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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 5 Jan 2016 20:55:49 +0100
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Subject: [rtcweb] Is "use_srtp" DTLS extension required if just DataChannel is requested?
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If a WebRTC endpoint acting as DTLS client just wants to offer a
DataChannel (and not audio or video) should it also include the
"use_srtp" DTLS extension?

Should a WebRTC endpoint acting as DTLS server reject the DTLS
handshake if the DTLS client did not include the "use_srtp" DTLS
extension?

Thanks.

--=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] Is "use_srtp" DTLS extension required if just DataChannel is requested?
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--089e01184168928ab405289d0758
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I think the decision was that "use_srtp" DTLS extension is required and
DTLS server should reject the handshake if this extension is not offered.
This was decided since it is difficult to predict if RTP would be added in
the bundle over the same transport in some point in the future.
Without "use_srtp"
DTLS extension being present from the start adding RTP would require
negotiating a whole new transport including ICE setup since "use_srtp"
extension cannot be added to an existing DTLS association.

_____________
Roman Shpount

On Tue, Jan 5, 2016 at 2:55 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> If a WebRTC endpoint acting as DTLS client just wants to offer a
> DataChannel (and not audio or video) should it also include the
> "use_srtp" DTLS extension?
>
> Should a WebRTC endpoint acting as DTLS server reject the DTLS
> handshake if the DTLS client did not include the "use_srtp" DTLS
> extension?
>
> Thanks.
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">I think the decision was that=C2=A0<span style=3D"color:rg=
b(0,0,0);font-size:12.8px">&quot;use_srtp&quot; DTLS extension is required =
and DTLS server should reject the handshake if this extension is not offere=
d. This was decided since it is difficult to predict if RTP would be added =
in the bundle over the same transport in some point in the future. Without=
=C2=A0</span><span style=3D"color:rgb(0,0,0);font-size:12.8px">&quot;use_sr=
tp&quot; DTLS extension being present from the start adding RTP would requi=
re negotiating a whole new transport including ICE setup since &quot;use_sr=
tp&quot; extension cannot be added to an existing DTLS association.</span><=
/div><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_=
signature">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Tue, Jan 5, 2016 at 2:55 PM, I=C3=B1aki B=
az Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=
=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">If a WebRTC endpoint acting as DTLS client just wants to offer a<br>
DataChannel (and not audio or video) should it also include the<br>
&quot;use_srtp&quot; DTLS extension?<br>
<br>
Should a WebRTC endpoint acting as DTLS server reject the DTLS<br>
handshake if the DTLS client did not include the &quot;use_srtp&quot; DTLS<=
br>
extension?<br>
<br>
Thanks.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>

--089e01184168928ab405289d0758--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 5 Jan 2016 22:37:16 +0100
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Subject: Re: [rtcweb] Is "use_srtp" DTLS extension required if just DataChannel is requested?
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2016-01-05 22:36 GMT+01:00 Roman Shpount <roman@telurix.com>:
> I think the decision was that "use_srtp" DTLS extension is required and D=
TLS
> server should reject the handshake if this extension is not offered. This
> was decided since it is difficult to predict if RTP would be added in the
> bundle over the same transport in some point in the future. Without
> "use_srtp" DTLS extension being present from the start adding RTP would
> require negotiating a whole new transport including ICE setup since
> "use_srtp" extension cannot be added to an existing DTLS association.

Clear. Thanks.


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


From nobody Thu Jan  7 07:37:49 2016
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Subject: [rtcweb] JSEP: "VoiceActivityDetection" option and cient-mixer "vad" extension
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JSEP Section 5.2.3.4 discusses the implications of the
"VoiceActivityDetection" option for the offer.

One thing that is not covered is whether the "VoiceActivityDetection"
option determines whether the client-mixer "vad" extension is set to "on"
or "off".

For example, I believe that the W3C WEBRTC WG determined that the
client-mixer extension (support for which is required in RTP-Usage) should
always be advertised in the offer.  RFC 6464 Section 4 gives an example of
the a=extmap line:

a=extmap:6 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on


So, my question is whether "vad=on" or "vad=off" is determined by the
"VoiceActivityDetection" option, or if not, whether there is a setting
(e.g. "vad=on") that is always present in the offer.

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<div dir=3D"ltr">JSEP Section 5.2.3.4 discusses the implications of the &qu=
ot;VoiceActivityDetection&quot; option for the offer.=C2=A0<div><br></div><=
div>One thing that is not covered is whether the &quot;VoiceActivityDetecti=
on&quot; option determines whether the client-mixer &quot;vad&quot; extensi=
on is set to &quot;on&quot; or &quot;off&quot;. =C2=A0</div><div><br></div>=
<div>For example, I believe that the W3C WEBRTC WG determined that the clie=
nt-mixer extension (support for which is required in RTP-Usage) should alwa=
ys be advertised in the offer.=C2=A0 RFC 6464 Section 4 gives an example of=
 the a=3Dextmap line:=C2=A0</div><div><br></div><div><pre class=3D"" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">=
a=3Dextmap:6 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=3Don</pre><pre=
 class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;c=
olor:rgb(0,0,0)"><br></pre><pre class=3D"" style=3D"margin-top:0px;margin-b=
ottom:0px"><font face=3D"arial, sans-serif"><span style=3D"white-space:norm=
al">So, my question is whether &quot;vad=3Don&quot; or &quot;vad=3Doff&quot=
; is determined by the &quot;VoiceActivityDetection&quot; option, or if not=
, whether there is a setting (e.g. &quot;vad=3Don&quot;) that is always pre=
sent in the offer.=C2=A0</span></font></pre></div></div>

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Finally got around to updating the document. Comments inline; new version
of the doc attached.

On Wed, Nov 4, 2015 at 11:15 PM Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> A. Section 3.2:
>
> I think the draft should note that some RTP Payload format, including
> several of the one recommended in
> draft-ietf-rtcweb-audio-codecs-for-interop-02, do support internal
> redundancy. The codecs that I know for a fact can do this within the
> payload format are: AMR and AMR-WB.
>
> I would suggest adding some sentence on that time delayed redundancy may
> be supported by codecs, and give AMR and AMR-WB as examples.
>

I added a section on AMR/AMR-WB that parallels the Opus section.

>
> B. Section 4.
>
> I think this draft should consider the recommended drafts in
> draft-ietf-rtcweb-audio-codecs-for-interop-02. Here we clearly can
> consider internal redundancy for AMR and AMR-WB.
>

Done.

>
> C. Section 4.2
>
> There are a parameter in AMR and AMR-WB that can negotiate how much
> redundancy one may include: max-red
>

I mentioned this parameter in the doc. Do you want to recommend a default
value?

>
> D. Section 5.1:
>
>     The receiver can demultiplex the incoming FEC stream by
>     SSRC and correlate it with the primary stream via the ssrc-group
>     mechanism.
>
> Given that FlexFEC has built in indication of which SSRC that are
> included in protection, why include it in SDP? For the purpose to
> indicate selectively which streams that should have FEC protection?
>
> You also need a reference for "ssrc-group" if intended to be supported
> going forward?
>

Agree. Left as a TODO for now until I can sort out the details.

>
> E. Section 5.2, first paragraph:
>
> This section needs clearer references to SDP.
>

Not sure exactly what you had in mind here.

>
> F. Section 6.
>
>     WebRTC also supports the ability to send generic application data,
>     and provides transport-level retransmission mechanisms that the
>     application can use to ensure that its data is delivered reliably.
>
>
> This should informationally reference data channel.
>

Done.

>
> G. Section 6.
>
> I think this section should say even less. Just note that SCTP has
> reliable and semi-reliable modes and the application can control it.
> Nothing more.
>

I don't see a problem with mentioning the app can do its own FEC if it
wants. I did adjust the first para to mention the reliability modes more
explicitly.

>
> H. Section 7.
>
>     To support the functionality recommended above, implementations MUST
>     support the redundant encoding mechanism described in [RFC2198]
>
> For which Codecs does on need to support this?
>

Added detail on this.

>
> I. Section 7.
>
> the FEC mechanism described in [RFC5956] and
>     [I-D.ietf-payload-flexible-fec-scheme]
>
> As RFC5956 is not a FEC mechanism, only a grouping mechanism I think
> this need to be reformulated.
>

Removed that part.

>
> J. Section 9.
>
> This section needs to be clearer that an implementation is expected to
> adapt the amount of FEC to the seen packet loss rate to ensure
> sufficient protection. And that adaptation must take into account the
> available network resources (Bandwidth) to trade-off media quality and
> what streams can be sent with use of FEC.
>

That's what this paragraph says, I think, but I rewrote it a bit to try to
make this more clear.

