desire to
validate that consensus on list, let's please ask the chairs to do that.


2. I noticed there is no requirement that we have a baseline for how SRTP
media is keyed (although there is a baseline requirement for codecs).  This
is a critical requirement.  I suggest adding "The browser MUST support a
baseline SRTP keying mechanism."  We have not reached consensus on that
keying mechanism, but the requirement is real.


3. I see the document restricts its scope to media streams in the
Introduction with:

  "The document focuses on requirements related to real-time media
   streams.  Requirements related to privacy, signalling between the
   browser and web server etc. are currently not considered."

However, RTCWEB is also supports a data communication between browsers.  I
am worried if we do not specify requirements for the data communication we
will have problems.  I believe the expectation is that if the audio/video
stream works, that the data communication stream also work.  We need to
capture requirements for the data communication stream somewhere:

  - a requirement to support data communication
  - that the chosen data communication protocol supports multiple streams
(which is why SCTP was chosen over TCP)
  - for NAT/firewall traversal of the data communication protocol (which is
why SCTP-over-UDP was chosen and another reason TCP was not chosen)
  - for encrypting that data communication session
  - a requirement for SCTP-over-UDP to work when UDP is blocked (aligning
with the existing F29 for audio/video)
  - a requirement to do ICE connectivity checks prior to bringing up DTLS (I
don't know if that is really a requirement, but I recall it mentioned at the
RTCWEB interim in Mountain View).  

Based on the scoping of the draft-ietf-rtcweb-use-cases-and-requirements,
the omission of the data communication stream is intentional.  If not in
draft-ietf-rtcweb-use-cases-and-requirements, where can we capture the
requirements for the data communication stream?

-d



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Subject: Re: [rtcweb] use cases, F20 and encryption, SCTP - comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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Dan Wing wrote:
> is a critical requirement.  I suggest adding "The browser MUST support a
> baseline SRTP keying mechanism."  We have not reached consensus on that
> keying mechanism, but the requirement is real.

On the contrary, the conclusion of the consensus call in 
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04125.html said: 
"...no one objected against making DTLS-SRTP a mandatory to implement 
and the default keying mechanism."

Regardless of the debate over SDES, that sounds like we have _a_ 
baseline SRTP keying mechanism.

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Subject: Re: [rtcweb] use cases, F20 and encryption, SCTP - comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 04/27/2012 06:48 PM, Dan Wing wrote:
>
> 3. I see the document restricts its scope to media streams in the
> Introduction with:
>
>    "The document focuses on requirements related to real-time media
>     streams.  Requirements related to privacy, signalling between the
>     browser and web server etc. are currently not considered."
It seems that the intro is inconsistent with the rest of the document, then.
The following requirements all require non-media browser-to-browser streams:

    F23             The browser must be able to send short
                    latency datagram traffic to a peer browser
    ----------------------------------------------------------------
   A18             In addition to the streams listed elsewhere,
                    the Web API MUST provide a mechanism for sending
                    and receiving isolated discrete chunks of data.
    ----------------------------------------------------------------

These are both derived from use case 4.2.11, "Multiparty on-line game 
with voice communication".

> However, RTCWEB is also supports a data communication between browsers.  I
> am worried if we do not specify requirements for the data communication we
> will have problems.  I believe the expectation is that if the audio/video
> stream works, that the data communication stream also work.  We need to
> capture requirements for the data communication stream somewhere:
>
>    - a requirement to support data communication
Present
>    - that the chosen data communication protocol supports multiple streams
> (which is why SCTP was chosen over TCP)
I'm not sure what scenario would support that - the discussion so far 
seemed to hinge on the idea that people would need multiple streams no 
matter what, so we'd either have multiple streams in the protocol, or 
people would invent their own multiplexing scenarios on top of what we 
provided.
>    - for NAT/firewall traversal of the data communication protocol (which is
> why SCTP-over-UDP was chosen and another reason TCP was not chosen)
>    - for encrypting that data communication session
>    - a requirement for SCTP-over-UDP to work when UDP is blocked (aligning
> with the existing F29 for audio/video)
This one I'm sure we need to discuss more. SCTP-over-DTLS-over-TURN?
>    - a requirement to do ICE connectivity checks prior to bringing up DTLS (I
> don't know if that is really a requirement, but I recall it mentioned at the
> RTCWEB interim in Mountain View).
I think this one is chiefly an implementation strategy to achieve the 
security requirements outlined in the -security draft.
> Based on the scoping of the draft-ietf-rtcweb-use-cases-and-requirements,
> the omission of the data communication stream is intentional.  If not in
> draft-ietf-rtcweb-use-cases-and-requirements, where can we capture the
> requirements for the data communication stream?
I think the scoping needs rewording.
> -d
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Harald, thank you very much for the careful read. I have updated =
addressing all your comments as well as some other fixes I got.=20

Latest minutes are at

http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.txt



On Apr 16, 2012, at 6:01 AM, Harald Alvestrand wrote:

> Thanks for these!
>=20
> The only serious comments I have are on the codec discussion near the =
end; otherwise, there are grammar nits.
>=20
> Notes:
> - Please convert to ASCII. Quotes and non-ASCII characters are broken.
> - "One attack by claim to link to something, but what you really =
connect to is some other site"
> Grammar issue: by claim -> by claiming
> - "In Jingle to enable early response with the ICE candidates and the =
username and password are anyway needed." -> could not parse this =
sentence. Please reword.
> - One instance of "Collin Perkins" (misspelled name)
> - "straight forward" -> "straightforward"
> - "Regarding the key-exchange and the round trip time. We will
> encounter cases, like the trombone where the signaling server is far
> away and the media peer is close." - the first full stop should be a =
comma for grammatical correctness.
> - "no overhead in a waste majority of cases." -> "no overhead in the =
vast majority of cases"
> - "The chairs declared No Consensus between these two options due to =
very
> evenly strength hums." - grammatically, it needs to be "even strength =
hums"; I would formulate it as "approximately even strength hums", given =
that I don't remember them as *that* even (but I was not at the front of =
the room).
>=20
> - "Harald: I hope we can limit the number of cases where SDP =
manipulation
> is limited." - I certainly hope I said "needed" as the last word.
> - "Harald regards hints as steering wheel and brake pedal,
> while SDP is plugging in to engine control. Hopes can do what we need
> via hints." - "in to" -> "into"; "Hopes can do" -> "He hopes that we =
can do".
>=20
> - "you are talking of SCT/DTLS/UDP." -> SCTP/DTLS/UDP
> - "Tim (Mozilla)" -> Tim Terriberry
> - Tim P -> Tim Panton
>=20
> "Cullen: raise your hand in favor of H.261
>                                    H.264
>                                    VP8"
>=20
> The numbers (or proportions) need to be part of the minutes, and the =
minutes need to make it clear that this was a straw poll, not a hum for =
consensus (Ted made that very clear, but it's not clear from the =
minutes.)
>=20
> Allistair ??: The discussion was the same in H.323, and G.711 was
> selected as audio codec because of IPR free. None believes it would be
> useful, but this stopped the discussions.
>=20
> I believe this was followed by a remark saying a huge amount of =
current traffic is G.711 or worse.
>=20
>=20
> On 04/16/2012 10:41 AM, Magnus Westerlund wrote:
>> On 2012-04-14 00:38, Ted Hardie wrote:
>>> The proceedings seem to be set up to prefer a single set of minutes,
>>> rather than one per session.  I've uploaded a set now that includes
>>> the ones Magnus originally put up, plus minutes from the second
>>> session.  Many thanks to all of our note takers for their hard work.
>>>=20
>>> Please review the minutes and send corrections,
>> A link to the version containing both sessions are:
>>=20
>> http://www.ietf.org/proceedings/83/minutes/minutes-83-rtcweb.txt
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=20
>> =
----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> 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
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] use cases, F20 and encryption, SCTP - comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 4/27/2012 12:48 PM, Dan Wing wrote:
>> The chairs would like to ask the working group to focus on the use
>> case draft.  If you have use cases that need to be added to the
>> document or text changes you'd like to suggest, please send them in
>> for discussion before May 15th.  After this round, we will look
>> toward having a working group last call on the document (hopefully
>> before the interim meeting).
> A few comments on draft-ietf-rtcweb-use-cases-and-requirements-07:
>
>
> 1. Requirement F20 states:
>
>     F20  It MUST be possible to protect streams from eavesdropping.
>
> Consensus in the room during my presentation to RTCWEB at IETF83 was that we
> don't need to support un-encrypted media (RTP) at all, and that all media
> would be SRTP.  Can that be captured in F20 by re-wording, or perhaps in a
> new requirement if we can't reword F20?  If there is a need or desire to
> validate that consensus on list, let's please ask the chairs to do that.
>
>
> 2. I noticed there is no requirement that we have a baseline for how SRTP
> media is keyed (although there is a baseline requirement for codecs).  This
> is a critical requirement.  I suggest adding "The browser MUST support a
> baseline SRTP keying mechanism."  We have not reached consensus on that
> keying mechanism, but the requirement is real.
>
>
> 3. I see the document restricts its scope to media streams in the
> Introduction with:
>
>    "The document focuses on requirements related to real-time media
>     streams.  Requirements related to privacy, signalling between the
>     browser and web server etc. are currently not considered."
>
> However, RTCWEB is also supports a data communication between browsers.  I
> am worried if we do not specify requirements for the data communication we
> will have problems.  I believe the expectation is that if the audio/video
> stream works, that the data communication stream also work.  We need to
> capture requirements for the data communication stream somewhere:

This is a surprise to me; I thought data channels (in some form) were 
always part of the requirements.  It seems like they got abstracted a 
bit in editing - 4.2.11 (online game with voice) covers it, and it still 
says "Quick updates of the game state is [sic] required".  This used to 
be F20, and is now F23

F23  The browser must be able to send short latency datagram traffic to a peer browser

However, I agree this is rather too vague, but I think we wanted to 
leave open one stream vs multiple, reliable vs not (though latency 
implies unreliable), etc.

>    - a requirement to support data communication

Covered by F23

>    - that the chosen data communication protocol supports multiple streams
> (which is why SCTP was chosen over TCP)
>    - for NAT/firewall traversal of the data communication protocol (which is
> why SCTP-over-UDP was chosen and another reason TCP was not chosen)

Implied by F23, F19 says "Streams MUST be able to pass through 
restrictive firewalls" -- not "Media streams", and F2 says   "The 
browser MUST be able to send streams to a peer in the presence of NATs."

The requirements use "streams", "audio streams", "video streams" and 
"media streams".  Some uses of plain "streams" only make sense for 
media; others can obviously make sense for media or data.

>    - for encrypting that data communication session

F20 says "It MUST be possible to protect streams from eavesdropping".  
Again, if "streams" includes data streams, it's already covered.  If 
not, then yes, this should be a requirement.


>    - a requirement for SCTP-over-UDP to work when UDP is blocked (aligning
> with the existing F29 for audio/video)

Ditto again the "streams" remark.

>    - a requirement to do ICE connectivity checks prior to bringing up DTLS (I
> don't know if that is really a requirement, but I recall it mentioned at the
> RTCWEB interim in Mountain View).
>
> Based on the scoping of the draft-ietf-rtcweb-use-cases-and-requirements,
> the omission of the data communication stream is intentional.  If not in
> draft-ietf-rtcweb-use-cases-and-requirements, where can we capture the
> requirements for the data communication stream?

I'm pretty certain it wasn't intentional, given F23 (which has been 
there since the start), and that was the understanding of the 
participants, I would assert.

I agree it could use fleshing out (and the other requirements could use 
more consistent uses of "streams" to avoid confusion).

-- 
Randell Jesup
randell-ietf@jesup.org


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> 1. Requirement F20 states:
>
> =C2=A0 F20 =C2=A0It MUST be possible to protect streams from eavesdroppin=
g.
>
> Consensus in the room during my presentation to RTCWEB at IETF83 was that=
 we
> don't need to support un-encrypted media (RTP) at all, and that all media
> would be SRTP. =C2=A0Can that be captured in F20 by re-wording, or perhap=
s in a
> new requirement if we can't reword F20? =C2=A0If there is a need or desir=
e to
> validate that consensus on list, let's please ask the chairs to do that.

This requirement probably requires a bit more examination.  I tend to
agree that something more specific regarding SRTP would be good, but
that doesn't cover the whole story.

SRTP alone doesn't protect against the site (or sites).  If there is
mutual authentication, then it might be possible to prevent an
application from being able to eavesdrop (conditional on some degree
of trust in the identity provider).