>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

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<div dir=3D"ltr">Finally got around to updating the document. Comments inli=
ne; new version of the doc attached.<br><br><div class=3D"gmail_quote"><div=
 dir=3D"ltr">On Wed, Nov 4, 2015 at 11:15 PM Magnus Westerlund &lt;<a href=
=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</=
a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
A. Section 3.2:<br>
<br>
I think the draft should note that some RTP Payload format, including<br>
several of the one recommended in<br>
draft-ietf-rtcweb-audio-codecs-for-interop-02, do support internal<br>
redundancy. The codecs that I know for a fact can do this within the<br>
payload format are: AMR and AMR-WB.<br>
<br>
I would suggest adding some sentence on that time delayed redundancy may<br=
>
be supported by codecs, and give AMR and AMR-WB as examples.<br></blockquot=
e><div><br></div><div>I added a section on AMR/AMR-WB that parallels the Op=
us section.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
B. Section 4.<br>
<br>
I think this draft should consider the recommended drafts in<br>
draft-ietf-rtcweb-audio-codecs-for-interop-02. Here we clearly can<br>
consider internal redundancy for AMR and AMR-WB.<br></blockquote><div><br><=
/div><div>Done.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
C. Section 4.2<br>
<br>
There are a parameter in AMR and AMR-WB that can negotiate how much<br>
redundancy one may include: max-red<br></blockquote><div><br></div><div>I m=
entioned this parameter in the doc. Do you want to recommend a default valu=
e?=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
<br>
D. Section 5.1:<br>
<br>
=C2=A0 =C2=A0 The receiver can demultiplex the incoming FEC stream by<br>
=C2=A0 =C2=A0 SSRC and correlate it with the primary stream via the ssrc-gr=
oup<br>
=C2=A0 =C2=A0 mechanism.<br>
<br>
Given that FlexFEC has built in indication of which SSRC that are<br>
included in protection, why include it in SDP? For the purpose to<br>
indicate selectively which streams that should have FEC protection?<br>
<br>
You also need a reference for &quot;ssrc-group&quot; if intended to be supp=
orted<br>
going forward?<br></blockquote><div><br></div><div>Agree. Left as a TODO fo=
r now until I can sort out the details.=C2=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">
<br>
E. Section 5.2, first paragraph:<br>
<br>
This section needs clearer references to SDP.<br></blockquote><div><br></di=
v><div>Not sure exactly what you had in mind here.=C2=A0</div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
<br>
F. Section 6.<br>
<br>
=C2=A0 =C2=A0 WebRTC also supports the ability to send generic application =
data,<br>
=C2=A0 =C2=A0 and provides transport-level retransmission mechanisms that t=
he<br>
=C2=A0 =C2=A0 application can use to ensure that its data is delivered reli=
ably.<br>
<br>
<br>
This should informationally reference data channel.<br></blockquote><div><b=
r></div><div>Done.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
G. Section 6.<br>
<br>
I think this section should say even less. Just note that SCTP has<br>
reliable and semi-reliable modes and the application can control it.<br>
Nothing more.<br></blockquote><div><br></div><div>I don&#39;t see a problem=
 with mentioning the app can do its own FEC if it wants. I did adjust the f=
irst para to mention the reliability modes more explicitly.=C2=A0</div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">
<br>
H. Section 7.<br>
<br>
=C2=A0 =C2=A0 To support the functionality recommended above, implementatio=
ns MUST<br>
=C2=A0 =C2=A0 support the redundant encoding mechanism described in [RFC219=
8]<br>
<br>
For which Codecs does on need to support this?<br></blockquote><div><br></d=
iv><div>Added detail on this.=C2=A0</div><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
I. Section 7.<br>
<br>
the FEC mechanism described in [RFC5956] and<br>
=C2=A0 =C2=A0 [I-D.ietf-payload-flexible-fec-scheme]<br>
<br>
As RFC5956 is not a FEC mechanism, only a grouping mechanism I think<br>
this need to be reformulated.<br></blockquote><div><br></div><div>Removed t=
hat part.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
J. Section 9.<br>
<br>
This section needs to be clearer that an implementation is expected to<br>
adapt the amount of FEC to the seen packet loss rate to ensure<br>
sufficient protection. And that adaptation must take into account the<br>
available network resources (Bandwidth) to trade-off media quality and<br>
what streams can be sent with use of FEC.<br></blockquote><div><br></div><d=
iv>That&#39;s what this paragraph says, I think, but I rewrote it a bit to =
try to make this more clear.=C2=A0</div><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =
Phone=C2=A0 +46 10 7148287<br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| Mobile +46 73 0949079<br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</blockquote></div></div>

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From: Justin Uberti <justin@uberti.name>
Date: Sat, 09 Jan 2016 00:21:10 +0000
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  draft-ietf-rtcweb-fec@ietf.org, Justin Uberti <juberti@google.com>
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[fixing wrong email address]

On Fri, Jan 8, 2016 at 4:19 PM Justin Uberti <justin@uberti.name> wrote:

> Finally got around to updating the document. Comments inline; new version
> of the doc attached.
>
> On Wed, Nov 4, 2015 at 11:15 PM Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
>
>> Hi,
>>
>> A. Section 3.2:
>>
>> I think the draft should note that some RTP Payload format, including
>> several of the one recommended in
>> draft-ietf-rtcweb-audio-codecs-for-interop-02, do support internal
>> redundancy. The codecs that I know for a fact can do this within the
>> payload format are: AMR and AMR-WB.
>>
>> I would suggest adding some sentence on that time delayed redundancy may
>> be supported by codecs, and give AMR and AMR-WB as examples.
>>
>
> I added a section on AMR/AMR-WB that parallels the Opus section.
>
>>
>> B. Section 4.
>>
>> I think this draft should consider the recommended drafts in
>> draft-ietf-rtcweb-audio-codecs-for-interop-02. Here we clearly can
>> consider internal redundancy for AMR and AMR-WB.
>>
>
> Done.
>
>>
>> C. Section 4.2
>>
>> There are a parameter in AMR and AMR-WB that can negotiate how much
>> redundancy one may include: max-red
>>
>
> I mentioned this parameter in the doc. Do you want to recommend a default
> value?
>
>>
>> D. Section 5.1:
>>
>>     The receiver can demultiplex the incoming FEC stream by
>>     SSRC and correlate it with the primary stream via the ssrc-group
>>     mechanism.
>>
>> Given that FlexFEC has built in indication of which SSRC that are
>> included in protection, why include it in SDP? For the purpose to
>> indicate selectively which streams that should have FEC protection?
>>
>> You also need a reference for "ssrc-group" if intended to be supported
>> going forward?
>>
>
> Agree. Left as a TODO for now until I can sort out the details.
>
>>
>> E. Section 5.2, first paragraph:
>>
>> This section needs clearer references to SDP.
>>
>
> Not sure exactly what you had in mind here.
>
>>
>> F. Section 6.
>>
>>     WebRTC also supports the ability to send generic application data,
>>     and provides transport-level retransmission mechanisms that the
>>     application can use to ensure that its data is delivered reliably.
>>
>>
>> This should informationally reference data channel.
>>
>
> Done.
>
>>
>> G. Section 6.
>>
>> I think this section should say even less. Just note that SCTP has
>> reliable and semi-reliable modes and the application can control it.
>> Nothing more.
>>
>
> I don't see a problem with mentioning the app can do its own FEC if it
> wants. I did adjust the first para to mention the reliability modes more
> explicitly.
>
>>
>> H. Section 7.
>>
>>     To support the functionality recommended above, implementations MUST
>>     support the redundant encoding mechanism described in [RFC2198]
>>
>> For which Codecs does on need to support this?
>>
>
> Added detail on this.
>
>>
>> I. Section 7.
>>
>> the FEC mechanism described in [RFC5956] and
>>     [I-D.ietf-payload-flexible-fec-scheme]
>>
>> As RFC5956 is not a FEC mechanism, only a grouping mechanism I think
>> this need to be reformulated.
>>
>
> Removed that part.
>
>>
>> J. Section 9.
>>
>> This section needs to be clearer that an implementation is expected to
>> adapt the amount of FEC to the seen packet loss rate to ensure
>> sufficient protection. And that adaptation must take into account the
>> available network resources (Bandwidth) to trade-off media quality and
>> what streams can be sent with use of FEC.
>>
>
> That's what this paragraph says, I think, but I rewrote it a bit to try t=
o
> make this more clear.
>
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>>

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<div dir=3D"ltr">[fixing wrong email address]<br><br><div class=3D"gmail_qu=
ote"><div dir=3D"ltr">On Fri, Jan 8, 2016 at 4:19 PM Justin Uberti &lt;<a h=
ref=3D"mailto:justin@uberti.name">justin@uberti.name</a>&gt; wrote:<br></di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">Finally got around to upd=
ating the document. Comments inline; new version of the doc attached.<br><b=
r><div class=3D"gmail_quote"></div></div><div dir=3D"ltr"><div class=3D"gma=
il_quote"><div dir=3D"ltr">On Wed, Nov 4, 2015 at 11:15 PM Magnus Westerlun=
d &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">m=
agnus.westerlund@ericsson.com</a>&gt; wrote:<br></div><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">Hi,<br>
<br>
A. Section 3.2:<br>
<br>
I think the draft should note that some RTP Payload format, including<br>
several of the one recommended in<br>
draft-ietf-rtcweb-audio-codecs-for-interop-02, do support internal<br>
redundancy. The codecs that I know for a fact can do this within the<br>
payload format are: AMR and AMR-WB.<br>
<br>
I would suggest adding some sentence on that time delayed redundancy may<br=
>
be supported by codecs, and give AMR and AMR-WB as examples.<br></blockquot=
e><div><br></div></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><d=
iv>I added a section on AMR/AMR-WB that parallels the Opus section.=C2=A0</=
div></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">
<br>
B. Section 4.<br>
<br>
I think this draft should consider the recommended drafts in<br>
draft-ietf-rtcweb-audio-codecs-for-interop-02. Here we clearly can<br>
consider internal redundancy for AMR and AMR-WB.<br></blockquote><div><br><=
/div></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><div>Done.=C2=
=A0</div></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">
<br>
C. Section 4.2<br>
<br>
There are a parameter in AMR and AMR-WB that can negotiate how much<br>
redundancy one may include: max-red<br></blockquote><div><br></div></div></=
div><div dir=3D"ltr"><div class=3D"gmail_quote"><div>I mentioned this param=
eter in the doc. Do you want to recommend a default value?=C2=A0</div></div=
></div><div dir=3D"ltr"><div class=3D"gmail_quote"><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">
<br>
D. Section 5.1:<br>
<br>
=C2=A0 =C2=A0 The receiver can demultiplex the incoming FEC stream by<br>
=C2=A0 =C2=A0 SSRC and correlate it with the primary stream via the ssrc-gr=
oup<br>
=C2=A0 =C2=A0 mechanism.<br>
<br>
Given that FlexFEC has built in indication of which SSRC that are<br>
included in protection, why include it in SDP? For the purpose to<br>
indicate selectively which streams that should have FEC protection?<br>
<br>
You also need a reference for &quot;ssrc-group&quot; if intended to be supp=
orted<br>
going forward?<br></blockquote><div><br></div></div></div><div dir=3D"ltr">=
<div class=3D"gmail_quote"><div>Agree. Left as a TODO for now until I can s=
ort out the details.=C2=A0</div></div></div><div dir=3D"ltr"><div class=3D"=
gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex">
<br>
E. Section 5.2, first paragraph:<br>
<br>
This section needs clearer references to SDP.<br></blockquote><div><br></di=
v></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><div>Not sure exa=
ctly what you had in mind here.=C2=A0</div></div></div><div dir=3D"ltr"><di=
v class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
F. Section 6.<br>
<br>
=C2=A0 =C2=A0 WebRTC also supports the ability to send generic application =
data,<br>
=C2=A0 =C2=A0 and provides transport-level retransmission mechanisms that t=
he<br>
=C2=A0 =C2=A0 application can use to ensure that its data is delivered reli=
ably.<br>
<br>
<br>
This should informationally reference data channel.<br></blockquote><div><b=
r></div></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><div>Done.=
=C2=A0</div></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">
<br>
G. Section 6.<br>
<br>
I think this section should say even less. Just note that SCTP has<br>
reliable and semi-reliable modes and the application can control it.<br>
Nothing more.<br></blockquote><div><br></div></div></div><div dir=3D"ltr"><=
div class=3D"gmail_quote"><div>I don&#39;t see a problem with mentioning th=
e app can do its own FEC if it wants. I did adjust the first para to mentio=
n the reliability modes more explicitly.=C2=A0</div></div></div><div dir=3D=
"ltr"><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
H. Section 7.<br>
<br>
=C2=A0 =C2=A0 To support the functionality recommended above, implementatio=
ns MUST<br>
=C2=A0 =C2=A0 support the redundant encoding mechanism described in [RFC219=
8]<br>
<br>
For which Codecs does on need to support this?<br></blockquote><div><br></d=
iv></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><div>Added detai=
l on this.=C2=A0</div></div></div><div dir=3D"ltr"><div class=3D"gmail_quot=
e"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">
<br>
I. Section 7.<br>
<br>
the FEC mechanism described in [RFC5956] and<br>
=C2=A0 =C2=A0 [I-D.ietf-payload-flexible-fec-scheme]<br>
<br>
As RFC5956 is not a FEC mechanism, only a grouping mechanism I think<br>
this need to be reformulated.<br></blockquote><div><br></div></div></div><d=
iv dir=3D"ltr"><div class=3D"gmail_quote"><div>Removed that part.=C2=A0</di=
v></div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<br>
J. Section 9.<br>
<br>
This section needs to be clearer that an implementation is expected to<br>
adapt the amount of FEC to the seen packet loss rate to ensure<br>
sufficient protection. And that adaptation must take into account the<br>
available network resources (Bandwidth) to trade-off media quality and<br>
what streams can be sent with use of FEC.<br></blockquote><div><br></div></=
div></div><div dir=3D"ltr"><div class=3D"gmail_quote"><div>That&#39;s what =
this paragraph says, I think, but I rewrote it a bit to try to make this mo=
re clear.=C2=A0</div></div></div><div dir=3D"ltr"><div class=3D"gmail_quote=
"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Services, Media and Network features, Ericsson Research EAB/TXM<br>
----------------------------------------------------------------------<br>
Ericsson AB=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =
Phone=C2=A0 +46 10 7148287<br>
F=C3=A4r=C3=B6gatan 6=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| Mobile +46 73 0949079<br>
SE-164 80 Stockholm, Sweden | mailto: <a href=3D"mailto:magnus.westerlund@e=
ricsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</blockquote></div></div></blockquote></div></div>

--047d7b2e12d9d8bc210528dbae50--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 8 Jan 2016 17:19:20 -0800
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Subject: Re: [rtcweb] JSEP: "VoiceActivityDetection" option and cient-mixer "vad" extension
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I don't think the VoiceActivityDetection setting would have any bearing on
the "vad" param. You could certainly imagine wanting to set the VAD bit in
client-mixer even when not using DTX.

IOW, I think that we should always send VAD in client-mixer. There's no
size penalty, and it's useful information for mixers.

On Thu, Jan 7, 2016 at 7:37 AM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> JSEP Section 5.2.3.4 discusses the implications of the
> "VoiceActivityDetection" option for the offer.
>
> One thing that is not covered is whether the "VoiceActivityDetection"
> option determines whether the client-mixer "vad" extension is set to "on"
> or "off".
>
> For example, I believe that the W3C WEBRTC WG determined that the
> client-mixer extension (support for which is required in RTP-Usage) should
> always be advertised in the offer.  RFC 6464 Section 4 gives an example of
> the a=extmap line:
>
> a=extmap:6 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on
>
>
> So, my question is whether "vad=on" or "vad=off" is determined by the "VoiceActivityDetection" option, or if not, whether there is a setting (e.g. "vad=on") that is always present in the offer.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">I don&#39;t think the VoiceActivityDetection setting would=
 have any bearing on the &quot;vad&quot; param. You could certainly imagine=
 wanting to set the VAD bit in client-mixer even when not using DTX.<div><b=
r></div><div>IOW, I think that we should always send VAD in client-mixer. T=
here&#39;s no size penalty, and it&#39;s useful information for mixers.</di=
v></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, J=
an 7, 2016 at 7:37 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt=
;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">JSEP Sec=
tion 5.2.3.4 discusses the implications of the &quot;VoiceActivityDetection=
&quot; option for the offer.=C2=A0<div><br></div><div>One thing that is not=
 covered is whether the &quot;VoiceActivityDetection&quot; option determine=
s whether the client-mixer &quot;vad&quot; extension is set to &quot;on&quo=
t; or &quot;off&quot;. =C2=A0</div><div><br></div><div>For example, I belie=
ve that the W3C WEBRTC WG determined that the client-mixer extension (suppo=
rt for which is required in RTP-Usage) should always be advertised in the o=
ffer.=C2=A0 RFC 6464 Section 4 gives an example of the a=3Dextmap line:=C2=
=A0</div><div><br></div><div><pre style=3D"font-size:13.3333px;margin-top:0=
px;margin-bottom:0px;color:rgb(0,0,0)">a=3Dextmap:6 urn:ietf:params:rtp-hdr=
ext:ssrc-audio-level vad=3Don</pre><pre style=3D"font-size:13.3333px;margin=
-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre style=3D"margin=
-top:0px;margin-bottom:0px"><font face=3D"arial, sans-serif"><span style=3D=
"white-space:normal">So, my question is whether &quot;vad=3Don&quot; or &qu=
ot;vad=3Doff&quot; is determined by the &quot;VoiceActivityDetection&quot; =
option, or if not, whether there is a setting (e.g. &quot;vad=3Don&quot;) t=
hat is always present in the offer.=C2=A0</span></font></pre></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

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Subject: Re: [rtcweb] JSEP: why is iceCandidatePoolSize an integer?
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[late to the party]

I'd be reluctant to add any more complexity to the operation of the
candidate pool - as Martin points out, it is already quite complex.