From bernard_aboba@hotmail.com  Sun Apr 29 10:20:59 2012
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I agree that the corporate call center use case is important.=20

> Date: Fri=2C 27 Apr 2012 09:35:55 -0700
> From: Jim.Barnett@genesyslab.com
> To: ted.ietf@gmail.com=3B rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
>=20
> I would like to see a corporate call center use case.  Specifically=2C a
> user downloads a web page from a corporate web site=2C clicks a 'call us'
> button and is connected to a gateway server that is controlled by the
> corporation.  The communication up to the corporate boundary cannot be
> eavesdropped=2C but=2C inside the corporate boundary:  1) the corporation
> can route the call to whoever it wants (meaning that the caller can
> verify that he is connected to the corporation=2C but is not necessarily
> assured of the identity of the person he is speaking to within the
> corporation) 2) the corporation can eavesdrop/record the call (n.b. this
> is mandatory in financial institutions=2C and common in most others).=20
>=20
> This corresponds to a very common current PSTN use case (except that
> with webRTC the call is more secure up to the corporate boundary).  I
> think that corporations will be eager to add webRTC support to their
> call centers - as long as it doesn't mess up their existing operations
> (call routing=2C recording=2C etc.)  They will most likely want to put in=
 a
> gateway=2C and treat it as the webRTC endpoint.  Inside the gateway the
> call should look just like one that came in from the PSTN (via a SIP
> trunk or PSTN/SIP gateway.)
>=20
> I think others have suggested use cases involving outbound calls from
> corporations=2C but I think that those should probably be treated
> separately. =20
>=20
> - Jim=20
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Ted Hardie
> Sent: Friday=2C April 27=2C 2012 12:15 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Use Case draft
>=20
> The chairs would like to ask the working group to focus on the use case
> draft.  If you have use cases that need to be added to the document or
> text changes you'd like to suggest=2C please send them in for discussion
> before May 15th.  After this round=2C we will look toward having a workin=
g
> group last call on the document (hopefully before the interim meeting).
>=20
> regards=2C
>=20
> Ted=2C Magnus=2C Cullen
> _______________________________________________
> 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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<body class=3D'hmmessage'><div dir=3D'ltr'>
I agree that the corporate call center use case is important. <br><br><div>=
<div id=3D"SkyDrivePlaceholder"></div>&gt=3B Date: Fri=2C 27 Apr 2012 09:35=
:55 -0700<br>&gt=3B From: Jim.Barnett@genesyslab.com<br>&gt=3B To: ted.ietf=
@gmail.com=3B rtcweb@ietf.org<br>&gt=3B Subject: Re: [rtcweb] Use Case draf=
t<br>&gt=3B <br>&gt=3B I would like to see a corporate call center use case=
.  Specifically=2C a<br>&gt=3B user downloads a web page from a corporate w=
eb site=2C clicks a 'call us'<br>&gt=3B button and is connected to a gatewa=
y server that is controlled by the<br>&gt=3B corporation.  The communicatio=
n up to the corporate boundary cannot be<br>&gt=3B eavesdropped=2C but=2C i=
nside the corporate boundary:  1) the corporation<br>&gt=3B can route the c=
all to whoever it wants (meaning that the caller can<br>&gt=3B verify that =
he is connected to the corporation=2C but is not necessarily<br>&gt=3B assu=
red of the identity of the person he is speaking to within the<br>&gt=3B co=
rporation) 2) the corporation can eavesdrop/record the call (n.b. this<br>&=
gt=3B is mandatory in financial institutions=2C and common in most others).=
 <br>&gt=3B <br>&gt=3B This corresponds to a very common current PSTN use c=
ase (except that<br>&gt=3B with webRTC the call is more secure up to the co=
rporate boundary).  I<br>&gt=3B think that corporations will be eager to ad=
d webRTC support to their<br>&gt=3B call centers - as long as it doesn't me=
ss up their existing operations<br>&gt=3B (call routing=2C recording=2C etc=
.)  They will most likely want to put in a<br>&gt=3B gateway=2C and treat i=
t as the webRTC endpoint.  Inside the gateway the<br>&gt=3B call should loo=
k just like one that came in from the PSTN (via a SIP<br>&gt=3B trunk or PS=
TN/SIP gateway.)<br>&gt=3B <br>&gt=3B I think others have suggested use cas=
es involving outbound calls from<br>&gt=3B corporations=2C but I think that=
 those should probably be treated<br>&gt=3B separately.  <br>&gt=3B <br>&gt=
=3B - Jim <br>&gt=3B <br>&gt=3B -----Original Message-----<br>&gt=3B From: =
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf<br>&gt=
=3B Of Ted Hardie<br>&gt=3B Sent: Friday=2C April 27=2C 2012 12:15 PM<br>&g=
t=3B To: rtcweb@ietf.org<br>&gt=3B Subject: [rtcweb] Use Case draft<br>&gt=
=3B <br>&gt=3B The chairs would like to ask the working group to focus on t=
he use case<br>&gt=3B draft.  If you have use cases that need to be added t=
o the document or<br>&gt=3B text changes you'd like to suggest=2C please se=
nd them in for discussion<br>&gt=3B before May 15th.  After this round=2C w=
e will look toward having a working<br>&gt=3B group last call on the docume=
nt (hopefully before the interim meeting).<br>&gt=3B <br>&gt=3B regards=2C<=
br>&gt=3B <br>&gt=3B Ted=2C Magnus=2C Cullen<br>&gt=3B ____________________=
___________________________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb@=
ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br>&gt=3B _=
______________________________________________<br>&gt=3B rtcweb mailing lis=
t<br>&gt=3B rtcweb@ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinfo=
/rtcweb<br></div> 		 	   		  </div></body>
</html>=

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, "jim.barnett@genesyslab.com" <jim.barnett@genesyslab.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Use Case draft
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+1

The consumer call center shall support "anonymous" calling wherein there is=
 no need of specific identity mechanism for the caller.

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bernard Aboba
Sent: Sunday, April 29, 2012 10:51 PM
To: jim.barnett@genesyslab.com; rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

I agree that the corporate call center use case is important.
> Date: Fri, 27 Apr 2012 09:35:55 -0700
> From: Jim.Barnett@genesyslab.com<mailto:Jim.Barnett@genesyslab.com>
> To: ted.ietf@gmail.com<mailto:ted.ietf@gmail.com>; rtcweb@ietf.org<mailto=
:rtcweb@ietf.org>
> Subject: Re: [rtcweb] Use Case draft
>
> I would like to see a corporate call center use case. Specifically, a
> user downloads a web page from a corporate web site, clicks a 'call us'
> button and is connected to a gateway server that is controlled by the
> corporation. The communication up to the corporate boundary cannot be
> eavesdropped, but, inside the corporate boundary: 1) the corporation
> can route the call to whoever it wants (meaning that the caller can
> verify that he is connected to the corporation, but is not necessarily
> assured of the identity of the person he is speaking to within the
> corporation) 2) the corporation can eavesdrop/record the call (n.b. this
> is mandatory in financial institutions, and common in most others).
>
> This corresponds to a very common current PSTN use case (except that
> with webRTC the call is more secure up to the corporate boundary). I
> think that corporations will be eager to add webRTC support to their
> call centers - as long as it doesn't mess up their existing operations
> (call routing, recording, etc.) They will most likely want to put in a
> gateway, and treat it as the webRTC endpoint. Inside the gateway the
> call should look just like one that came in from the PSTN (via a SIP
> trunk or PSTN/SIP gateway.)
>
> I think others have suggested use cases involving outbound calls from
> corporations, but I think that those should probably be treated
> separately.
>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtc=
web-bounces@ietf.org] On Behalf
> Of Ted Hardie
> Sent: Friday, April 27, 2012 12:15 PM
> To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> Subject: [rtcweb] Use Case draft
>
> The chairs would like to ask the working group to focus on the use case
> draft. If you have use cases that need to be added to the document or
> text changes you'd like to suggest, please send them in for discussion
> before May 15th. After this round, we will look toward having a working
> group last call on the document (hopefully before the interim meeting).
>
> regards,
>
> Ted, Magnus, Cullen
> _______________________________________________
> 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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/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">The consumer call center =
shall support &#8220;anonymous&#8221; calling wherein there is no need of s=
pecific identity mechanism for the caller.&nbsp; &nbsp;&nbsp;<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
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<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-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Sunday, April 29, 2012 10:51 PM<br>
<b>To:</b> jim.barnett@genesyslab.com; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Use Case draft<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">I agree t=
hat the corporate call center use case is important.
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt; Date: Fri, 27 Apr 2012 09:35:55 -07=
00<br>
&gt; From: <a href=3D"mailto:Jim.Barnett@genesyslab.com">Jim.Barnett@genesy=
slab.com</a><br>
&gt; To: <a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>; <a h=
ref=3D"mailto:rtcweb@ietf.org">
rtcweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] Use Case draft<br>
&gt; <br>
&gt; I would like to see a corporate call center use case. Specifically, a<=
br>
&gt; user downloads a web page from a corporate web site, clicks a 'call us=
'<br>
&gt; button and is connected to a gateway server that is controlled by the<=
br>
&gt; corporation. The communication up to the corporate boundary cannot be<=
br>
&gt; eavesdropped, but, inside the corporate boundary: 1) the corporation<b=
r>
&gt; can route the call to whoever it wants (meaning that the caller can<br=
>
&gt; verify that he is connected to the corporation, but is not necessarily=
<br>
&gt; assured of the identity of the person he is speaking to within the<br>
&gt; corporation) 2) the corporation can eavesdrop/record the call (n.b. th=
is<br>
&gt; is mandatory in financial institutions, and common in most others). <b=
r>
&gt; <br>
&gt; This corresponds to a very common current PSTN use case (except that<b=
r>
&gt; with webRTC the call is more secure up to the corporate boundary). I<b=
r>
&gt; think that corporations will be eager to add webRTC support to their<b=
r>
&gt; call centers - as long as it doesn't mess up their existing operations=
<br>
&gt; (call routing, recording, etc.) They will most likely want to put in a=
<br>
&gt; gateway, and treat it as the webRTC endpoint. Inside the gateway the<b=
r>
&gt; call should look just like one that came in from the PSTN (via a SIP<b=
r>
&gt; trunk or PSTN/SIP gateway.)<br>
&gt; <br>
&gt; I think others have suggested use cases involving outbound calls from<=
br>
&gt; corporations, but I think that those should probably be treated<br>
&gt; separately. <br>
&gt; <br>
&gt; - Jim <br>
&gt; <br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.o=
rg</a> [<a href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ie=
tf.org</a>] On Behalf<br>
&gt; Of Ted Hardie<br>
&gt; Sent: Friday, April 27, 2012 12:15 PM<br>
&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: [rtcweb] Use Case draft<br>
&gt; <br>
&gt; The chairs would like to ask the working group to focus on the use cas=
e<br>
&gt; draft. If you have use cases that need to be added to the document or<=
br>
&gt; text changes you'd like to suggest, please send them in for discussion=
<br>
&gt; before May 15th. After this round, we will look toward having a workin=
g<br>
&gt; group last call on the document (hopefully before the interim meeting)=
.<br>
&gt; <br>
&gt; regards,<br>
&gt; <br>
&gt; Ted, Magnus, Cullen<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>
&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><o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</body>
</html>

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From pravindran@sonusnet.com  Mon Apr 30 00:01:46 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Justin Uberti <juberti@google.com>, Cullen Jennings <fluffy@iii.ca>
Thread-Topic: SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Justin/Cullen,

Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected b=
ehavior. The exact callflow is as follows:


Browser1-------------------------Browser2(SIP-JSEP gateway)
    |        SDP_OFFER(1)            |  INV with offer1
    |------------------------------->|--->
    |       SDP_PRANSWER(1)          |  183 with answer1
    |<-------------------------------|<---
    |       SDP_OFFER(2)             |   UPDATE with offer2
    |<-------------------------------|<---  =20
    |       SDP_ANSWER(2?)           |
    |--------------------->?

The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in Browser 2 (SIP-JSEP gateway).=20

Thanks
Partha      =20

PS: For simplicity, PRACK message exchange is not chosen in SIP side.
  =20
                      =20
     =20



From magnus.westerlund@ericsson.com  Mon Apr 30 00:16:31 2012
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Hi,

My suggestion is that the use case document should be clarified. And I
think the starting point is that the data channel must have the same
security requirements as RTP media, both are media from the point of the
application.

Cheers

Magnus


On 2012-04-27 18:48, Dan Wing wrote:
>> The chairs would like to ask the working group to focus on the use
>> case draft.  If you have use cases that need to be added to the
>> document or text changes you'd like to suggest, please send them in
>> for discussion before May 15th.  After this round, we will look
>> toward having a working group last call on the document (hopefully
>> before the interim meeting).
> 
> A few comments on draft-ietf-rtcweb-use-cases-and-requirements-07:
> 
> 
> 1. Requirement F20 states:
> 
>    F20  It MUST be possible to protect streams from eavesdropping.
> 
> Consensus in the room during my presentation to RTCWEB at IETF83 was that we
> don't need to support un-encrypted media (RTP) at all, and that all media
> would be SRTP.  Can that be captured in F20 by re-wording, or perhaps in a
> new requirement if we can't reword F20?  If there is a need or desire to
> validate that consensus on list, let's please ask the chairs to do that.
> 
> 
> 2. I noticed there is no requirement that we have a baseline for how SRTP
> media is keyed (although there is a baseline requirement for codecs).  This
> is a critical requirement.  I suggest adding "The browser MUST support a
> baseline SRTP keying mechanism."  We have not reached consensus on that
> keying mechanism, but the requirement is real.
> 
> 
> 3. I see the document restricts its scope to media streams in the
> Introduction with:
> 
>   "The document focuses on requirements related to real-time media
>    streams.  Requirements related to privacy, signalling between the
>    browser and web server etc. are currently not considered."
> 
> However, RTCWEB is also supports a data communication between browsers.  I
> am worried if we do not specify requirements for the data communication we
> will have problems.  I believe the expectation is that if the audio/video
> stream works, that the data communication stream also work.  We need to
> capture requirements for the data communication stream somewhere:
> 
>   - a requirement to support data communication
>   - that the chosen data communication protocol supports multiple streams
> (which is why SCTP was chosen over TCP)
>   - for NAT/firewall traversal of the data communication protocol (which is
> why SCTP-over-UDP was chosen and another reason TCP was not chosen)
>   - for encrypting that data communication session
>   - a requirement for SCTP-over-UDP to work when UDP is blocked (aligning
> with the existing F29 for audio/video)
>   - a requirement to do ICE connectivity checks prior to bringing up DTLS (I
> don't know if that is really a requirement, but I recall it mentioned at the
> RTCWEB interim in Mountain View).  
> 
> Based on the scoping of the draft-ietf-rtcweb-use-cases-and-requirements,
> the omission of the data communication stream is intentional.  If not in
> draft-ietf-rtcweb-use-cases-and-requirements, where can we capture the
> requirements for the data communication stream?
> 
> -d
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From tim@phonefromhere.com  Mon Apr 30 00:41:51 2012
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From: Tim Panton <tim@phonefromhere.com>
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com>
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My experience of this (with phonefromhere.com) is that there is very =
limited demand for=20
pure click-to-call in the browser. Users don't see any benefit.

The benefit comes if the call in tied into the rest of the web session, =
so full anonymity=20
isn't desirable in this case.
The best results come when the call center is also using similar =
in-browser technology=20
and the session can be properly shared.

So while this use-case has some merit, it shouldn't be seen as critical.

Tim.