If you don't want to pregather all types of candidates, just don't use it.



On Sat, Dec 5, 2015 at 1:40 AM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 5 December 2015 at 04:29, Philipp Hancke <fippo@goodadvice.pages.de>
> wrote:
> > But if this is about components, how can the application make a decision
> > whether to pre-gather and hold on to stun or relay candidates (which
> > consumes resources)?
>
> The application might be able to defer the gathering of candidates via
> relays (I don't think that STUN is a real resource concern) by only
> providing the relays as stun URIs, or not providing them at all, then
> updating the iceServers with setConfiguration.
>
> I don't know if this is something that would work, we'd have to spec that.
>
> I'm actually beginning to question the utility of the pre-warming
> thing.  Maybe it's a performance gain, but it's certainly quite
> complex.
>
> >> No, this counts components.
> >
> >
> > Calling it iceCandidatePoolSize in the w3c spec seems odd to me then ;-)
>
> Yeah, I'm sure that - given it's current level of interoperability
> (i.e., zero) - the working group might be receptive to suggestions of
> a more accurate name.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
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<div dir=3D"ltr">[late to the party]<div><br></div><div>I&#39;d be reluctan=
t to add any more complexity to the operation of the candidate pool - as Ma=
rtin points out, it is already quite complex.</div><div><br></div><div>If y=
ou don&#39;t want to pregather all types of candidates, just don&#39;t use =
it.</div><div><br></div><div><br></div></div><div class=3D"gmail_extra"><br=
><div class=3D"gmail_quote">On Sat, Dec 5, 2015 at 1:40 AM, Martin Thomson =
<span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D=
"_blank">martin.thomson@gmail.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"><span class=3D"">On 5 December 2015 at 04:29, Philipp Hancke=
 &lt;<a href=3D"mailto:fippo@goodadvice.pages.de">fippo@goodadvice.pages.de=
</a>&gt; wrote:<br>
&gt; But if this is about components, how can the application make a decisi=
on<br>
&gt; whether to pre-gather and hold on to stun or relay candidates (which<b=
r>
&gt; consumes resources)?<br>
<br>
</span>The application might be able to defer the gathering of candidates v=
ia<br>
relays (I don&#39;t think that STUN is a real resource concern) by only<br>
providing the relays as stun URIs, or not providing them at all, then<br>
updating the iceServers with setConfiguration.<br>
<br>
I don&#39;t know if this is something that would work, we&#39;d have to spe=
c that.<br>
<br>
I&#39;m actually beginning to question the utility of the pre-warming<br>
thing.=C2=A0 Maybe it&#39;s a performance gain, but it&#39;s certainly quit=
e<br>
complex.<br>
<span class=3D""><br>
&gt;&gt; No, this counts components.<br>
&gt;<br>
&gt;<br>
&gt; Calling it iceCandidatePoolSize in the w3c spec seems odd to me then ;=
-)<br>
<br>
</span>Yeah, I&#39;m sure that - given it&#39;s current level of interopera=
bility<br>
(i.e., zero) - the working group might be receptive to suggestions of<br>
a more accurate name.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
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 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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From: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Proposed resolution on comfort noise
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I support this resolution.

To Felix's point, this text just says that WebRTC endpoints MUST support
CN, not MUST use it. Applications are free to not offer it (via the
VoiceActivityDetection parameter).

On Mon, Dec 14, 2015 at 4:31 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> I support this resolution.
>
> I note that there's material in this discussion for an extensive treatise
> on the pros and cons of offering CN at gateways. If someone wishes to
> extract this as a PR against the -gateways draft, the draft is at
> https://github.com/rtcweb-wg/gateways.
>
>
>
> On 12/10/2015 11:14 PM, Ted Hardie wrote:
>
> After reading the discussion on the comfort noise issue, it appears that
> one way forward might be to adjust this language:
>
> "comfort noise (CN).  Receivers MUST support RFC3389 CN for streams
> encoded with G.711 or any other supported codec that  does not provide it=
s
> own CN.=E2=80=9D
>
> to say instead
>
> "comfort noise (CN).  WebRTC endpoints MUST support RFC3389 CN for stream=
s
> encoded with G.711 or any other supported codec that  does not provide it=
s
> own CN.=E2=80=9D
>
> This would still allow non-WebRTC endpoints to omit CN support, while
> retaining the mandate for WebRTC endpoints.
>
> Does this resolve the issue?
>
> regards,
>
> Ted
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/r=
tcweb
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a114267483e9b7e0528dca45a
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<div dir=3D"ltr">I support this resolution.<div><br></div><div>To Felix&#39=
;s point, this text just says that WebRTC endpoints MUST support CN, not MU=
ST use it. Applications are free to not offer it (via the VoiceActivityDete=
ction parameter).</div></div><div class=3D"gmail_extra"><br><div class=3D"g=
mail_quote">On Mon, Dec 14, 2015 at 4:31 AM, Harald Alvestrand <span dir=3D=
"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@=
alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    <div>I support this resolution.<br>
      <br>
      I note that there&#39;s material in this discussion for an extensive
      treatise on the pros and cons of offering CN at gateways. If
      someone wishes to extract this as a PR against the -gateways
      draft, the draft is at <a href=3D"https://github.com/rtcweb-wg/gatewa=
ys" target=3D"_blank">https://github.com/rtcweb-wg/gateways</a>.<div><div c=
lass=3D"h5"><br>
      <br>
      <br>
      On 12/10/2015 11:14 PM, Ted Hardie wrote:<br>
    </div></div></div>
    <blockquote type=3D"cite"><div><div class=3D"h5">
      <div dir=3D"ltr">
        <div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small">After
          reading the discussion on the comfort noise issue, it appears
          that one way forward might be to adjust this language:<br>
          <br>
          &quot;comfort noise (CN).=C2=A0 Receivers MUST support RFC3389 CN=
 for
          streams encoded with G.711 or any other supported codec that
          =C2=A0does not provide its own CN.=E2=80=9D<br>
          <br>
          to say instead<br>
          <span></span><br>
          <span></span>&quot;comfort
          noise (CN).=C2=A0 WebRTC endpoints MUST support RFC3389 CN for
          streams encoded with G.711 or any other supported codec that
          =C2=A0does not provide its own CN.=E2=80=9D<br>
          <br>
        </div>
        <div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small">This
          would still allow non-WebRTC endpoints to omit CN support,
          while retaining the mandate for WebRTC endpoints.<br>
          <br>
        </div>
        <div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small">Does
          this resolve the issue?<br>
          <br>
        </div>
        <div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small">regards,<br>
          <br>
        </div>
        <div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif=
;font-size:small">Ted<br>
        </div>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><span class=3D""><pre>___________________________________=
____________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </span></blockquote><span class=3D"HOEnZb"><font color=3D"#888888">
    <br>
    <br>
    <pre cols=3D"72">--=20
Surveillance is pervasive. Go Dark.
</pre>
  </font></span></div>

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

--001a114267483e9b7e0528dca45a--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 8 Jan 2016 17:41:27 -0800
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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Unfortunately, that would contradict 5245. See
https://tools.ietf.org/html/rfc5245#section-4.3, 6th para.

Perhaps we could say that trickle with non-mux-exclusive should use
a=3Drtcp:10 or some such...

On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
>
>
> Section 5.2.1 of draft-jsep contains the following statement:
>
>
>
>    =E2=80=9Co  An "a=3Drtcp" line, as specified in [RFC3605], Section 2.1=
,
>
>       containing the dummy value "9 IN IP4 0.0.0.0", because no
>
>       candidates have yet been gathered.=E2=80=9D
>
>
>
> =E2=80=A6i.e. the same port:address as in the associated c/m-line.
>
>
>
> The problem is that the currently suggested mechanism to indicate
> exclusive support of RTP/RTCP-mux also says that the a=3Drtcp line shall =
be
> identical to the c/m-line.
>
>
>
> My question is: is there really a need to always include the rtcp
> attribute in trickle offers? Isn=E2=80=99t it enough to indicate port 9 i=
n the m-
> line, and address 0.0.0.0 in the c-line? In my opinion that would also
> apply to RTCP.
>
>
>
> Example:
>
>
>
> JSEP trickle offer **without** mux-exclusive indication:
>
>
>
> c=3DIN IP4 0.0.0.0
>
> m=3Daudio 9=E2=80=A6
>
> a=3Drtcp-mux
>
>
>
> JSEP trickle offer **with** mux-exclusive indication:
>
>
>
> c=3DIN IP4 0.0.0.0
>
> m=3Daudio 9=E2=80=A6
>
>               *a=3Drtcp: 9 IN IP4 0.0.0.0*
>
>               a=3Drtcp-mux
>
>
>
> Regards,
>
>
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a1143ff3c93a12a0528dcce40
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<div dir=3D"ltr">Unfortunately, that would contradict 5245. See=C2=A0<a hre=
f=3D"https://tools.ietf.org/html/rfc5245#section-4.3">https://tools.ietf.or=
g/html/rfc5245#section-4.3</a>, 6th para.<div><br></div><div>Perhaps we cou=
ld say that trickle with non-mux-exclusive should use a=3Drtcp:10 or some s=
uch...</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmberg <span dir=3D"ltr">&lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer=
.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">





<div lang=3D"EN-GB" link=3D"#0563C1" vlink=3D"#954F72">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Section 5.2.1 of draft-jsep contains the following s=
tatement:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0 =E2=80=9Co =C2=A0An &quot;a=3Drtcp&quot=
; line, as specified in [RFC3605], Section 2.1,<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 containing the dummy =
value &quot;9 IN IP4 0.0.0.0&quot;, because no<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 candidates have yet b=
een gathered.=E2=80=9D<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">=E2=80=A6i.e. the same port:address as in the associ=
ated c/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">The problem is that the currently suggested mechanis=
m to indicate exclusive support of RTP/RTCP-mux also says that the a=3Drtcp=
 line shall be identical to the c/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">My question is: is there really a need to always inc=
lude the rtcp attribute in trickle offers? Isn=E2=80=99t it enough to indic=
ate port 9 in the m- line, and address 0.0.0.0 in the c-line? In my opinion=
 that would also apply to RTCP.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Example:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>without</b>* mux-exclusive in=
dication:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=E2=80=A6<u>=
</u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">a=3Drtcp-mux<u></u><u><=
/u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>with</b>* mux-exclusive indic=
ation:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=E2=80=A6<u>=
</u><u></u></p>
<p class=3D"MsoNormal">=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 <b>a=3Drtcp: 9 IN IP4 0.0.0.0<u></u><u></u></b>=
</p>
<p class=3D"MsoNormal">=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 a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Regards,<span class=3D"HOEnZb"><font color=3D"#88888=
8"><u></u><u></u></font></span></p><span class=3D"HOEnZb"><font color=3D"#8=
88888">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
</font></span></div>
</div>

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

--001a1143ff3c93a12a0528dcce40--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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Hi,

Eventhough the text in 5245 is a little unclear (should probably be fixed i=
n 5245bis) I don't think it says that you must include the rtcp attribute -=
 it describes how to encode the rtcp candidate.

Regards,

Christer


Sent from my Windows Phone
________________________________
From: Justin Uberti<mailto:juberti@google.com>
Sent: =FD09/=FD01/=FD2016 03:41
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP: Usage of a=3Drtcp attribute

Unfortunately, that would contradict 5245. See https://tools.ietf.org/html/=
rfc5245#section-4.3, 6th para.

Perhaps we could say that trickle with non-mux-exclusive should use a=3Drtc=
p:10 or some such...

On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

Section 5.2.1 of draft-jsep contains the following statement:

   =93o  An "a=3Drtcp" line, as specified in [RFC3605], Section 2.1,
      containing the dummy value "9 IN IP4 0.0.0.0", because no
      candidates have yet been gathered.=94

=85i.e. the same port:address as in the associated c/m-line.

The problem is that the currently suggested mechanism to indicate exclusive=
 support of RTP/RTCP-mux also says that the a=3Drtcp line shall be identica=
l to the c/m-line.

My question is: is there really a need to always include the rtcp attribute=
 in trickle offers? Isn=92t it enough to indicate port 9 in the m- line, an=
d address 0.0.0.0 in the c-line? In my opinion that would also apply to RTC=
P.

Example:

JSEP trickle offer *without* mux-exclusive indication:

c=3DIN IP4 0.0.0.0
m=3Daudio 9=85
a=3Drtcp-mux

JSEP trickle offer *with* mux-exclusive indication:

c=3DIN IP4 0.0.0.0
m=3Daudio 9=85
              a=3Drtcp: 9 IN IP4 0.0.0.0
              a=3Drtcp-mux

Regards,

Christer

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



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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
Eventhough the text in 5245 is a little unclear (should probably be fixed i=
n 5245bis) I don't think it says that you must include the rtcp attribute -=
 it describes how to encode the rtcp candidate.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:juberti@google.com">Justin Uberti</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD09=
/=FD01/=FD2016 03:41</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] JSEP: Usage of a=3Drtcp attribute</span><br>
<br>
</div>
<div>
<div dir=3D"ltr">Unfortunately, that would contradict 5245. See&nbsp;<a hre=
f=3D"https://tools.ietf.org/html/rfc5245#section-4.3">https://tools.ietf.or=
g/html/rfc5245#section-4.3</a>, 6th para.
<div><br>
</div>
<div>Perhaps we could say that trickle with non-mux-exclusive should use a=
=3Drtcp:10 or some such...</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div lang=3D"EN-GB">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Section 5.2.1 of draft-jsep contains the following s=
tatement:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; =93o &nbsp;An &quot;a=3Drtcp&quot; line=
, as specified in [RFC3605], Section 2.1,<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; containing the dummy =
value &quot;9 IN IP4 0.0.0.0&quot;, because no<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; candidates have yet b=
een gathered.=94<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">=85i.e. the same port:address as in the associated c=
/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">The problem is that the currently suggested mechanis=
m to indicate exclusive support of RTP/RTCP-mux also says that the a=3Drtcp=
 line shall be identical to the c/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">My question is: is there really a need to always inc=
lude the rtcp attribute in trickle offers? Isn=92t it enough to indicate po=
rt 9 in the m- line, and address 0.0.0.0 in the c-line? In my opinion that =
would also apply to RTCP.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Example:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>without</b>* mux-exclusive in=
dication:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=85<u></u><u=
></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">a=3Drtcp-mux<u></u><u><=
/u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>with</b>* mux-exclusive indic=
ation:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=85<u></u><u=
></u></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; <b>a=3Drtcp: 9 IN IP4 0.0.0.0<u></u><u></u></b><=
/p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Regards,<span class=3D"HOEnZb"><font color=3D"#88888=
8"><u></u><u></u></font></span></p>
<span class=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
</font></span></div>
</div>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</body>
</html>

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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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Hi,

To add to above, if default RTCP candidate is on the same address as RTP
and is using RTP port+1, why is SDP rtcp attribute is still required? It is
totally redundant. Does anybody know the reason why rtcp SDP attribute was
made a requirement in RFC 5245?