On 30 Apr 2012, at 05:59, Ravindran, Parthasarathi wrote:

> +1
> =20
> The consumer call center shall support =93anonymous=94 calling wherein =
there is no need of specific identity mechanism for the caller.   =20
> =20
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Bernard Aboba
> Sent: Sunday, April 29, 2012 10:51 PM
> To: jim.barnett@genesyslab.com; rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
> =20
> I agree that the corporate call center use case is important.
>=20
> > Date: Fri, 27 Apr 2012 09:35:55 -0700
> > From: Jim.Barnett@genesyslab.com
> > To: ted.ietf@gmail.com; rtcweb@ietf.org
> > Subject: Re: [rtcweb] Use Case draft
> >=20
> > I would like to see a corporate call center use case. Specifically, =
a
> > user downloads a web page from a corporate web site, clicks a 'call =
us'
> > button and is connected to a gateway server that is controlled by =
the
> > corporation. The communication up to the corporate boundary cannot =
be
> > eavesdropped, but, inside the corporate boundary: 1) the corporation
> > can route the call to whoever it wants (meaning that the caller can
> > verify that he is connected to the corporation, but is not =
necessarily
> > assured of the identity of the person he is speaking to within the
> > corporation) 2) the corporation can eavesdrop/record the call (n.b. =
this
> > is mandatory in financial institutions, and common in most others).=20=

> >=20
> > This corresponds to a very common current PSTN use case (except that
> > with webRTC the call is more secure up to the corporate boundary). I
> > think that corporations will be eager to add webRTC support to their
> > call centers - as long as it doesn't mess up their existing =
operations
> > (call routing, recording, etc.) They will most likely want to put in =
a
> > gateway, and treat it as the webRTC endpoint. Inside the gateway the
> > call should look just like one that came in from the PSTN (via a SIP
> > trunk or PSTN/SIP gateway.)
> >=20
> > I think others have suggested use cases involving outbound calls =
from
> > corporations, but I think that those should probably be treated
> > separately.=20
> >=20
> > - Jim=20
> >=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
> > Of Ted Hardie
> > Sent: Friday, April 27, 2012 12:15 PM
> > To: rtcweb@ietf.org
> > Subject: [rtcweb] Use Case draft
> >=20
> > The chairs would like to ask the working group to focus on the use =
case
> > draft. If you have use cases that need to be added to the document =
or
> > text changes you'd like to suggest, please send them in for =
discussion
> > before May 15th. After this round, we will look toward having a =
working
> > group last call on the document (hopefully before the interim =
meeting).
> >=20
> > regards,
> >=20
> > Ted, Magnus, Cullen
> > _______________________________________________
> > 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html><head><base href=3D"x-msg://471/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; ">My experience of this (with <a =
href=3D"http://phonefromhere.com">phonefromhere.com</a>) is that there =
is very limited demand for&nbsp;<div>pure click-to-call in the browser. =
Users don't see any benefit.</div><div><br></div><div>The benefit comes =
if the call in tied into the rest of the web session, so full =
anonymity&nbsp;</div><div>isn't desirable in this case.</div><div>The =
best results come when the call center is also using similar in-browser =
technology&nbsp;</div><div>and the session can be properly =
shared.</div><div><br></div><div>So while this use-case has some merit, =
it shouldn't be seen as =
critical.</div><div><br></div><div>Tim.</div><div><br><div><div>On 30 =
Apr 2012, at 05:59, Ravindran, Parthasarathi wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-right: 0in; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; margin-top: 0in; margin-bottom: 0.0001pt; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">+1<o:p></o:p></span></div><div style=3D"margin-right: =
0in; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; margin-top: 0in; margin-bottom: 0.0001pt; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; margin-top: 0in; margin-bottom: =
0.0001pt; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">The consumer call center shall =
support =93anonymous=94 calling wherein there is no need of specific =
identity mechanism for the caller.&nbsp; =
&nbsp;&nbsp;<o:p></o:p></span></div><div style=3D"margin-right: 0in; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; margin-top: 0in; margin-bottom: 0.0001pt; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div =
style=3D"border-top-style: none; border-right-style: none; =
border-bottom-style: none; border-width: initial; border-color: initial; =
border-left-style: solid; border-left-color: blue; border-left-width: =
1.5pt; padding-top: 0in; padding-right: 0in; padding-bottom: 0in; =
padding-left: 4pt; "><div><div style=3D"border-right-style: none; =
border-bottom-style: none; border-left-style: none; border-width: =
initial; border-color: initial; border-top-style: solid; =
border-top-color: rgb(181, 196, 223); border-top-width: 1pt; =
padding-top: 3pt; padding-right: 0in; padding-bottom: 0in; padding-left: =
0in; "><div style=3D"margin-right: 0in; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; margin-top: 0in; =
margin-bottom: 0.0001pt; "><b><span style=3D"font-size: 10pt; =
font-family: Tahoma, sans-serif; ">From:</span></b><span =
style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; "><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[mailto:rtcweb-bounces@ietf.org]<span =
class=3D"Apple-converted-space">&nbsp;</span><b>On Behalf Of<span =
class=3D"Apple-converted-space">&nbsp;</span></b>Bernard =
Aboba<br><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Sunday, April 29, 2012 =
10:51 PM<br><b>To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:jim.barnett@genesyslab.com">jim.barnett@genesyslab.com</a>;=
 <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><b>Subject:</b><spa=
n class=3D"Apple-converted-space">&nbsp;</span>Re: [rtcweb] Use Case =
draft<o:p></o:p></span></div></div></div><div style=3D"margin-right: =
0in; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; margin-top: 0in; margin-bottom: 0.0001pt; =
"><o:p>&nbsp;</o:p></div><div><p class=3D"MsoNormal" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; margin-top: 0in; margin-bottom: =
12pt; "><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; =
">I agree that the corporate call center use case is =
important.<o:p></o:p></span></p><div><div style=3D"margin-right: 0in; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; margin-top: 0in; margin-bottom: 0.0001pt; "><span =
style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; ">&gt; Date: =
Fri, 27 Apr 2012 09:35:55 -0700<br>&gt; From:<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:Jim.Barnett@genesyslab.com" style=3D"color: blue; =
text-decoration: underline; ">Jim.Barnett@genesyslab.com</a><br>&gt; =
To:<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:ted.ietf@gmail.com" style=3D"color: blue; =
text-decoration: underline; ">ted.ietf@gmail.com</a>;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">rtcweb@ietf.org</a><br>&gt; Subject: Re: [rtcweb] Use Case =
draft<br>&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>&gt; =
I would like to see a corporate call center use case. Specifically, =
a<br>&gt; user downloads a web page from a corporate web site, clicks a =
'call us'<br>&gt; button and is connected to a gateway server that is =
controlled by the<br>&gt; corporation. The communication up to the =
corporate boundary cannot be<br>&gt; eavesdropped, but, inside the =
corporate boundary: 1) the corporation<br>&gt; can route the call to =
whoever it wants (meaning that the caller can<br>&gt; verify that he is =
connected to the corporation, but is not necessarily<br>&gt; assured of =
the identity of the person he is speaking to within the<br>&gt; =
corporation) 2) the corporation can eavesdrop/record the call (n.b. =
this<br>&gt; is mandatory in financial institutions, and common in most =
others).<span class=3D"Apple-converted-space">&nbsp;</span><br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; This corresponds =
to a very common current PSTN use case (except that<br>&gt; with webRTC =
the call is more secure up to the corporate boundary). I<br>&gt; think =
that corporations will be eager to add webRTC support to their<br>&gt; =
call centers - as long as it doesn't mess up their existing =
operations<br>&gt; (call routing, recording, etc.) They will most likely =
want to put in a<br>&gt; gateway, and treat it as the webRTC endpoint. =
Inside the gateway the<br>&gt; call should look just like one that came =
in from the PSTN (via a SIP<br>&gt; trunk or PSTN/SIP =
gateway.)<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; I think others =
have suggested use cases involving outbound calls from<br>&gt; =
corporations, but I think that those should probably be treated<br>&gt; =
separately.<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; - Jim<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; -----Original =
Message-----<br>&gt; From:<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb-bounces@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">rtcweb-bounces@ietf.org</a><span =
class=3D"Apple-converted-space">&nbsp;</span>[<a =
href=3D"mailto:rtcweb-bounces@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">mailto:rtcweb-bounces@ietf.org</a>] On =
Behalf<br>&gt; Of Ted Hardie<br>&gt; Sent: Friday, April 27, 2012 12:15 =
PM<br>&gt; To:<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">rtcweb@ietf.org</a><br>&gt; Subject: [rtcweb] Use Case =
draft<br>&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>&gt; =
The chairs would like to ask the working group to focus on the use =
case<br>&gt; draft. If you have use cases that need to be added to the =
document or<br>&gt; text changes you'd like to suggest, please send them =
in for discussion<br>&gt; before May 15th. After this round, we will =
look toward having a working<br>&gt; group last call on the document =
(hopefully before the interim meeting).<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; =
regards,<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; Ted, Magnus, =
Cullen<br>&gt; _______________________________________________<br>&gt; =
rtcweb mailing list<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">rtcweb@ietf.org</a><br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>&gt; =
_______________________________________________<br>&gt; rtcweb mailing =
list<br>&gt;<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">rtcweb@ietf.org</a><br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></div>=
</div></div></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</div></span></blockquote></div><br></div></body>=
</html>=

--Apple-Mail=_9CAE8C61-4AEC-4689-B9C1-82F36390B372--

From harald@alvestrand.no  Mon Apr 30 01:31:50 2012
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I have formatted the proposal I made some time ago into an I-D format 
and added some supporting material, including a lot of points based on 
Magnus' mail.
This is strictly an individual contribution; the chairs will determine 
how to progress with reaching and documenting consensus on this issue.

Comments welcome!

     Harald

-------- Original Message --------
Subject: 	New Version Notification for 
draft-alvestrand-rtcweb-resolution-00.txt
Date: 	Mon, 30 Apr 2012 01:17:09 -0700
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no



A new version of I-D, draft-alvestrand-rtcweb-resolution-00.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-alvestrand-rtcweb-resolution
Revision:	 00
Title:		 RTCWEB Resolution Negotiation
Creation date:	 2012-04-30
WG ID:		 Individual Submission
Number of pages: 10

Abstract:
    This draft offers a proposal for a fragment of the SDP usage rules
    for RTCWEB: Requirements for supporting resolution negotiation.

    It proposes to use SDP both for initial and within-call resolution
    configuration, and suggests that COP should be mentioned as an
    optional, not mandatory, mechanism.





The IETF Secretariat



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    I have formatted the proposal I made some time ago into an I-D
    format and added some supporting material, including a lot of points
    based on Magnus' mail.<br>
    This is strictly an individual contribution; the chairs will
    determine how to progress with reaching and documenting consensus on
    this issue.<br>
    <br>
    Comments welcome!<br>
    <br>
    Â Â Â  Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>New Version Notification for
            draft-alvestrand-rtcweb-resolution-00.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Mon, 30 Apr 2012 01:17:09 -0700</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
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    <br>
    <br>
    <pre>A new version of I-D, draft-alvestrand-rtcweb-resolution-00.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-alvestrand-rtcweb-resolution
Revision:	 00
Title:		 RTCWEB Resolution Negotiation
Creation date:	 2012-04-30
WG ID:		 Individual Submission
Number of pages: 10

Abstract:
   This draft offers a proposal for a fragment of the SDP usage rules
   for RTCWEB: Requirements for supporting resolution negotiation.

   It proposes to use SDP both for initial and within-call resolution
   configuration, and suggests that COP should be mentioned as an
   optional, not mandatory, mechanism.


                                                                                  


The IETF Secretariat

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From pravindran@sonusnet.com  Mon Apr 30 01:42:34 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] Use Case draft
Thread-Index: AQHNJJD6fraw229Zx0W4DjUXXLWGX5augkyAgAMxP4CAAR3d8P//0qUAgABiT3A=
Date: Mon, 30 Apr 2012 08:42:25 +0000
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com>
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Subject: Re: [rtcweb] Use Case draft
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Tim,

My experience is different. Click-to-call is attractive in case of toll-fre=
e number in the site. WebRTC provides complete free call without any toll.

The registered users should not mandated as it is not generically applicabl=
e for all sites. In case of  camera selling site wherein the anonymous cust=
omer wish to get the details about the product wherein the sites will not m=
andate for registration before calling the shop as today, these sites provi=
de their phone number for the customer to contact without any identity.  I =
really don't know why some identity MUST be mandated  for calling those Web=
RTC sites. The customer support folks will get the relevant information dur=
ing the course of the call :)

Having spammer@gmail.com<mailto:spammer@gmail.com> as a identity is not goi=
ng to much helpful in these sites whereas captcha will confirm the caller a=
s a human.

Thanks
Partha

From: Tim Panton [mailto:tim@phonefromhere.com]
Sent: Monday, April 30, 2012 1:12 PM
To: Ravindran, Parthasarathi
Cc: Bernard Aboba; jim.barnett@genesyslab.com; rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

My experience of this (with phonefromhere.com<http://phonefromhere.com>) is=
 that there is very limited demand for
pure click-to-call in the browser. Users don't see any benefit.

The benefit comes if the call in tied into the rest of the web session, so =
full anonymity
isn't desirable in this case.
The best results come when the call center is also using similar in-browser=
 technology
and the session can be properly shared.

So while this use-case has some merit, it shouldn't be seen as critical.

Tim.

On 30 Apr 2012, at 05:59, Ravindran, Parthasarathi wrote:


+1

The consumer call center shall support "anonymous" calling wherein there is=
 no need of specific identity mechanism for the caller.

From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtcwe=
b-bounces@ietf.org] On Behalf Of Bernard Aboba
Sent: Sunday, April 29, 2012 10:51 PM
To: jim.barnett@genesyslab.com<mailto:jim.barnett@genesyslab.com>; rtcweb@i=
etf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Use Case draft

I agree that the corporate call center use case is important.
> Date: Fri, 27 Apr 2012 09:35:55 -0700
> From: Jim.Barnett@genesyslab.com<mailto:Jim.Barnett@genesyslab.com>
> To: ted.ietf@gmail.com<mailto:ted.ietf@gmail.com>; rtcweb@ietf.org<mailto=
:rtcweb@ietf.org>
> Subject: Re: [rtcweb] Use Case draft
>
> I would like to see a corporate call center use case. Specifically, a
> user downloads a web page from a corporate web site, clicks a 'call us'
> button and is connected to a gateway server that is controlled by the
> corporation. The communication up to the corporate boundary cannot be
> eavesdropped, but, inside the corporate boundary: 1) the corporation
> can route the call to whoever it wants (meaning that the caller can
> verify that he is connected to the corporation, but is not necessarily
> assured of the identity of the person he is speaking to within the
> corporation) 2) the corporation can eavesdrop/record the call (n.b. this
> is mandatory in financial institutions, and common in most others).
>
> This corresponds to a very common current PSTN use case (except that
> with webRTC the call is more secure up to the corporate boundary). I
> think that corporations will be eager to add webRTC support to their
> call centers - as long as it doesn't mess up their existing operations
> (call routing, recording, etc.) They will most likely want to put in a
> gateway, and treat it as the webRTC endpoint. Inside the gateway the
> call should look just like one that came in from the PSTN (via a SIP
> trunk or PSTN/SIP gateway.)
>
> I think others have suggested use cases involving outbound calls from
> corporations, but I think that those should probably be treated
> separately.
>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtc=
web-bounces@ietf.org] On Behalf
> Of Ted Hardie
> Sent: Friday, April 27, 2012 12:15 PM
> To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> Subject: [rtcweb] Use Case draft
>
> The chairs would like to ask the working group to focus on the use case
> draft. If you have use cases that need to be added to the document or
> text changes you'd like to suggest, please send them in for discussion
> before May 15th. After this round, we will look toward having a working
> group last call on the document (hopefully before the interim meeting).
>
> regards,
>
> Ted, Magnus, Cullen
> _______________________________________________
> 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
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Tim,<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">My experience is differen=
t. Click-to-call is attractive in case of toll-free number in the site. Web=
RTC provides complete free call without any toll.<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">The registered users shou=
ld not mandated as it is not generically applicable for all sites. In case =
of &nbsp;camera selling site wherein the anonymous customer wish
 to get the details about the product wherein the sites will not mandate fo=
r registration before calling the shop as today, these sites provide their =
phone number for the customer to contact without any identity. &nbsp;I real=
ly don&#8217;t know why some identity MUST
 be mandated &nbsp;for calling those WebRTC sites. The customer support fol=
ks will get the relevant information during the course of the call
</span><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1F497D"=
>J</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">
<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">Having
<a href=3D"mailto:spammer@gmail.com">spammer@gmail.com</a> as a identity is=
 not going to much helpful in these sites whereas captcha will confirm the =
caller as a human.
<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">Thanks<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">Partha<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>
<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;"> Tim Pant=
on [mailto:tim@phonefromhere.com]
<br>
<b>Sent:</b> Monday, April 30, 2012 1:12 PM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> Bernard Aboba; jim.barnett@genesyslab.com; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Use Case draft<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">My experience of this (with <a href=3D"http://phonef=
romhere.com">
phonefromhere.com</a>) is that there is very limited demand for&nbsp;<o:p><=
/o:p></p>
<div>
<p class=3D"MsoNormal">pure click-to-call in the browser. Users don't see a=
ny benefit.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">The benefit comes if the call in tied into the rest =
of the web session, so full anonymity&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">isn't desirable in this case.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">The best results come when the call center is also u=
sing similar in-browser technology&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">and the session can be properly shared.<o:p></o:p></=
p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">So while this use-case has some merit, it shouldn't =
be seen as critical.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Tim.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On 30 Apr 2012, at 05:59, Ravindran, Parthasarathi w=
rote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></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">&#43;1</span><o:p></o:p><=
/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">&nbsp;</span><o:p></o:p><=
/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">The consumer call center =
shall support &#8220;anonymous&#8221; calling wherein there is no need of s=
pecific identity mechanism for the caller.&nbsp; &nbsp;&nbsp;</span><o:p></=
o:p></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">&nbsp;</span><o:p></o:p><=
/p>
</div>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt;border-width:initial;border-color:initial">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in;border-width:initial;border-color:initial">
<div>
<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 class=3D"apple-=
converted-space"><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&q=
uot;,&quot;sans-serif&quot;">&nbsp;</span></span><span style=3D"font-size:1=
0.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"><a href=3D"mai=
lto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
 [<a href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org=
</a>]<span class=3D"apple-converted-space">&nbsp;</span><b>On Behalf Of<spa=
n class=3D"apple-converted-space">&nbsp;</span></b>Bernard Aboba<br>
<b>Sent:</b><span class=3D"apple-converted-space">&nbsp;</span>Sunday, Apri=
l 29, 2012 10:51 PM<br>
<b>To:</b><span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mai=
lto:jim.barnett@genesyslab.com">jim.barnett@genesyslab.com</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b><span class=3D"apple-converted-space">&nbsp;</span>Re: [rtcw=
eb] Use Case draft</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">I agree t=
hat the corporate call center use case is important.</span><o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt; Date: Fri, 27 Apr 2012 09:35:55 -07=
00<br>
&gt; From:<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mai=
lto:Jim.Barnett@genesyslab.com">Jim.Barnett@genesyslab.com</a><br>
&gt; To:<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mailt=
o:ted.ietf@gmail.com">ted.ietf@gmail.com</a>;<span class=3D"apple-converted=
-space">&nbsp;</span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>=
<br>
&gt; Subject: Re: [rtcweb] Use Case draft<br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; I would like to see a corporate call center use case. Specifically, a<=
br>
&gt; user downloads a web page from a corporate web site, clicks a 'call us=
'<br>
&gt; button and is connected to a gateway server that is controlled by the<=
br>
&gt; corporation. The communication up to the corporate boundary cannot be<=
br>
&gt; eavesdropped, but, inside the corporate boundary: 1) the corporation<b=
r>
&gt; can route the call to whoever it wants (meaning that the caller can<br=
>
&gt; verify that he is connected to the corporation, but is not necessarily=
<br>
&gt; assured of the identity of the person he is speaking to within the<br>
&gt; corporation) 2) the corporation can eavesdrop/record the call (n.b. th=
is<br>
&gt; is mandatory in financial institutions, and common in most others).<sp=
an class=3D"apple-converted-space">&nbsp;</span><br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; This corresponds to a very common current PSTN use case (except that<b=
r>
&gt; with webRTC the call is more secure up to the corporate boundary). I<b=
r>
&gt; think that corporations will be eager to add webRTC support to their<b=
r>
&gt; call centers - as long as it doesn't mess up their existing operations=
<br>
&gt; (call routing, recording, etc.) They will most likely want to put in a=
<br>
&gt; gateway, and treat it as the webRTC endpoint. Inside the gateway the<b=
r>
&gt; call should look just like one that came in from the PSTN (via a SIP<b=
r>
&gt; trunk or PSTN/SIP gateway.)<br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; I think others have suggested use cases involving outbound calls from<=
br>
&gt; corporations, but I think that those should probably be treated<br>
&gt; separately.<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; - Jim<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; -----Original Message-----<br>
&gt; From:<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mai=
lto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a><span class=3D"appl=
e-converted-space">&nbsp;</span>[<a href=3D"mailto:rtcweb-bounces@ietf.org"=
>mailto:rtcweb-bounces@ietf.org</a>] On Behalf<br>
&gt; Of Ted Hardie<br>
&gt; Sent: Friday, April 27, 2012 12:15 PM<br>
&gt; To:<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mailt=
o:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: [rtcweb] Use Case draft<br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; The chairs would like to ask the working group to focus on the use cas=
e<br>
&gt; draft. If you have use cases that need to be added to the document or<=
br>
&gt; text changes you'd like to suggest, please send them in for discussion=
<br>
&gt; before May 15th. After this round, we will look toward having a workin=
g<br>
&gt; group last call on the document (hopefully before the interim meeting)=
.<br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; regards,<br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><br>
&gt; Ted, Magnus, Cullen<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mailto:rt=
cweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"https://w=
ww.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/=
rtcweb</a><br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mailto:rt=
cweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"https://w=
ww.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/=
rtcweb</a></span><o:p></o:p></p>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<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.o=
rg/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

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Subject: Re: [rtcweb] Use Case draft (privacy)
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On 4/27/12 6:35 PM, Jim Barnett wrote:
> I would like to see a corporate call center use case.  Specifically, a
> user downloads a web page from a corporate web site, clicks a 'call us'
> button and is connected to a gateway server that is controlled by the
> corporation.  The communication up to the corporate boundary cannot be
> eavesdropped, but, inside the corporate boundary:  1) the corporation
> can route the call to whoever it wants (meaning that the caller can
> verify that he is connected to the corporation, but is not necessarily
> assured of the identity of the person he is speaking to within the
> corporation) 2) the corporation can eavesdrop/record the call (n.b. this
> is mandatory in financial institutions, and common in most others). 

In that case, from a privacy perspective, it's HIGHLY RELEVANT to show
in the UI to the user that the call does it's encrypted up to a gateway
and not up to another peer.

Please get back the thread on end-to-end vs end-to-site security.

The user *must known and be aware* if a call is secured between two
peers or if it's not secured up to a gateway (and who control such a
gateway).

Fabio

From pravindran@sonusnet.com  Mon Apr 30 02:12:04 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Use Case draft (privacy)
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Fabio,

Please note that gateway shall acts as web-browser and compliance to RTCWeb=
 specifications. Here, WebRTC session is between general-purpose web-browse=
r like IE, Chrome in the customer side and customized web-browser in the si=
te side.=20

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Fabio Pietrosanti (naif)
>Sent: Monday, April 30, 2012 2:35 PM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Use Case draft (privacy)
>
>On 4/27/12 6:35 PM, Jim Barnett wrote:
>> I would like to see a corporate call center use case.  Specifically, a
>> user downloads a web page from a corporate web site, clicks a 'call
>us'
>> button and is connected to a gateway server that is controlled by the
>> corporation.  The communication up to the corporate boundary cannot be
>> eavesdropped, but, inside the corporate boundary:  1) the corporation
>> can route the call to whoever it wants (meaning that the caller can
>> verify that he is connected to the corporation, but is not necessarily
>> assured of the identity of the person he is speaking to within the
>> corporation) 2) the corporation can eavesdrop/record the call (n.b.
>> this is mandatory in financial institutions, and common in most
>others).
>
>In that case, from a privacy perspective, it's HIGHLY RELEVANT to show
>in the UI to the user that the call does it's encrypted up to a gateway
>and not up to another peer.
>
>Please get back the thread on end-to-end vs end-to-site security.
>
>The user *must known and be aware* if a call is secured between two
>peers or if it's not secured up to a gateway (and who control such a
>gateway).
>
>Fabio
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Mon Apr 30 02:13:47 2012
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On 2012-04-25 18:13, Harald Alvestrand wrote:
> On 04/25/2012 04:35 PM, Magnus Westerlund wrote:
>> On 2012-04-25 05:20, Justin Uberti wrote:
>>> On Tue, Apr 24, 2012 at 10:25 AM, Magnus Westerlund
>>> <magnus.westerlund@ericsson.com<mailto:magnus.westerlund@ericsson.com>>
>>> wrote:
>>>
>>>      A) Using COP and using SDP based signaling for the dynamic changes are
>>>      two quite different models in relation to how the interaction happens.
>>>
>>>      For COP this all happens in the browser, normally initiated by the
>>>      browser's own determination that a COP request or notification is
>>>      needed. Harald's proposal appears to require that the JS initiate a
>>>      renegotiation. This puts a requirement on the implementation to listen
>>>      to the correct callbacks to know when changes happens, such as window
>>>      resize. To my knowledge there are not yet any proposals for how the
>>>      browser can initiate a JSEP renegotiation.
>>>
>>>
>>> Maybe I misunderstand what you are saying, but the application will
>>> definitely know when the browser size changes, and can then trigger a
>>> JSEP renegotiation by calling createOffer and shipping it off.
>> Yes, they can but that requires them to write code to do this. With COP
>> the browser can do it and it works well for all applications, not only
>> the ones that have implementors that are aware of the need to do this.
>>
>> I am proposing a mechanism that makes the default behavior if you care
>> nothing about controlling or optimizing this behavior in your JS to be
>> good. In addition JS that wants to control it can affect the behavior
>> through generic mechanisms, such as controlling the video element. And
>> if clear need is shown and some functionality is missing, W3C can create
>> an API to provide explicit control.
> It's not at all clear to me that the browser being in control is right 
> for all cases.
> Renegotiating has a cost, and some of the costs are known only to the 
> application.
> 
> We may possibly handle this via constraints like 
> "VideoEnumRenegotiationTendency" or something like that - when you get 
> the proposal together more, I'd like to see the controls you're proposing.
>>> I have the opposite concern - how can the browser know when the
>>> application makes a change, for instance to display a participant in a
>>> large view versus a small view? This may be possible if using a<video/>
>>> for display, but I don't think it will be possible when using WebGL for
>>> rendering, where the size of the view will be dictated only by the
>>> geometry of the scene.
>> What I understand from my colleagues is that all rendering of video have
>> to go through a video element, even if not displayed directly. Thus the
>> application can control the size of the video element used and possibly
>> consumed by WebGL to be included in rendering. Thus the application will
>> have control over this even when COP is being used.
> We have on the "list of things that need doing" the connection of a 
> mediastream directly to a canvas, a WebGL element or being captured to a 
> file, so don't gamble too much on the video element to stay in the path.
> 
> Example code for capturing video to a canvas (taken from a real app):
> 
>         video.src = webkitURL.createObjectURL(s);
>         canvas.getContext("2d").drawImage(video, 0, 0, 300, 300, 0, 0, 
> 300, 300);

According to my Colleague Per-Erik you can use video.width and
video.height to set the resolution the video element should have.

see
http://dev.w3.org/html5/spec/the-video-element.html#the-video-element

Thus the code you presented should be something like.

video.src = webkitURL.createObjectURL(s);
video.width = 300;
video.height = 300;
canvas.getContext("2d").drawImage(video, 0, 0, 300, 300);

That would set the video objects size to a 300, 300 video resolution.
The drawImage call I have removed the part which selects a cropping of
the source video to automatically show the full video in the specified
area.

The adaptation of the video will of course not happen instantly so until
that happens the video will be scaled to the specified width and height.
And the actual delivered resolution will depend on the COP request and
what the sender support.

>> I agree that it needs to be clearly specified. But COP appears to be
>> less prone to complete failures by using target values. Then at least it
>> will receive video that can be used, even if not optimal. imageattr can
>> result in failures in the negotiation requiring at minimal a second
>> round potentially no agreed on acceptable resolution then that is
>> clearly sub-optimal. That is even without considering involving JS
>> mainipulating this.
>
> I certainly hope that we can avoid the need for SDP manipulation of JS 
> in all of these proposals.
> 

Looking forward to a bit more detailed explanation of how that is
accomplished in relation to the API.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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 andrew.hutton@siemens-enterprise.com  Mon Apr 30 04:38:22 2012
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To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi,

I also agree that calling a corporate call center is an interesting use cas=
e and I think we discussed it in the past in the context of media recording=
 and identity.

When calling your bank it is only necessary to know that you are talking to=
 a representative of mybank.com the bank certainly does not want to release=
 the identity of the call centre agent you are talking to. From the consume=
r perspective they want to be sure that the call really did reach mybank.co=
m but from there on they have to trust the bank with their information and =
really this is no different for audio/video than it is for text entered on =
a web page and of course the bank will record the audio.