Regards,

_____________
Roman Shpount

On Sat, Jan 9, 2016 at 12:32 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> Eventhough the text in 5245 is a little unclear (should probably be fixed
> in 5245bis) I don't think it says that you must include the rtcp attribut=
e
> - it describes how to encode the rtcp candidate.
>
> Regards,
>
> Christer
>
>
> Sent from my Windows Phone
> ------------------------------
> From: Justin Uberti <juberti@google.com>
> Sent: =E2=80=8E09/=E2=80=8E01/=E2=80=8E2016 03:41
> To: Christer Holmberg <christer.holmberg@ericsson.com>
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] JSEP: Usage of a=3Drtcp attribute
>
> Unfortunately, that would contradict 5245. See
> https://tools.ietf.org/html/rfc5245#section-4.3, 6th para.
>
> Perhaps we could say that trickle with non-mux-exclusive should use
> a=3Drtcp:10 or some such...
>
> On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>>
>>
>> Section 5.2.1 of draft-jsep contains the following statement:
>>
>>
>>
>>    =E2=80=9Co  An "a=3Drtcp" line, as specified in [RFC3605], Section 2.=
1,
>>
>>       containing the dummy value "9 IN IP4 0.0.0.0", because no
>>
>>       candidates have yet been gathered.=E2=80=9D
>>
>>
>>
>> =E2=80=A6i.e. the same port:address as in the associated c/m-line.
>>
>>
>>
>> The problem is that the currently suggested mechanism to indicate
>> exclusive support of RTP/RTCP-mux also says that the a=3Drtcp line shall=
 be
>> identical to the c/m-line.
>>
>>
>>
>> My question is: is there really a need to always include the rtcp
>> attribute in trickle offers? Isn=E2=80=99t it enough to indicate port 9 =
in the m-
>> line, and address 0.0.0.0 in the c-line? In my opinion that would also
>> apply to RTCP.
>>
>>
>>
>> Example:
>>
>>
>>
>> JSEP trickle offer **without** mux-exclusive indication:
>>
>>
>>
>> c=3DIN IP4 0.0.0.0
>>
>> m=3Daudio 9=E2=80=A6
>>
>> a=3Drtcp-mux
>>
>>
>>
>> JSEP trickle offer **with** mux-exclusive indication:
>>
>>
>>
>> c=3DIN IP4 0.0.0.0
>>
>> m=3Daudio 9=E2=80=A6
>>
>>               *a=3Drtcp: 9 IN IP4 0.0.0.0*
>>
>>               a=3Drtcp-mux
>>
>>
>>
>> Regards,
>>
>>
>>
>> Christer
>>
>> _______________________________________________
>> 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
>
>

--047d7b10ca09ffaa460528f34cc6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi,<div><br></div><div>To add to above, if default RTCP ca=
ndidate is on the same address as RTP and is using RTP port+1, why is SDP r=
tcp attribute is still required? It is totally redundant. Does anybody know=
 the reason why rtcp SDP attribute was made a requirement in RFC=C2=A0<span=
 style=3D"color:rgb(0,0,0);font-size:12.8px">5245</span>?</div><div><br></d=
iv><div>Regards,</div></div><div class=3D"gmail_extra"><br clear=3D"all"><d=
iv><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></div=
>
<br><div class=3D"gmail_quote">On Sat, Jan 9, 2016 at 12:32 AM, Christer Ho=
lmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">



<div>
<div>
<div style=3D"font-family:Calibri,sans-serif;font-size:11pt">Hi,<br>
<br>
Eventhough the text in 5245 is a little unclear (should probably be fixed i=
n 5245bis) I don&#39;t think it says that you must include the rtcp attribu=
te - it describes how to encode the rtcp candidate.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">From:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:juberti@google.com" target=3D"_blank">Justin Uberti</a></span><=
br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Sent:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">=E2=80=
=8E09/=E2=80=8E01/=E2=80=8E2016 03:41</span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">To:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">Christer Holm=
berg</a></span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Cc:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Subject:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">Re: [r=
tcweb] JSEP: Usage of a=3Drtcp attribute</span><br>
<br>
</div><div><div class=3D"h5">
<div>
<div dir=3D"ltr">Unfortunately, that would contradict 5245. See=C2=A0<a hre=
f=3D"https://tools.ietf.org/html/rfc5245#section-4.3" target=3D"_blank">htt=
ps://tools.ietf.org/html/rfc5245#section-4.3</a>, 6th para.
<div><br>
</div>
<div>Perhaps we could say that trickle with non-mux-exclusive should use a=
=3Drtcp:10 or some such...</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div lang=3D"EN-GB">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Section 5.2.1 of draft-jsep contains the following s=
tatement:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0 =E2=80=9Co =C2=A0An &quot;a=3Drtcp&quot=
; line, as specified in [RFC3605], Section 2.1,<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 containing the dummy =
value &quot;9 IN IP4 0.0.0.0&quot;, because no<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 candidates have yet b=
een gathered.=E2=80=9D<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">=E2=80=A6i.e. the same port:address as in the associ=
ated c/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">The problem is that the currently suggested mechanis=
m to indicate exclusive support of RTP/RTCP-mux also says that the a=3Drtcp=
 line shall be identical to the c/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">My question is: is there really a need to always inc=
lude the rtcp attribute in trickle offers? Isn=E2=80=99t it enough to indic=
ate port 9 in the m- line, and address 0.0.0.0 in the c-line? In my opinion=
 that would also apply to RTCP.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Example:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>without</b>* mux-exclusive in=
dication:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=E2=80=A6<u>=
</u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">a=3Drtcp-mux<u></u><u><=
/u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>with</b>* mux-exclusive indic=
ation:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=E2=80=A6<u>=
</u><u></u></p>
<p class=3D"MsoNormal">=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 <b>a=3Drtcp: 9 IN IP4 0.0.0.0<u></u><u></u></b>=
</p>
<p class=3D"MsoNormal">=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 a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Regards,<span><font color=3D"#888888"><u></u><u></u>=
</font></span></p>
<span><font color=3D"#888888">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
</font></span></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div></div></div>

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

--047d7b10ca09ffaa460528f34cc6--


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I agree with Justin. Remember that this is an optimization and it only
impacts
the *timing* of the delivery of the candidates. If we get to the point
where w
find that this wasn't enough control, we should be able deal with that
later i
a backward compatible way.

-Ekr


On Fri, Jan 8, 2016 at 5:22 PM, Justin Uberti <juberti@google.com> wrote:

> [late to the party]
>
> I'd be reluctant to add any more complexity to the operation of the
> candidate pool - as Martin points out, it is already quite complex.
>
> If you don't want to pregather all types of candidates, just don't use it.
>
>
>
> On Sat, Dec 5, 2015 at 1:40 AM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>
>> On 5 December 2015 at 04:29, Philipp Hancke <fippo@goodadvice.pages.de>
>> wrote:
>> > But if this is about components, how can the application make a decision
>> > whether to pre-gather and hold on to stun or relay candidates (which
>> > consumes resources)?
>>
>> The application might be able to defer the gathering of candidates via
>> relays (I don't think that STUN is a real resource concern) by only
>> providing the relays as stun URIs, or not providing them at all, then
>> updating the iceServers with setConfiguration.
>>
>> I don't know if this is something that would work, we'd have to spec that.
>>
>> I'm actually beginning to question the utility of the pre-warming
>> thing.  Maybe it's a performance gain, but it's certainly quite
>> complex.
>>
>> >> No, this counts components.
>> >
>> >
>> > Calling it iceCandidatePoolSize in the w3c spec seems odd to me then ;-)
>>
>> Yeah, I'm sure that - given it's current level of interoperability
>> (i.e., zero) - the working group might be receptive to suggestions of
>> a more accurate name.
>>
>> _______________________________________________
>> 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
>
>

--94eb2c08192a4d2b760528fe16e0
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<div dir=3D"ltr">I agree with Justin. Remember that this is an optimization=
 and it only impacts<div>the *timing* of the delivery of the candidates. If=
 we get to the point where w</div><div>find that this wasn&#39;t enough con=
trol, we should be able deal with that later i</div><div>a backward compati=
ble way.</div><div><br></div><div><div><div>-Ekr</div><div><br></div><div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jan 8, 201=
6 at 5:22 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti=
@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">[late to the party]<div><br=
></div><div>I&#39;d be reluctant to add any more complexity to the operatio=
n of the candidate pool - as Martin points out, it is already quite complex=
.</div><div><br></div><div>If you don&#39;t want to pregather all types of =
candidates, just don&#39;t use it.</div><div><br></div><div><br></div></div=
><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><di=
v class=3D"gmail_quote">On Sat, Dec 5, 2015 at 1:40 AM, Martin Thomson <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_bl=
ank">martin.thomson@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"><span>On 5 December 2015 at 04:29, Philipp Hancke &lt;<a href=3D=
"mailto:fippo@goodadvice.pages.de" target=3D"_blank">fippo@goodadvice.pages=
.de</a>&gt; wrote:<br>
&gt; But if this is about components, how can the application make a decisi=
on<br>
&gt; whether to pre-gather and hold on to stun or relay candidates (which<b=
r>
&gt; consumes resources)?<br>
<br>
</span>The application might be able to defer the gathering of candidates v=
ia<br>
relays (I don&#39;t think that STUN is a real resource concern) by only<br>
providing the relays as stun URIs, or not providing them at all, then<br>
updating the iceServers with setConfiguration.<br>
<br>
I don&#39;t know if this is something that would work, we&#39;d have to spe=
c that.<br>
<br>
I&#39;m actually beginning to question the utility of the pre-warming<br>
thing.=C2=A0 Maybe it&#39;s a performance gain, but it&#39;s certainly quit=
e<br>
complex.<br>
<span><br>
&gt;&gt; No, this counts components.<br>
&gt;<br>
&gt;<br>
&gt; Calling it iceCandidatePoolSize in the w3c spec seems odd to me then ;=
-)<br>
<br>
</span>Yeah, I&#39;m sure that - given it&#39;s current level of interopera=
bility<br>
(i.e., zero) - the working group might be receptive to suggestions of<br>
a more accurate name.<br>
<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div></div></div></div>