In this scenario it is really not useful to confuse the calling user with i=
ndications of whether the bank encrypts the audio/video when handling it in=
ternally or not it is enough to know that they have a secure connection to =
their bank.=20

Regards
Andy =20



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ravindran, Parthasarathi
> Sent: 30 April 2012 10:12
> To: Fabio Pietrosanti (naif); rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft (privacy)
>=20
> Fabio,
>=20
> Please note that gateway shall acts as web-browser and compliance to
> RTCWeb specifications. Here, WebRTC session is between general-purpose
> web-browser like IE, Chrome in the customer side and customized web-
> browser in the site side.
>=20
> Thanks
> Partha
>=20
> >-----Original Message-----
> >From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf
> >Of Fabio Pietrosanti (naif)
> >Sent: Monday, April 30, 2012 2:35 PM
> >To: rtcweb@ietf.org
> >Subject: Re: [rtcweb] Use Case draft (privacy)
> >
> >On 4/27/12 6:35 PM, Jim Barnett wrote:
> >> I would like to see a corporate call center use case.  Specifically,
> a
> >> user downloads a web page from a corporate web site, clicks a 'call
> >us'
> >> button and is connected to a gateway server that is controlled by
> the
> >> corporation.  The communication up to the corporate boundary cannot
> be
> >> eavesdropped, but, inside the corporate boundary:  1) the
> corporation
> >> can route the call to whoever it wants (meaning that the caller can
> >> verify that he is connected to the corporation, but is not
> necessarily
> >> assured of the identity of the person he is speaking to within the
> >> corporation) 2) the corporation can eavesdrop/record the call (n.b.
> >> this is mandatory in financial institutions, and common in most
> >others).
> >
> >In that case, from a privacy perspective, it's HIGHLY RELEVANT to show
> >in the UI to the user that the call does it's encrypted up to a
> gateway
> >and not up to another peer.
> >
> >Please get back the thread on end-to-end vs end-to-site security.
> >
> >The user *must known and be aware* if a call is secured between two
> >peers or if it's not secured up to a gateway (and who control such a
> >gateway).
> >
> >Fabio
> >_______________________________________________
> >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

From stefan.lk.hakansson@ericsson.com  Mon Apr 30 04:41:53 2012
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Subject: Re: [rtcweb] Use Case draft
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On 04/27/2012 06:43 PM, Timothy B. Terriberry wrote:
> Ted Hardie wrote:
>> The chairs would like to ask the working group to focus on the use
>> case draft.  If you have use cases that need to be added to the
>> document or text changes you'd like to suggest, please send them in
>
> I proposed the following use-case back in February, but there wasn't
> much discussion on actually adding it to the document:
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg03357.html
>
> Let me know if the WG would like to proceed with something like this.

I think adding another small derivative of the simple video chat (this 
one with peer-to-peer file transfer added) makes a lot of sense. For 
one, we get a use case that requires reliable data (now we only have a 
req for "short latency datagram" which sound like unreliable to me); in 
addition we get a requirement for the data channel API to be able to use 
blobs (as defined in the File API W3C rec) as input/output.

A third good requirement that can be derived (if we want to) is the 
ability to prioritize data in relation to audio/video.

So, I think we should add it.

Stefan

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


From Jim.Barnett@genesyslab.com  Mon Apr 30 06:08:01 2012
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Tim Panton" <tim@phonefromhere.com>, "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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I think that there are two different issues here. =20

=20

1)      In many cases the call center would like to have information
about the caller's web session:  what page he is on, browsing history
information, etc.  This can be passed in the data channel.

2)      However, this does not require that the caller be authenticated
(i.e. identified as a specific individual).  The call center wants  to
know that the caller has been looking at, e.g., warranty info for
cameras on their site, but doesn't  particularly need to know the
caller's name (they can get that during the call if they need it.)  =20

=20

-          Jim

=20

From: Tim Panton [mailto:tim@phonefromhere.com]=20
Sent: Monday, April 30, 2012 3:42 AM
To: Ravindran, Parthasarathi
Cc: Bernard Aboba; Jim Barnett; rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

=20

My experience of this (with phonefromhere.com) is that there is very
limited demand for=20

pure click-to-call in the browser. Users don't see any benefit.

=20

The benefit comes if the call in tied into the rest of the web session,
so full anonymity=20

isn't desirable in this case.

The best results come when the call center is also using similar
in-browser technology=20

and the session can be properly shared.

=20

So while this use-case has some merit, it shouldn't be seen as critical.

=20

Tim.

=20

On 30 Apr 2012, at 05:59, Ravindran, Parthasarathi wrote:





+1

=20

The consumer call center shall support "anonymous" calling wherein there
is no need of specific identity mechanism for the caller.   =20

=20

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Bernard Aboba
Sent: Sunday, April 29, 2012 10:51 PM
To: jim.barnett@genesyslab.com; rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

=20

I agree that the corporate call center use case is important.

> Date: Fri, 27 Apr 2012 09:35:55 -0700
> From: Jim.Barnett@genesyslab.com
> To: ted.ietf@gmail.com; rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
>=20
> I would like to see a corporate call center use case. Specifically, a
> user downloads a web page from a corporate web site, clicks a 'call
us'
> button and is connected to a gateway server that is controlled by the
> corporation. The communication up to the corporate boundary cannot be
> eavesdropped, but, inside the corporate boundary: 1) the corporation
> can route the call to whoever it wants (meaning that the caller can
> verify that he is connected to the corporation, but is not necessarily
> assured of the identity of the person he is speaking to within the
> corporation) 2) the corporation can eavesdrop/record the call (n.b.
this
> is mandatory in financial institutions, and common in most others).=20
>=20
> This corresponds to a very common current PSTN use case (except that
> with webRTC the call is more secure up to the corporate boundary). I
> think that corporations will be eager to add webRTC support to their
> call centers - as long as it doesn't mess up their existing operations
> (call routing, recording, etc.) They will most likely want to put in a
> gateway, and treat it as the webRTC endpoint. Inside the gateway the
> call should look just like one that came in from the PSTN (via a SIP
> trunk or PSTN/SIP gateway.)
>=20
> I think others have suggested use cases involving outbound calls from
> corporations, but I think that those should probably be treated
> separately.=20
>=20
> - Jim=20
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf
> Of Ted Hardie
> Sent: Friday, April 27, 2012 12:15 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Use Case draft
>=20
> The chairs would like to ask the working group to focus on the use
case
> draft. If you have use cases that need to be added to the document or
> text changes you'd like to suggest, please send them in for discussion
> before May 15th. After this round, we will look toward having a
working
> group last call on the document (hopefully before the interim
meeting).
>=20
> regards,
>=20
> Ted, Magnus, Cullen
> _______________________________________________
> 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

_______________________________________________
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'>I think that there are two different issues here.&nbsp; =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
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D'><span style=3D'mso-list:Ignore'>1)<span style=3D'font:7.0pt "Times =
New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>In many cases the call center would like to have information about =
the caller&#8217;s web session:&nbsp; what page he is on, browsing =
history information, etc. &nbsp;This can be passed in the data =
channel.<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l1 level1 lfo1'><![if =
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D'><span style=3D'mso-list:Ignore'>2)<span style=3D'font:7.0pt "Times =
New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>However, this does not require that the caller be authenticated (i.e. =
identified as a specific individual).&nbsp; The call center wants&nbsp; =
to know that the caller has been looking at, e.g., warranty info for =
cameras on their site, but doesn&#8217;t&nbsp; particularly need to know =
the caller&#8217;s name (they can get that during the call if they need =
it.)&nbsp;&nbsp; <o:p></o:p></span></p><p class=3DMsoNormal><span =
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D'><span style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Jim<o:p></o:p></span></p><p class=3DMsoNormal><span =
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D'><o:p>&nbsp;</o:p></span></p><div><div =
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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"'> =
Tim Panton [mailto:tim@phonefromhere.com] <br><b>Sent:</b> Monday, April =
30, 2012 3:42 AM<br><b>To:</b> Ravindran, Parthasarathi<br><b>Cc:</b> =
Bernard Aboba; Jim Barnett; rtcweb@ietf.org<br><b>Subject:</b> Re: =
[rtcweb] Use Case draft<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>My =
experience of this (with <a =
href=3D"http://phonefromhere.com">phonefromhere.com</a>) is that there =
is very limited demand for&nbsp;<o:p></o:p></p><div><p =
class=3DMsoNormal>pure click-to-call in the browser. Users don't see any =
benefit.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>The benefit comes if the call in tied into the rest of =
the web session, so full anonymity&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>isn't desirable in this =
case.<o:p></o:p></p></div><div><p class=3DMsoNormal>The best results =
come when the call center is also using similar in-browser =
technology&nbsp;<o:p></o:p></p></div><div><p class=3DMsoNormal>and the =
session can be properly shared.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>So while this use-case has some merit, it shouldn't be =
seen as critical.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Tim.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=3DMsoNormal>On =
30 Apr 2012, at 05:59, Ravindran, Parthasarathi =
wrote:<o:p></o:p></p></div><p =
class=3DMsoNormal><br><br><o:p></o:p></p><div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>+1</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The consumer call center shall support &#8220;anonymous&#8221; =
calling wherein there is no need of specific identity mechanism for the =
caller.&nbsp; &nbsp;&nbsp;</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p></div><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt;border-width:initial;border-color:initial'><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in;border-width:initial;border-color:initial'><div><p =
class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</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"'><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=
>]<span class=3Dapple-converted-space>&nbsp;</span><b>On Behalf Of<span =
class=3Dapple-converted-space>&nbsp;</span></b>Bernard =
Aboba<br><b>Sent:</b><span =
class=3Dapple-converted-space>&nbsp;</span>Sunday, April 29, 2012 10:51 =
PM<br><b>To:</b><span class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:jim.barnett@genesyslab.com">jim.barnett@genesyslab.com</a>=
; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><b>Subject:</b><sp=
an class=3Dapple-converted-space>&nbsp;</span>Re: [rtcweb] Use Case =
draft</span><o:p></o:p></p></div></div></div><div><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>I agree =
that the corporate call center use case is =
important.</span><o:p></o:p></p><div><div><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>&gt; Date: =
Fri, 27 Apr 2012 09:35:55 -0700<br>&gt; From:<span =
class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:Jim.Barnett@genesyslab.com">Jim.Barnett@genesyslab.com</a>=
<br>&gt; To:<span class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>;<span =
class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; Subject: Re: =
[rtcweb] Use Case draft<br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; I would like to see =
a corporate call center use case. Specifically, a<br>&gt; user downloads =
a web page from a corporate web site, clicks a 'call us'<br>&gt; button =
and is connected to a gateway server that is controlled by the<br>&gt; =
corporation. The communication up to the corporate boundary cannot =
be<br>&gt; eavesdropped, but, inside the corporate boundary: 1) the =
corporation<br>&gt; can route the call to whoever it wants (meaning that =
the caller can<br>&gt; verify that he is connected to the corporation, =
but is not necessarily<br>&gt; assured of the identity of the person he =
is speaking to within the<br>&gt; corporation) 2) the corporation can =
eavesdrop/record the call (n.b. this<br>&gt; is mandatory in financial =
institutions, and common in most others).<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; This corresponds to =
a very common current PSTN use case (except that<br>&gt; with webRTC the =
call is more secure up to the corporate boundary). I<br>&gt; think that =
corporations will be eager to add webRTC support to their<br>&gt; call =
centers - as long as it doesn't mess up their existing =
operations<br>&gt; (call routing, recording, etc.) They will most likely =
want to put in a<br>&gt; gateway, and treat it as the webRTC endpoint. =
Inside the gateway the<br>&gt; call should look just like one that came =
in from the PSTN (via a SIP<br>&gt; trunk or PSTN/SIP =
gateway.)<br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; I think others have =
suggested use cases involving outbound calls from<br>&gt; corporations, =
but I think that those should probably be treated<br>&gt; =
separately.<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; - Jim<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; -----Original =
Message-----<br>&gt; From:<span =
class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a><span =
class=3Dapple-converted-space>&nbsp;</span>[<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] On Behalf<br>&gt; Of Ted Hardie<br>&gt; Sent: Friday, April 27, 2012 =
12:15 PM<br>&gt; To:<span class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; Subject: =
[rtcweb] Use Case draft<br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; The chairs would =
like to ask the working group to focus on the use case<br>&gt; draft. If =
you have use cases that need to be added to the document or<br>&gt; text =
changes you'd like to suggest, please send them in for =
discussion<br>&gt; before May 15th. After this round, we will look =
toward having a working<br>&gt; group last call on the document =
(hopefully before the interim meeting).<br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; =
regards,<br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><br>&gt; Ted, Magnus, =
Cullen<br>&gt; _______________________________________________<br>&gt; =
rtcweb mailing list<br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><br>&gt; =
_______________________________________________<br>&gt; rtcweb mailing =
list<br>&gt;<span class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt;<span =
class=3Dapple-converted-space>&nbsp;</span><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a></span><o:p></o:p></p></div></div></div></di=
v><p class=3DMsoNormal><span =
style=3D'font-size:13.5pt;font-family:"Helvetica","sans-serif"'>_________=
______________________________________<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.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
------_=_NextPart_001_01CD26D2.42D17C29--

From tim@phonefromhere.com  Mon Apr 30 06:54:48 2012
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From: Tim Panton <tim@phonefromhere.com>
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Date: Mon, 30 Apr 2012 14:54:50 +0100
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com> <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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On 30 Apr 2012, at 09:42, Ravindran, Parthasarathi wrote:

> Tim,
> =20
> My experience is different. Click-to-call is attractive in case of =
toll-free number in the site. WebRTC provides complete free call without =
any toll.
> =20

I can't tell you the actual numbers, but when presented with the choice =
of calling a toll free number
or clicking a button marked "free internet call" - almost no-one on a =
real, busy site clicked the button.
( for every button click there were several orders of magnitude more =
0800 calls from that page).


So from my perspective this is a legacy interop use case with almost =
zero user acceptance.

(as far as I can see no-one has made this use-case desirable in practice =
yet.)
Tim.


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<html><head><base href=3D"x-msg://471/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><br><div><div>On 30 Apr 2012, at 09:42, Ravindran, =
Parthasarathi wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Tim,<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">My =
experience is different. Click-to-call is attractive in case of =
toll-free number in the site. WebRTC provides complete free call without =
any toll.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div></div></div></span></blockquote><br></div>=
<div>I can't tell you the actual numbers, but when presented with the =
choice of calling a toll free number</div><div>or clicking a button =
marked "free internet call" - almost no-one on a real, busy site clicked =
the button.</div><div>( for every button click there were several orders =
of magnitude more 0800 calls from that =
page).</div><div><br></div><div><br></div><div>So from my perspective =
this is a legacy interop use case with almost zero user =
acceptance.</div><div><br></div><div>(as far as I can see no-one has =
made this use-case desirable in practice =
yet.)</div><div>Tim.</div><div><br></div></body></html>=

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From eckelcu@cisco.com  Mon Apr 30 09:22:50 2012
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, "Justin Uberti" <juberti@google.com>, "Cullen Jennings" <fluffy@iii.ca>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi Partha,

I believe such a flow illegal, per RFC 3311:

      o  If the UPDATE is being sent before the completion of the INVITE
         transaction, and the initial INVITE contained an offer, the
         UPDATE cannot be sent with an offer unless the callee has
         generated an answer in a reliable provisional response, has
         received a PRACK for that reliable provisional response, has
         not received any requests (PRACK or UPDATE) with offers that it
         has not answered, and has not sent any UPDATE requests
         containing offers that have not been answered.

Cheers,
Charles

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ravindran, Parthasarathi
> Sent: Monday, April 30, 2012 12:02 AM
> To: Justin Uberti; Cullen Jennings
> Cc: rtcweb@ietf.org
> Subject: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
>=20
> Justin/Cullen,
>=20
> Could you please explain in case for an SDP_OFFER(1), the remote
entity
> replies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the
> expected behavior. The exact callflow is as follows:
>=20
>=20
> Browser1-------------------------Browser2(SIP-JSEP gateway)
>     |        SDP_OFFER(1)            |  INV with offer1
>     |------------------------------->|--->
>     |       SDP_PRANSWER(1)          |  183 with answer1
>     |<-------------------------------|<---
>     |       SDP_OFFER(2)             |   UPDATE with offer2
>     |<-------------------------------|<---
>     |       SDP_ANSWER(2?)           |
>     |--------------------->?
>=20
> The reason for this O/A callflow is due to UPDATE method (RFC 3311)
> mapping in Browser 2 (SIP-JSEP gateway).
>=20
> Thanks
> Partha
>=20
> PS: For simplicity, PRACK message exchange is not chosen in SIP side.
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Without numbers it is impossible to argue, but, if we talk about the 
perceived need, I disagree.  Think of the people who travel abroad and 
cannot call the 800 number. (I routinely use Web interface for calls 
when traveling.)

I am all for  the use case, as described by Jim.

Igor

On 4/30/2012 9:54 AM, Tim Panton wrote:
> ...
> I can't tell you the actual numbers, but when presented with the 
> choice of calling a toll free number
> or clicking a button marked "free internet call" - almost no-one on a 
> real, busy site clicked the button.
> ( for every button click there were several orders of magnitude more 
> 0800 calls from that page).
>
>
> So from my perspective this is a legacy interop use case with almost 
> zero user acceptance.
>
> (as far as I can see no-one has made this use-case desirable in 
> practice yet.)
> Tim.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--------------040806050105080003090604
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    Without numbers it is impossible to argue, but, if we talk about the
    perceived need, I disagree.&nbsp; Think of the people who travel abroad
    and cannot call the 800 number. (I routinely use Web interface for
    calls when traveling.)<br>
    <br>
    I am all for&nbsp; the use case, as described by Jim.<br>
    <br>
    Igor<br>
    <br>
    On 4/30/2012 9:54 AM, Tim Panton wrote:
    <blockquote
      cite="mid:E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com"
      type="cite"><base href="x-msg://471/">...
      <div>I can't tell you the actual numbers, but when presented with
        the choice of calling a toll free number</div>
      <div>or clicking a button marked "free internet call" - almost
        no-one on a real, busy site clicked the button.</div>
      <div>( for every button click there were several orders of
        magnitude more 0800 calls from that page).</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div>So from my perspective this is a legacy interop use case with
        almost zero user acceptance.</div>
      <div><br>
      </div>
      <div>(as far as I can see no-one has made this use-case desirable
        in practice yet.)</div>
      <div>Tim.</div>
      <div><br>
      </div>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
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>
  </body>
</html>

--------------040806050105080003090604--

From nataraju.sip@gmail.com  Mon Apr 30 10:55:18 2012
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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If the scenario considered is without reliable provisional responses. Then
the first ANSWER must be in 200-INV and no more O/A allowed during
INVITE(initial) transaction.

Basic requirement for reliable and unambiguous O/A is -* At any point in
time there could be only one O/A open. *

Also only one O/A suggested during INVITE(initial) transaction.

For reference, rfc6337, list outs different O/A models...

<rfc6337>

            Offer                Answer             RFC    Ini Est Early
     -------------------------------------------------------------------
     1. INVITE Req.          2xx INVITE Resp.     RFC 3261  Y   Y    N
     2. 2xx INVITE Resp.     ACK Req.             RFC 3261  Y   Y    N
     3. INVITE Req.          1xx-rel INVITE Resp. RFC 3262  Y   Y    N
     4. 1xx-rel INVITE Resp. PRACK Req.           RFC 3262  Y   Y    N
     5. PRACK Req.           200 PRACK Resp.      RFC 3262  N   Y    Y
     6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y

          Table 1: Summary of SIP Usage of the Offer/Answer Model

</rfc6337>

Thanks,
Nataraju A B

On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

> Justin/Cullen,
>
> Could you please explain in case for an SDP_OFFER(1), the remote entity
> replies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the
> expected behavior. The exact callflow is as follows:
>
>
> Browser1-------------------------Browser2(SIP-JSEP gateway)
>    |        SDP_OFFER(1)            |  INV with offer1
>    |------------------------------->|--->
>    |       SDP_PRANSWER(1)          |  183 with answer1
>    |<-------------------------------|<---
>    |       SDP_OFFER(2)             |   UPDATE with offer2
>    |<-------------------------------|<---
>    |       SDP_ANSWER(2?)           |
>    |--------------------->?
>
> The reason for this O/A callflow is due to UPDATE method (RFC 3311)
> mapping in Browser 2 (SIP-JSEP gateway).
>
> Thanks
> Partha
>
> PS: For simplicity, PRACK message exchange is not chosen in SIP side.
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



-- 
Thanks,
Nataraju A.B.

--f46d0401fde7f440bb04bee92796
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

If the scenario considered is without reliable provisional responses. Then =
the first ANSWER must be in 200-INV and no more O/A allowed during INVITE(i=
nitial) transaction.<div><br></div><div>Basic requirement for reliable and =
unambiguous O/A is -<b> At any point in time there could be only one O/A op=
en.=A0</b></div>
<div><br></div><div>Also only one O/A suggested during INVITE(initial)=A0tr=
ansaction.</div><div><br></div><div>For reference, rfc6337, list outs diffe=
rent O/A models...</div><div><br></div><div>&lt;rfc6337&gt;</div><div><pre =
style=3D"word-wrap:break-word;white-space:pre-wrap">
            Offer                Answer             RFC    Ini Est Early
     -------------------------------------------------------------------
     1. INVITE Req.          2xx INVITE Resp.     RFC 3261  Y   Y    N
     2. 2xx INVITE Resp.     ACK Req.             RFC 3261  Y   Y    N
     3. INVITE Req.          1xx-rel INVITE Resp. RFC 3262  Y   Y    N
     4. 1xx-rel INVITE Resp. PRACK Req.           RFC 3262  Y   Y    N
     5. PRACK Req.           200 PRACK Resp.      RFC 3262  N   Y    Y
     6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y

          Table 1: Summary of SIP Usage of the Offer/Answer Model=A0</pre><=
/div><div>&lt;/rfc6337&gt;</div><div><br></div><div>Thanks,</div><div>Natar=
aju A B</div><div><br><div class=3D"gmail_quote">On Mon, Apr 30, 2012 at 12=
:31 PM, Ravindran, Parthasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:pr=
avindran@sonusnet.com" target=3D"_blank">pravindran@sonusnet.com</a>&gt;</s=
pan> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Justin/Cullen,<br>
<br>
Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected b=
ehavior. The exact callflow is as follows:<br>
<br>
<br>
Browser1-------------------------Browser2(SIP-JSEP gateway)<br>
 =A0 =A0| =A0 =A0 =A0 =A0SDP_OFFER(1) =A0 =A0 =A0 =A0 =A0 =A0| =A0INV with =
offer1<br>
 =A0 =A0|-------------------------------&gt;|---&gt;<br>
 =A0 =A0| =A0 =A0 =A0 SDP_PRANSWER(1) =A0 =A0 =A0 =A0 =A0| =A0183 with answ=
er1<br>
 =A0 =A0|&lt;-------------------------------|&lt;---<br>
 =A0 =A0| =A0 =A0 =A0 SDP_OFFER(2) =A0 =A0 =A0 =A0 =A0 =A0 | =A0 UPDATE wit=
h offer2<br>
 =A0 =A0|&lt;-------------------------------|&lt;---<br>
 =A0 =A0| =A0 =A0 =A0 SDP_ANSWER(2?) =A0 =A0 =A0 =A0 =A0 |<br>
 =A0 =A0|---------------------&gt;?<br>
<br>
The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in Browser 2 (SIP-JSEP gateway).<br>
<br>
Thanks<br>
Partha<br>
<br>
PS: For simplicity, PRACK message exchange is not chosen in SIP side.<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><br clear=3D"all"><div><br></div>-- <br><font color=
=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Tha=
nks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>

<br>
</div>

--f46d0401fde7f440bb04bee92796--

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To: "igor.faynberg@alcatel-lucent.com" <igor.faynberg@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 30 Apr 2012 20:31:48 +0200
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Subject: Re: [rtcweb] Use Case draft
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Whether anybody has been successful in the past with this type of use case =
is I think irrelevant.

The enterprise call centre use case is I think a vital use case because it =
is a scenario in which one user is only concerned that they can securely re=
ach an organization/domain and is not concerned about the individual within=
 that domain  that they communicate with.  A suspect quite a large percenta=
ge of RTCWEB applications will be like this and it is not covered in the cu=
rrent use case draft.

So I think we need it.

Regards
Andy




From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Igor Faynberg
Sent: 30 April 2012 17:41
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

Without numbers it is impossible to argue, but, if we talk about the percei=
ved need, I disagree.  Think of the people who travel abroad and cannot cal=
l the 800 number. (I routinely use Web interface for calls when traveling.)

I am all for  the use case, as described by Jim.

Igor

On 4/30/2012 9:54 AM, Tim Panton wrote:
...
I can't tell you the actual numbers, but when presented with the choice of =
calling a toll free number
or clicking a button marked "free internet call" - almost no-one on a real,=
 busy site clicked the button.
( for every button click there were several orders of magnitude more 0800 c=
alls from that page).


So from my perspective this is a legacy interop use case with almost zero u=
ser acceptance.

(as far as I can see no-one has made this use-case desirable in practice ye=
t.)
Tim.






_______________________________________________

rtcweb mailing list

rtcweb@ietf.org<mailto:rtcweb@ietf.org>

https://www.ietf.org/mailman/listinfo/rtcweb

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 link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>=
<span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1=
F497D'>Whether anybody has been successful in the past with this type of us=
e case is I think irrelevant.<o:p></o:p></span></p><p class=3DMsoNormal><sp=
an style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F49=
7D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-si=
ze:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The enterprise =
call centre use case is I think a vital use case because it is a scenario i=
n which one user is only concerned that they can securely reach an organiza=
tion/domain and is not concerned about the individual within that domain&nb=
sp; that they communicate with.&nbsp; A suspect quite a large percentage of=
 RTCWEB applications will be like this and it is not covered in the current=
 use case draft.<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'f=
ont-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbs=
p;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;fon=
t-family:"Calibri","sans-serif";color:#1F497D'>So I think we need it.<o:p><=
/o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-f=
amily:"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","sa=
ns-serif";color:#1F497D'>Regards<o:p></o:p></span></p><p class=3DMsoNormal>=
<span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1=
F497D'>Andy<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-s=
ize: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-fam=
ily:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p cl=
ass=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><sp=
an style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F49=
7D'><o:p>&nbsp;</o:p></span></p><div style=3D'border:none;border-left:solid=
 blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'><div><div style=3D'border:none;borde=
r-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:w=
indowtext'>From:</span></b><span style=3D'font-size:10.0pt;font-family:"Tah=
oma","sans-serif";color:windowtext'> rtcweb-bounces@ietf.org [mailto:rtcweb=
-bounces@ietf.org] <b>On Behalf Of </b>Igor Faynberg<br><b>Sent:</b> 30 Apr=
il 2012 17:41<br><b>To:</b> rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb]=
 Use Case draft<o:p></o:p></span></p></div></div><p class=3DMsoNormal><o:p>=
&nbsp;</o:p></p><p class=3DMsoNormal>Without numbers it is impossible to ar=
gue, but, if we talk about the perceived need, I disagree.&nbsp; Think of t=
he people who travel abroad and cannot call the 800 number. (I routinely us=
e Web interface for calls when traveling.)<br><br>I am all for&nbsp; the us=
e case, as described by Jim.<br><br>Igor<br><br>On 4/30/2012 9:54 AM, Tim P=
anton wrote: <o:p></o:p></p><p class=3DMsoNormal>... <o:p></o:p></p><div><p=
 class=3DMsoNormal>I can't tell you the actual numbers, but when presented =
with the choice of calling a toll free number<o:p></o:p></p></div><div><p c=
lass=3DMsoNormal>or clicking a button marked &quot;free internet call&quot;=
 - almost no-one on a real, busy site clicked the button.<o:p></o:p></p></d=
iv><div><p class=3DMsoNormal>( for every button click there were several or=
ders of magnitude more 0800 calls from that page).<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>So from my perspective=
 this is a legacy interop use case with almost zero user acceptance.<o:p></=
o:p></p></div><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>(as far as I can see no-one has made this use-case desira=
ble in practice yet.)<o:p></o:p></p></div><div><p class=3DMsoNormal>Tim.<o:=
p></o:p></p></div><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><pre=
><o:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>__________________=
_____________________________<o:p></o:p></pre><pre>rtcweb mailing list<o:p>=
</o:p></pre><pre><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p=
></o:p></pre><pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre></div></di=
v></body></html>=

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On 4/30/2012 7:41 AM, Stefan Hakansson LK wrote:
> On 04/27/2012 06:43 PM, Timothy B. Terriberry wrote:
>> Ted Hardie wrote:
>>> The chairs would like to ask the working group to focus on the use
>>> case draft. If you have use cases that need to be added to the
>>> document or text changes you'd like to suggest, please send them in
>>
>> I proposed the following use-case back in February, but there wasn't
>> much discussion on actually adding it to the document:
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg03357.html
>>
>> Let me know if the WG would like to proceed with something like this.
>
> I think adding another small derivative of the simple video chat (this
> one with peer-to-peer file transfer added) makes a lot of sense. For
> one, we get a use case that requires reliable data (now we only have a
> req for "short latency datagram" which sound like unreliable to me); in
> addition we get a requirement for the data channel API to be able to use
> blobs (as defined in the File API W3C rec) as input/output.