--94eb2c08192a4d2b760528fe16e0--


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From: Justin Uberti <juberti@google.com>
Date: Sun, 10 Jan 2016 14:24:37 -0800
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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I think the text is pretty clear that you need to add an a=3Drtcp attribute=
:

   The default candidates are added to the SDP as the default
   destination for media.  For streams based on RTP, this is done by
   placing the IP address and port of the RTP candidate into the c and m
   lines, respectively.  *If the agent is utilizing RTCP, it MUST encode
   the RTCP candidate using the a=3Drtcp attribute as defined in RFC
3605 <https://tools.ietf.org/html/rfc3605>
   [RFC3605 <https://tools.ietf.org/html/rfc3605>].*  If RTCP is not
in use, the agent MUST signal that using
   b=3DRS:0 and b=3DRR:0 as defined in RFC 3556
<https://tools.ietf.org/html/rfc3556> [RFC3556
<https://tools.ietf.org/html/rfc3556>].


On Fri, Jan 8, 2016 at 9:32 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> Eventhough the text in 5245 is a little unclear (should probably be fixed
> in 5245bis) I don't think it says that you must include the rtcp attribut=
e
> - it describes how to encode the rtcp candidate.
>
> Regards,
>
> Christer
>
>
> Sent from my Windows Phone
> ------------------------------
> From: Justin Uberti <juberti@google.com>
> Sent: =E2=80=8E09/=E2=80=8E01/=E2=80=8E2016 03:41
> To: Christer Holmberg <christer.holmberg@ericsson.com>
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] JSEP: Usage of a=3Drtcp attribute
>
> Unfortunately, that would contradict 5245. See
> https://tools.ietf.org/html/rfc5245#section-4.3, 6th para.
>
> Perhaps we could say that trickle with non-mux-exclusive should use
> a=3Drtcp:10 or some such...
>
> On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>>
>>
>> Section 5.2.1 of draft-jsep contains the following statement:
>>
>>
>>
>>    =E2=80=9Co  An "a=3Drtcp" line, as specified in [RFC3605], Section 2.=
1,
>>
>>       containing the dummy value "9 IN IP4 0.0.0.0", because no
>>
>>       candidates have yet been gathered.=E2=80=9D
>>
>>
>>
>> =E2=80=A6i.e. the same port:address as in the associated c/m-line.
>>
>>
>>
>> The problem is that the currently suggested mechanism to indicate
>> exclusive support of RTP/RTCP-mux also says that the a=3Drtcp line shall=
 be
>> identical to the c/m-line.
>>
>>
>>
>> My question is: is there really a need to always include the rtcp
>> attribute in trickle offers? Isn=E2=80=99t it enough to indicate port 9 =
in the m-
>> line, and address 0.0.0.0 in the c-line? In my opinion that would also
>> apply to RTCP.
>>
>>
>>
>> Example:
>>
>>
>>
>> JSEP trickle offer **without** mux-exclusive indication:
>>
>>
>>
>> c=3DIN IP4 0.0.0.0
>>
>> m=3Daudio 9=E2=80=A6
>>
>> a=3Drtcp-mux
>>
>>
>>
>> JSEP trickle offer **with** mux-exclusive indication:
>>
>>
>>
>> c=3DIN IP4 0.0.0.0
>>
>> m=3Daudio 9=E2=80=A6
>>
>>               *a=3Drtcp: 9 IN IP4 0.0.0.0*
>>
>>               a=3Drtcp-mux
>>
>>
>>
>> Regards,
>>
>>
>>
>> Christer
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

--001a1142674857b73b0529024a20
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<div dir=3D"ltr">I think the text is pretty clear that you need to add an a=
=3Drtcp attribute:<div><br></div><div><pre class=3D"" style=3D"font-size:13=
.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   The default c=
andidates are added to the SDP as the default
   destination for media.  For streams based on RTP, this is done by
   placing the IP address and port of the RTP candidate into the c and m
   lines, respectively.  <b>If the agent is utilizing RTCP, it MUST encode
   the RTCP candidate using the a=3Drtcp attribute as defined in <a href=3D=
"https://tools.ietf.org/html/rfc3605">RFC 3605</a>
   [<a href=3D"https://tools.ietf.org/html/rfc3605" title=3D"&quot;Real Tim=
e Control Protocol (RTCP) attribute in Session Description Protocol (SDP)&q=
uot;">RFC3605</a>].</b>  If RTCP is not in use, the agent MUST signal that =
using
   b=3DRS:0 and b=3DRR:0 as defined in <a href=3D"https://tools.ietf.org/ht=
ml/rfc3556">RFC 3556</a> [<a href=3D"https://tools.ietf.org/html/rfc3556" t=
itle=3D"&quot;Session Description Protocol (SDP) Bandwidth Modifiers for RT=
P Control Protocol (RTCP) Bandwidth&quot;">RFC3556</a>].</pre></div></div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jan 8, 201=
6 at 9:32 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:chr=
ister.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.c=
om</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">



<div>
<div>
<div style=3D"font-family:Calibri,sans-serif;font-size:11pt">Hi,<br>
<br>
Eventhough the text in 5245 is a little unclear (should probably be fixed i=
n 5245bis) I don&#39;t think it says that you must include the rtcp attribu=
te - it describes how to encode the rtcp candidate.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">From:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:juberti@google.com" target=3D"_blank">Justin Uberti</a></span><=
br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Sent:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">=E2=80=
=8E09/=E2=80=8E01/=E2=80=8E2016 03:41</span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">To:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">Christer Holm=
berg</a></span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Cc:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Subject:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">Re: [r=
tcweb] JSEP: Usage of a=3Drtcp attribute</span><br>
<br>
</div><div><div class=3D"h5">
<div>
<div dir=3D"ltr">Unfortunately, that would contradict 5245. See=C2=A0<a hre=
f=3D"https://tools.ietf.org/html/rfc5245#section-4.3" target=3D"_blank">htt=
ps://tools.ietf.org/html/rfc5245#section-4.3</a>, 6th para.
<div><br>
</div>
<div>Perhaps we could say that trickle with non-mux-exclusive should use a=
=3Drtcp:10 or some such...</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Sun, Dec 13, 2015 at 7:02 AM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div lang=3D"EN-GB">
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Section 5.2.1 of draft-jsep contains the following s=
tatement:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0 =E2=80=9Co =C2=A0An &quot;a=3Drtcp&quot=
; line, as specified in [RFC3605], Section 2.1,<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 containing the dummy =
value &quot;9 IN IP4 0.0.0.0&quot;, because no<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 candidates have yet b=
een gathered.=E2=80=9D<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">=E2=80=A6i.e. the same port:address as in the associ=
ated c/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">The problem is that the currently suggested mechanis=
m to indicate exclusive support of RTP/RTCP-mux also says that the a=3Drtcp=
 line shall be identical to the c/m-line.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">My question is: is there really a need to always inc=
lude the rtcp attribute in trickle offers? Isn=E2=80=99t it enough to indic=
ate port 9 in the m- line, and address 0.0.0.0 in the c-line? In my opinion=
 that would also apply to RTCP.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Example:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>without</b>* mux-exclusive in=
dication:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=E2=80=A6<u>=
</u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">a=3Drtcp-mux<u></u><u><=
/u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">JSEP trickle offer *<b>with</b>* mux-exclusive indic=
ation:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">c=3DIN IP4 0.0.0.0 <u><=
/u><u></u></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt">m=3Daudio 9=E2=80=A6<u>=
</u><u></u></p>
<p class=3D"MsoNormal">=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 <b>a=3Drtcp: 9 IN IP4 0.0.0.0<u></u><u></u></b>=
</p>
<p class=3D"MsoNormal">=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 a=3Drtcp-mux<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Regards,<span><font color=3D"#888888"><u></u><u></u>=
</font></span></p>
<span><font color=3D"#888888">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
</font></span></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div></div></div>

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

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] JSEP: Usage of a=rtcp attribute
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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On Mon, Jan 11, 2016 at 7:35 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> I think we could clarify that the attribute is used only if there is a
> real port, and if something is to be sent.
>
>
>
This most likely be a change to RFC 5245 not a clarification. It can
definitely be proposed for RFC 5245 bis to ice working group.
_____________
Roman Shpount

--089e01538448af4bab052910c929
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Jan 11, 2016 at 7:35 AM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br></div></div><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex">





<div lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125);font-family:Cali=
bri,sans-serif;font-size:11pt">I think we could clarify that the attribute =
is used only if there is a real port, and if something is to be sent.</span=
><br></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>This most likely be a change to RFC 5245 not a clarification. It can defin=
itely be proposed for RFC 5245 bis to ice working group.</div><div><div cla=
ss=3D"gmail_signature">_____________<br>Roman Shpount</div></div><div>=C2=
=A0</div></div></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] JSEP: Usage of a=rtcp attribute
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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Hi,

If we put it in 5245bis I assume JSEP (and trickle ICE) would have to refer=
ence 5245bis. Currently I think both of those specs reference 5245, but I m=
ay be wrong.