The game case can need both reliable and unreliable data at the same 
time, which gets us reliable, unreliable and multiple streams.

> A third good requirement that can be derived (if we want to) is the
> ability to prioritize data in relation to audio/video.

File transfer or "let me show you the photo I took" while talking would 
do I think.


-- 
Randell Jesup
randell-ietf@jesup.org

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On 4/30/2012 9:54 AM, Tim Panton wrote:
>
> On 30 Apr 2012, at 09:42, Ravindran, Parthasarathi wrote:

>> My experience is different. Click-to-call is attractive in case of
>> toll-free number in the site. WebRTC provides complete free call
>> without any toll.
>
> I can't tell you the actual numbers, but when presented with the choice
> of calling a toll free number
> or clicking a button marked "free internet call" - almost no-one on a
> real, busy site clicked the button.
> ( for every button click there were several orders of magnitude more
> 0800 calls from that page).
>
>
> So from my perspective this is a legacy interop use case with almost
> zero user acceptance.

How many people browsing have mic, speakers, or headphones, set up, and 
are comfortable with it?

Things are changing, but a friend who consults for non-tech small 
businesspeople and individual home-based businesses avoids using audio 
when doing screen-share type stuff and instead starts a parallel audio 
POTS call, because it's too frequently a waste of time for them to find 
and set up headphones, get their mic to work.

If it's people who are already using Skype/etc, there's a better chance 
- but they still may only be comfortable with Skype for "calls".


-- 
Randell Jesup
randell-ietf@jesup.org

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From: Stephan Wenger <stewe@stewe.org>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Use Case draft
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Hi Randell,
I don't buy this argument.
Speakers are a standard piece of equipment now; almost everyone with a PC
has them in some form.  Without a webcam, webrtc would be pretty useless,
and every webcam nowadays has a microphone.
I have no reason to distrust Tim's assessment, but the reason for the lack
of success of PC facilitated sales doers you cited, IMO, cannot be the
main reason, and certainly not among the expected users of webrtc (who
WILL have speakers, camera and mic set up, ready to go, and used to make
calls from their machine).  And if your argument refers to the call center
site--well, those guys will get their equipment right and train their
employees, or go out of business.
Lets not dismiss a use case based on short term observations.
Stephan

On 4.30.2012 23:09 , "Randell Jesup" <randell-ietf@jesup.org> wrote:

>On 4/30/2012 9:54 AM, Tim Panton wrote:
>>
>> On 30 Apr 2012, at 09:42, Ravindran, Parthasarathi wrote:
>
>>> My experience is different. Click-to-call is attractive in case of
>>> toll-free number in the site. WebRTC provides complete free call
>>> without any toll.
>>
>> I can't tell you the actual numbers, but when presented with the choice
>> of calling a toll free number
>> or clicking a button marked "free internet call" - almost no-one on a
>> real, busy site clicked the button.
>> ( for every button click there were several orders of magnitude more
>> 0800 calls from that page).
>>
>>
>> So from my perspective this is a legacy interop use case with almost
>> zero user acceptance.
>
>How many people browsing have mic, speakers, or headphones, set up, and
>are comfortable with it?
>
>Things are changing, but a friend who consults for non-tech small
>businesspeople and individual home-based businesses avoids using audio
>when doing screen-share type stuff and instead starts a parallel audio
>POTS call, because it's too frequently a waste of time for them to find
>and set up headphones, get their mic to work.
>
>If it's people who are already using Skype/etc, there's a better chance
>- but they still may only be comfortable with Skype for "calls".
>
>
>--=20
>Randell Jesup
>randell-ietf@jesup.org
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



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On 4/30/2012 6:27 PM, Stephan Wenger wrote:
> Hi Randell,
> I don't buy this argument.
> Speakers are a standard piece of equipment now; almost everyone with a PC
> has them in some form.  Without a webcam, webrtc would be pretty useless,
> and every webcam nowadays has a microphone.

And why do you believe every desktop computer has a webcam?  All 
nowadays have connectors for mic and speakers, and almost all have 
speakers, but probably (guess) 50% of desktops out there don't have a 
microphone attached (probably more than 50%).

Most *laptops* have builtin cameras and mics now, but a few years ago 
that wasn't the case.

> I have no reason to distrust Tim's assessment, but the reason for the lack
> of success of PC facilitated sales doers you cited, IMO, cannot be the
> main reason, and certainly not among the expected users of webrtc (who
> WILL have speakers, camera and mic set up, ready to go, and used to make
> calls from their machine).

One hopes so, yes, though webrtc does not require sending video.

>  And if your argument refers to the call center
> site--well, those guys will get their equipment right and train their
> employees, or go out of business.
> Lets not dismiss a use case based on short term observations.

Agreed.  I was just giving an alternate explanation why "no one clicked 
Free Internet Call", and why that was NOT a reason to invalidate the 
usecase.


-- 
Randell Jesup
randell-ietf@jesup.org

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Yes, I was under the impression that SIP enforced this requirement,
although I am probably not aware of all the corner cases. Is there a
real-world scenario where this flow is required?

On Mon, Apr 30, 2012 at 10:55 AM, Nataraju A.B <nataraju.sip@gmail.com>wrote:

> If the scenario considered is without reliable provisional responses. Then
> the first ANSWER must be in 200-INV and no more O/A allowed during
> INVITE(initial) transaction.
>
> Basic requirement for reliable and unambiguous O/A is -* At any point in
> time there could be only one O/A open. *
>
> Also only one O/A suggested during INVITE(initial) transaction.
>
> For reference, rfc6337, list outs different O/A models...
>
> <rfc6337>
>
>
>             Offer                Answer             RFC    Ini Est Early
>      -------------------------------------------------------------------
>      1. INVITE Req.          2xx INVITE Resp.     RFC 3261  Y   Y    N
>      2. 2xx INVITE Resp.     ACK Req.             RFC 3261  Y   Y    N
>      3. INVITE Req.          1xx-rel INVITE Resp. RFC 3262  Y   Y    N
>      4. 1xx-rel INVITE Resp. PRACK Req.           RFC 3262  Y   Y    N
>      5. PRACK Req.           200 PRACK Resp.      RFC 3262  N   Y    Y
>      6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y
>
>           Table 1: Summary of SIP Usage of the Offer/Answer Model
>
> </rfc6337>
>
> Thanks,
> Nataraju A B
>
> On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthasarathi <
> pravindran@sonusnet.com> wrote:
>
>> Justin/Cullen,
>>
>> Could you please explain in case for an SDP_OFFER(1), the remote entity
>> replies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the
>> expected behavior. The exact callflow is as follows:
>>
>>
>> Browser1-------------------------Browser2(SIP-JSEP gateway)
>>    |        SDP_OFFER(1)            |  INV with offer1
>>    |------------------------------->|--->
>>    |       SDP_PRANSWER(1)          |  183 with answer1
>>    |<-------------------------------|<---
>>    |       SDP_OFFER(2)             |   UPDATE with offer2
>>    |<-------------------------------|<---
>>    |       SDP_ANSWER(2?)           |
>>    |--------------------->?
>>
>> The reason for this O/A callflow is due to UPDATE method (RFC 3311)
>> mapping in Browser 2 (SIP-JSEP gateway).
>>
>> Thanks
>> Partha
>>
>> PS: For simplicity, PRACK message exchange is not chosen in SIP side.
>>
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
>
> --
> Thanks,
> Nataraju A.B.
>
>

--20cf306677dbfe043e04bef0a443
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Yes, I was under the impression that SIP enforced this requirement, althoug=
h I am probably not aware of all the corner cases. Is there a real-world sc=
enario where this flow is required?<br><br><div class=3D"gmail_quote">On Mo=
n, Apr 30, 2012 at 10:55 AM, Nataraju A.B <span dir=3D"ltr">&lt;<a href=3D"=
mailto:nataraju.sip@gmail.com" target=3D"_blank">nataraju.sip@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">If the scenario considered is without reliab=
le provisional responses. Then the first ANSWER must be in 200-INV and no m=
ore O/A allowed during INVITE(initial) transaction.<div>


<br></div><div>Basic requirement for reliable and unambiguous O/A is -<b> A=
t any point in time there could be only one O/A open.=C2=A0</b></div>
<div><br></div><div>Also only one O/A suggested during INVITE(initial)=C2=
=A0transaction.</div><div><br></div><div>For reference, rfc6337, list outs =
different O/A models...</div><div><br></div><div>&lt;rfc6337&gt;</div><div>=
<pre style=3D"word-wrap:break-word;white-space:pre-wrap">

            Offer                Answer             RFC    Ini Est Early
     -------------------------------------------------------------------
     1. INVITE Req.          2xx INVITE Resp.     RFC 3261  Y   Y    N
     2. 2xx INVITE Resp.     ACK Req.             RFC 3261  Y   Y    N
     3. INVITE Req.          1xx-rel INVITE Resp. RFC 3262  Y   Y    N
     4. 1xx-rel INVITE Resp. PRACK Req.           RFC 3262  Y   Y    N
     5. PRACK Req.           200 PRACK Resp.      RFC 3262  N   Y    Y
     6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y

          Table 1: Summary of SIP Usage of the Offer/Answer Model=C2=A0</pr=
e></div><div>&lt;/rfc6337&gt;</div><div><br></div><div>Thanks,</div><div>Na=
taraju A B</div><div><br><div class=3D"gmail_quote"><div><div>On Mon, Apr 3=
0, 2012 at 12:31 PM, Ravindran, Parthasarathi <span dir=3D"ltr">&lt;<a href=
=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">pravindran@sonusnet.c=
om</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>Justin/Cullen,<br>
<br>
Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected b=
ehavior. The exact callflow is as follows:<br>
<br>
<br>
Browser1-------------------------Browser2(SIP-JSEP gateway)<br>
 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0SDP_OFFER(1) =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0INV with offer1<br>
 =C2=A0 =C2=A0|-------------------------------&gt;|---&gt;<br>
 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 SDP_PRANSWER(1) =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0| =C2=A0183 with answer1<br>
 =C2=A0 =C2=A0|&lt;-------------------------------|&lt;---<br>
 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 SDP_OFFER(2) =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 | =C2=A0 UPDATE with offer2<br>
 =C2=A0 =C2=A0|&lt;-------------------------------|&lt;---<br>
 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 SDP_ANSWER(2?) =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 |<br>
 =C2=A0 =C2=A0|---------------------&gt;?<br>
<br>
The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in Browser 2 (SIP-JSEP gateway).<br>
<br>
Thanks<br>
Partha<br>
<br>
PS: For simplicity, PRACK message exchange is not chosen in SIP side.<br>
<br>
<br>
<br>
<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" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></blockquote></div><span><font color=3D"#888888"><br><br clear=3D"all=
"><div><br></div>-- <br><font color=3D"#000099"><font face=3D"&#39;courier =
new&#39;, monospace" size=3D"1">Thanks,</font></font><div><font color=3D"#0=
00099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Nataraju =
A.B.</font></font></div>




<br>
</font></span></div>
</blockquote></div><br>

--20cf306677dbfe043e04bef0a443--

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From: Justin Uberti <juberti@google.com>
Date: Mon, 30 Apr 2012 20:02:59 -0700
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Subject: Re: [rtcweb] Alternative Proposal for Dynamic Codec Parameter Change (Was: Re: Resolution negotiation - a contribution)
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On Mon, Apr 30, 2012 at 2:13 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2012-04-25 18:13, Harald Alvestrand wrote:
> > On 04/25/2012 04:35 PM, Magnus Westerlund wrote:
> >> On 2012-04-25 05:20, Justin Uberti wrote:
> >>> On Tue, Apr 24, 2012 at 10:25 AM, Magnus Westerlund
> >>> <magnus.westerlund@ericsson.com<mailto:magnus.westerlund@ericsson.com
> >>
> >>> wrote:
> >>>
> >>>      A) Using COP and using SDP based signaling for the dynamic
> changes are
> >>>      two quite different models in relation to how the interaction
> happens.
> >>>
> >>>      For COP this all happens in the browser, normally initiated by t=
he
> >>>      browser's own determination that a COP request or notification i=
s
> >>>      needed. Harald's proposal appears to require that the JS initiat=
e
> a
> >>>      renegotiation. This puts a requirement on the implementation to
> listen
> >>>      to the correct callbacks to know when changes happens, such as
> window
> >>>      resize. To my knowledge there are not yet any proposals for how
> the
> >>>      browser can initiate a JSEP renegotiation.
> >>>
> >>>
> >>> Maybe I misunderstand what you are saying, but the application will
> >>> definitely know when the browser size changes, and can then trigger a
> >>> JSEP renegotiation by calling createOffer and shipping it off.
> >> Yes, they can but that requires them to write code to do this. With CO=
P
> >> the browser can do it and it works well for all applications, not only
> >> the ones that have implementors that are aware of the need to do this.
> >>
> >> I am proposing a mechanism that makes the default behavior if you care
> >> nothing about controlling or optimizing this behavior in your JS to be
> >> good. In addition JS that wants to control it can affect the behavior
> >> through generic mechanisms, such as controlling the video element. And
> >> if clear need is shown and some functionality is missing, W3C can crea=
te
> >> an API to provide explicit control.
> > It's not at all clear to me that the browser being in control is right
> > for all cases.
> > Renegotiating has a cost, and some of the costs are known only to the
> > application.
> >
> > We may possibly handle this via constraints like
> > "VideoEnumRenegotiationTendency" or something like that - when you get
> > the proposal together more, I'd like to see the controls you're
> proposing.
> >>> I have the opposite concern - how can the browser know when the
> >>> application makes a change, for instance to display a participant in =
a
> >>> large view versus a small view? This may be possible if using a<video=
/>
> >>> for display, but I don't think it will be possible when using WebGL f=
or
> >>> rendering, where the size of the view will be dictated only by the
> >>> geometry of the scene.
> >> What I understand from my colleagues is that all rendering of video ha=
ve
> >> to go through a video element, even if not displayed directly. Thus th=
e
> >> application can control the size of the video element used and possibl=
y
> >> consumed by WebGL to be included in rendering. Thus the application wi=
ll
> >> have control over this even when COP is being used.
> > We have on the "list of things that need doing" the connection of a
> > mediastream directly to a canvas, a WebGL element or being captured to =
a
> > file, so don't gamble too much on the video element to stay in the path=
.
> >
> > Example code for capturing video to a canvas (taken from a real app):
> >
> >         video.src =3D webkitURL.createObjectURL(s);
> >         canvas.getContext("2d").drawImage(video, 0, 0, 300, 300, 0, 0,
> > 300, 300);
>
> According to my Colleague Per-Erik you can use video.width and
> video.height to set the resolution the video element should have.
>
> see
> http://dev.w3.org/html5/spec/the-video-element.html#the-video-element
>
> Thus the code you presented should be something like.
>
> video.src =3D webkitURL.createObjectURL(s);
> video.width =3D 300;
> video.height =3D 300;
> canvas.getContext("2d").drawImage(video, 0, 0, 300, 300);
>
> That would set the video objects size to a 300, 300 video resolution.
> The drawImage call I have removed the part which selects a cropping of
> the source video to automatically show the full video in the specified
> area.
>
> The adaptation of the video will of course not happen instantly so until
> that happens the video will be scaled to the specified width and height.
> And the actual delivered resolution will depend on the COP request and
> what the sender support.
>

I think this is a reasonable way to approach the problem as long as a
<video/> element stays in the path. But as Harald mentions we have already
considered several usages where a <video/> would not be in the path, i.e.
connecting a MediaStream to a file, or perhaps as input to another
PeerConnection.

Perhaps we need to provide hooks directly on the MediaStream to allow the
resolution to be specified, and these hooks are invoked automatically when
the MediaStream is connected to a <video/>.

>
> >> I agree that it needs to be clearly specified. But COP appears to be
> >> less prone to complete failures by using target values. Then at least =
it
> >> will receive video that can be used, even if not optimal. imageattr ca=
n
> >> result in failures in the negotiation requiring at minimal a second
> >> round potentially no agreed on acceptable resolution then that is
> >> clearly sub-optimal. That is even without considering involving JS
> >> mainipulating this.
> >
> > I certainly hope that we can avoid the need for SDP manipulation of JS
> > in all of these proposals.
> >
>
> Looking forward to a bit more detailed explanation of how that is
> accomplished in relation to the API.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> 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
> ----------------------------------------------------------------------
>
>

--20cf30667c2f2faaa304bef0d06a
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<br><br><div class=3D"gmail_quote">On Mon, Apr 30, 2012 at 2:13 AM, Magnus =
Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</span> wro=
te:<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=
012-04-25 18:13, Harald Alvestrand wrote:<br>
&gt; On 04/25/2012 04:35 PM, Magnus Westerlund wrote:<br>
&gt;&gt; On 2012-04-25 05:20, Justin Uberti wrote:<br>
&gt;&gt;&gt; On Tue, Apr 24, 2012 at 10:25 AM, Magnus Westerlund<br>
&gt;&gt;&gt; &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.w=
esterlund@ericsson.com</a>&lt;mailto:<a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a>&gt;&gt;<br>
&gt;&gt;&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0A) Using COP and using SDP based signaling=
 for the dynamic changes are<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0two quite different models in relation to =
how the interaction happens.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0For COP this all happens in the browser, n=
ormally initiated by the<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0browser&#39;s own determination that a COP=
 request or notification is<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0needed. Harald&#39;s proposal appears to r=
equire that the JS initiate a<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0renegotiation. This puts a requirement on =
the implementation to listen<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0to the correct callbacks to know when chan=
ges happens, such as window<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0resize. To my knowledge there are not yet =
any proposals for how the<br>
&gt;&gt;&gt; =C2=A0 =C2=A0 =C2=A0browser can initiate a JSEP renegotiation.=
<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Maybe I misunderstand what you are saying, but the application=
 will<br>
&gt;&gt;&gt; definitely know when the browser size changes, and can then tr=
igger a<br>
&gt;&gt;&gt; JSEP renegotiation by calling createOffer and shipping it off.=
<br>
&gt;&gt; Yes, they can but that requires them to write code to do this. Wit=
h COP<br>
&gt;&gt; the browser can do it and it works well for all applications, not =
only<br>
&gt;&gt; the ones that have implementors that are aware of the need to do t=
his.<br>
&gt;&gt;<br>
&gt;&gt; I am proposing a mechanism that makes the default behavior if you =
care<br>
&gt;&gt; nothing about controlling or optimizing this behavior in your JS t=
o be<br>
&gt;&gt; good. In addition JS that wants to control it can affect the behav=
ior<br>
&gt;&gt; through generic mechanisms, such as controlling the video element.=
 And<br>
&gt;&gt; if clear need is shown and some functionality is missing, W3C can =
create<br>
&gt;&gt; an API to provide explicit control.<br>
&gt; It&#39;s not at all clear to me that the browser being in control is r=
ight<br>
&gt; for all cases.<br>
&gt; Renegotiating has a cost, and some of the costs are known only to the<=
br>
&gt; application.<br>
&gt;<br>
&gt; We may possibly handle this via constraints like<br>
&gt; &quot;VideoEnumRenegotiationTendency&quot; or something like that - wh=
en you get<br>
&gt; the proposal together more, I&#39;d like to see the controls you&#39;r=
e proposing.<br>
&gt;&gt;&gt; I have the opposite concern - how can the browser know when th=
e<br>
&gt;&gt;&gt; application makes a change, for instance to display a particip=
ant in a<br>
&gt;&gt;&gt; large view versus a small view? This may be possible if using =
a&lt;video/&gt;<br>
&gt;&gt;&gt; for display, but I don&#39;t think it will be possible when us=
ing WebGL for<br>
&gt;&gt;&gt; rendering, where the size of the view will be dictated only by=
 the<br>
&gt;&gt;&gt; geometry of the scene.<br>
&gt;&gt; What I understand from my colleagues is that all rendering of vide=
o have<br>
&gt;&gt; to go through a video element, even if not displayed directly. Thu=
s the<br>
&gt;&gt; application can control the size of the video element used and pos=
sibly<br>
&gt;&gt; consumed by WebGL to be included in rendering. Thus the applicatio=
n will<br>
&gt;&gt; have control over this even when COP is being used.<br>
&gt; We have on the &quot;list of things that need doing&quot; the connecti=
on of a<br>
&gt; mediastream directly to a canvas, a WebGL element or being captured to=
 a<br>
&gt; file, so don&#39;t gamble too much on the video element to stay in the=
 path.<br>
&gt;<br>
&gt; Example code for capturing video to a canvas (taken from a real app):<=
br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 video.src =3D webkitURL.createObjectURL(s)=
;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 canvas.getContext(&quot;2d&quot;).drawImag=
e(video, 0, 0, 300, 300, 0, 0,<br>
&gt; 300, 300);<br>
<br>
</div></div>According to my Colleague Per-Erik you can use video.width and<=
br>
video.height to set the resolution the video element should have.<br>
<br>
see<br>
<a href=3D"http://dev.w3.org/html5/spec/the-video-element.html#the-video-el=
ement" target=3D"_blank">http://dev.w3.org/html5/spec/the-video-element.htm=
l#the-video-element</a><br>
<br>
Thus the code you presented should be something like.<br>
<br>
video.src =3D webkitURL.createObjectURL(s);<br>
video.width =3D 300;<br>
video.height =3D 300;<br>
canvas.getContext(&quot;2d&quot;).drawImage(video, 0, 0, 300, 300);<br>
<br>
That would set the video objects size to a 300, 300 video resolution.<br>
The drawImage call I have removed the part which selects a cropping of<br>
the source video to automatically show the full video in the specified<br>
area.<br>
<br>
The adaptation of the video will of course not happen instantly so until<br=
>
that happens the video will be scaled to the specified width and height.<br=
>
And the actual delivered resolution will depend on the COP request and<br>
what the sender support.<br></blockquote><div><br></div><div>I think this i=
s a reasonable way to approach the problem as long as a &lt;video/&gt; elem=
ent stays in the path. But as Harald mentions we have already considered se=
veral usages where a &lt;video/&gt; would not be in the path, i.e. connecti=
ng a MediaStream to a file, or perhaps as input to another PeerConnection.<=
/div>

<div><br></div><div>Perhaps we need to provide hooks directly on the MediaS=
tream to allow the resolution to be specified, and these hooks are invoked =
automatically when the MediaStream is connected to a &lt;video/&gt;.</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"><br>
&gt;&gt; I agree that it needs to be clearly specified. But COP appears to =
be<br>
&gt;&gt; less prone to complete failures by using target values. Then at le=
ast it<br>
&gt;&gt; will receive video that can be used, even if not optimal. imageatt=
r can<br>
&gt;&gt; result in failures in the negotiation requiring at minimal a secon=
d<br>
&gt;&gt; round potentially no agreed on acceptable resolution then that is<=
br>
&gt;&gt; clearly sub-optimal. That is even without considering involving JS=
<br>
&gt;&gt; mainipulating this.<br>
&gt;<br>
&gt; I certainly hope that we can avoid the need for SDP manipulation of JS=
<br>
&gt; in all of these proposals.<br>
&gt;<br>
<br>
</div>Looking forward to a bit more detailed explanation of how that is<br>
accomplished in relation to the API.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<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 71=
48287</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">+46=
 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</div></div></blockquote></div><br>

--20cf30667c2f2faaa304bef0d06a--

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Date: Mon, 30 Apr 2012 23:41:37 -0400
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From: Marshall Eubanks <marshall.eubanks@gmail.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
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Subject: Re: [rtcweb] Use Case draft
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On Mon, Apr 30, 2012 at 2:31 PM, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:
> Whether anybody has been successful in the past with this type of use cas=
e
> is I think irrelevant.
>
>
>
> The enterprise call centre use case is I think a vital use case because i=
t
> is a scenario in which one user is only concerned that they can securely
> reach an organization/domain and is not concerned about the individual
> within that domain=A0 that they communicate with.=A0 A suspect quite a la=
rge
> percentage of RTCWEB applications will be like this and it is not covered=
 in
> the current use case draft.

I agree that this is a very useful use case and one I think is going
to get a lot of traction. There is a very solid
business case for this.  However, I have a fair amount of experience
with a video call center for a client, and it is not as simple as it
might seem.

The essence of course is that you get the next available person, i.e.,
it is anycast. Determining who the next available person is is not
trivial, nor is error recovery. (If I call you, and you don't answer
or the call drops or whatever,  I can leave a message or try later. If
I call a help desk, and this happens, I want a new agent, ideally
automatically.) Call forwarding (e.g., first tier to second tier
technical support) is essential, and it may be anycast or directed.
There are also some security oddities  - if I am connecting from home,
I may need to authenticate, use a credit card, etc. If I am connecting
from inside a store, and providing in store video technical support is
big part of the market, then the store authenticates me off line and
the call really should just be a button push, which implies that the
store has previously authenticated some sort of master session. In
addition, unlike most video calls, in the enterprise call center a
supervisor may need to be able to monitor (i.e., watch) a call, and in
some circumstances (financial or medical calls, for example) there
will need to be third party recording. I believe that  these details
would be different from the typical RTCWEB scenario.

Also, there will be a temptation to do the anycasting by the
techniques used to load balance servers in a data center, but I think
that may not be sufficient. The call "center" may in fact be spread
completely across the planet (daytime support in the US, nighttime
support in India, for example) and be on multiple autonomous systems
(and even from people's homes), which gives rise to some of the
transport issues NVO3 may face, but without any opportunity for packet
tagging. Plus, there will complicated rules about who can be selected
next. RTCWEB shouldn't worry about the intricacies of bathroom break
policies; these complexities should be dealt with by an
enterprise-side database, which to me (together with some of the other
issues above) suggests that this would probably benefit from  API
support.

Regards
Marshall


>
>
>
> So I think we need it.
>
>
>
> Regards
>
> Andy
>
>
>
>
>
>
>
>
>
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
> Igor Faynberg
> Sent: 30 April 2012 17:41
> To: rtcweb@ietf.org
>
>
> Subject: Re: [rtcweb] Use Case draft
>
>
>
> Without numbers it is impossible to argue, but, if we talk about the
> perceived need, I disagree.=A0 Think of the people who travel abroad and
> cannot call the 800 number. (I routinely use Web interface for calls when
> traveling.)
>
>
>
> I am all for=A0 the use case, as described by Jim.
>
> Igor
>
> On 4/30/2012 9:54 AM, Tim Panton wrote:
>
> ...
>
> I can't tell you the actual numbers, but when presented with the choice o=
f
> calling a toll free number
>
> or clicking a button marked "free internet call" - almost no-one on a rea=
l,
> busy site clicked the button.
>
> ( for every button click there were several orders of magnitude more 0800
> calls from that page).
>
>
>
>
>
> So from my perspective this is a legacy interop use case with almost zero
> user acceptance.
>
>
>
> (as far as I can see no-one has made this use-case desirable in practice
> yet.)
>
> Tim.
>
>
>
>
>
>
>
> _______________________________________________
>
> 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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Subject: Re: [rtcweb] Use Case draft
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On 04/30/2012 11:00 PM, Randell Jesup wrote:
> On 4/30/2012 7:41 AM, Stefan Hakansson LK wrote:
>> On 04/27/2012 06:43 PM, Timothy B. Terriberry wrote:
>>> Ted Hardie wrote:
>>>> The chairs would like to ask the working group to focus on the use
>>>> case draft. If you have use cases that need to be added to the
>>>> document or text changes you'd like to suggest, please send them in
>>>
>>> I proposed the following use-case back in February, but there wasn't
>>> much discussion on actually adding it to the document:
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg03357.html
>>>
>>> Let me know if the WG would like to proceed with something like this.
>>
>> I think adding another small derivative of the simple video chat (this
>> one with peer-to-peer file transfer added) makes a lot of sense. For
>> one, we get a use case that requires reliable data (now we only have a
>> req for "short latency datagram" which sound like unreliable to me); in
>> addition we get a requirement for the data channel API to be able to use
>> blobs (as defined in the File API W3C rec) as input/output.
>
> The game case can need both reliable and unreliable data at the same
> time, which gets us reliable, unreliable and multiple streams.

You're right; but currently only game state updates is mentioned. 
Probably that use case should be updated.
>
>> A third good requirement that can be derived (if we want to) is the
>> ability to prioritize data in relation to audio/video.
>
> File transfer or "let me show you the photo I took" while talking would
> do I think.
>
>