It can be discussed whether it's a change or clarification. 5245 says that =
the RTCP candidate shall be encoded using the rtcp attribute - but if you a=
re doing trickle there is no candidate to encode.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Roman Shpount<mailto:roman@telurix.com>
Sent: =FD11/=FD01/=FD2016 17:42
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
Cc: Justin Uberti<mailto:juberti@google.com>; rtcweb@ietf.org<mailto:rtcweb=
@ietf.org>
Subject: Re: [rtcweb] JSEP: Usage of a=3Drtcp attribute

On Mon, Jan 11, 2016 at 7:35 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
I think we could clarify that the attribute is used only if there is a real=
 port, and if something is to be sent.


This most likely be a change to RFC 5245 not a clarification. It can defini=
tely be proposed for RFC 5245 bis to ice working group.
_____________
Roman Shpount


--_000_7594FB04B1934943A5C02806D1A2204B37D1240DESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
If we put it in 5245bis I assume JSEP (and trickle ICE) would have to refer=
ence 5245bis. Currently I think both of those specs reference 5245, but I m=
ay be wrong.<br>
<br>
It can be discussed whether it's a change or clarification. 5245 says that =
the RTCP candidate shall be encoded using the rtcp attribute - but if you a=
re doing trickle there is no candidate to encode.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD11=
/=FD01/=FD2016 17:42</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:juberti@google.com">Justin Uberti</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] JSEP: Usage of a=3Drtcp attribute</span><br>
<br>
</div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div>
<div class=3D"gmail_signature">On Mon, Jan 11, 2016 at 7:35 AM, Christer Ho=
lmberg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
</div>
</div>
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex; border=
-left-width:1px; border-left-color:rgb(204,204,204); border-left-style:soli=
d; padding-left:1ex">
<div lang=3D"FI">
<div>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125); font-family:Cal=
ibri,sans-serif; font-size:11pt">I think we could clarify that the attribut=
e is used only if there is a real port, and if something is to be sent.</sp=
an><br>
</p>
<p class=3D"MsoNormal"><br>
</p>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>This most likely be a change to RFC 5245 not a clarification. It can d=
efinitely be proposed for RFC 5245 bis to ice working group.</div>
<div>
<div class=3D"gmail_signature">_____________<br>
Roman Shpount</div>
</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</body>
</html>

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References: <7594FB04B1934943A5C02806D1A2204B37CC9AF8@ESESSMB209.ericsson.se> <CAOJ7v-2_=cfayWcJQ6vo0ejL-b2U5OAR=DmSszVMgjXspeoGqg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37D1020D@ESESSMB209.ericsson.se> <CAOJ7v-0hMOO_HFuP+oKhf8fJ2d2rkDKz9uYZFUFn7wUXKhbcug@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37D11DAE@ESESSMB209.ericsson.se> <CAD5OKxv6LrNddpQDVgvtQryz2HoGG=xrkdCSbD=sgExknSee-Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37D1240D@ESESSMB209.ericsson.se>
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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Having the attribute indicates that the device is capable of rtcp on a =
port other than the rtp+1 port=20


> On Jan 11, 2016, at 9:12 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi,
>=20
> If we put it in 5245bis I assume JSEP (and trickle ICE) would have to =
reference 5245bis. Currently I think both of those specs reference 5245, =
but I may be wrong.
>=20
> It can be discussed whether it's a change or clarification. 5245 says =
that the RTCP candidate shall be encoded using the rtcp attribute - but =
if you are doing trickle there is no candidate to encode.
>=20
> Regards,
>=20
> Christer
>=20
> Sent from my Windows Phone
> From: Roman Shpount
> Sent: =E2=80=8E11/=E2=80=8E01/=E2=80=8E2016 17:42
> To: Christer Holmberg
> Cc: Justin Uberti; rtcweb@ietf.org
> Subject: Re: [rtcweb] JSEP: Usage of a=3Drtcp attribute
>=20
> On Mon, Jan 11, 2016 at 7:35 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> I think we could clarify that the attribute is used only if there is a =
real port, and if something is to be sent.
>=20
>=20
> This most likely be a change to RFC 5245 not a clarification. It can =
definitely be proposed for RFC 5245 bis to ice working group.
> _____________
> Roman Shpount
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] JSEP: why is iceCandidatePoolSize an integer?
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+1

I think that most people that want to use the pre-gather can make a =
reasonable guess of how many candidates they need to get "quickly" (to =
get the main call up). If it takes a bit longer for all the variable =
things such as thumb nails, that does not impact the UX in the same way.


> On Jan 10, 2016, at 10:23 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
> I agree with Justin. Remember that this is an optimization and it only =
impacts
> the *timing* of the delivery of the candidates. If we get to the point =
where w
> find that this wasn't enough control, we should be able deal with that =
later i
> a backward compatible way.
>=20
> -Ekr
>=20
>=20
> On Fri, Jan 8, 2016 at 5:22 PM, Justin Uberti <juberti@google.com> =
wrote:
> [late to the party]
>=20
> I'd be reluctant to add any more complexity to the operation of the =
candidate pool - as Martin points out, it is already quite complex.
>=20
> If you don't want to pregather all types of candidates, just don't use =
it.
>=20
>=20
>=20
> On Sat, Dec 5, 2015 at 1:40 AM, Martin Thomson =
<martin.thomson@gmail.com> wrote:
> On 5 December 2015 at 04:29, Philipp Hancke =
<fippo@goodadvice.pages.de> wrote:
> > But if this is about components, how can the application make a =
decision
> > whether to pre-gather and hold on to stun or relay candidates (which
> > consumes resources)?
>=20
> The application might be able to defer the gathering of candidates via
> relays (I don't think that STUN is a real resource concern) by only
> providing the relays as stun URIs, or not providing them at all, then
> updating the iceServers with setConfiguration.
>=20
> I don't know if this is something that would work, we'd have to spec =
that.
>=20
> I'm actually beginning to question the utility of the pre-warming
> thing.  Maybe it's a performance gain, but it's certainly quite
> complex.
>=20
> >> No, this counts components.
> >
> >
> > Calling it iceCandidatePoolSize in the w3c spec seems odd to me then =
;-)
>=20
> Yeah, I'm sure that - given it's current level of interoperability
> (i.e., zero) - the working group might be receptive to suggestions of
> a more accurate name.
>=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
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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To: Bernard Aboba <bernard.aboba@gmail.com>
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Subject: Re: [rtcweb] JSEP: "VoiceActivityDetection" option and cient-mixer "vad" extension
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Bernard,=20

Two things....

To answer your questions ... I think the controll of the V bit in =
rfc6464 is totally indepneded of this VoiceActivityDetection option in =
JS.=20

We  probably need a PR for jsep to clarify this. The key thing to me is =
that if the VoiceActivityDetection is set to false by the JS, the =
browser MUST NOT do silence suppressions but must send all the audio for =
all packets like we would need for cases such as an E911 call or =
detecting modem, fax, other tones etc. I think =
file:///Volumes/Main2/fluffy/src/drafts/jsep/draft-ietf-rtcweb-jsep.html#r=
fc.section.5.2.3.4 could be clearer about that. That has nothing to do =
with the=20

On the setting of the rfc6464 signaling for v bit, It seems to me the =
browser should always offer that and honor whatever is negotiated. I =
don't think the VoiceActivityDetection should effect this and the JS can =
munch the SDP to controls what happens if it wants.=20



> On Jan 7, 2016, at 8:37 AM, Bernard Aboba <bernard.aboba@gmail.com> =
wrote:
>=20
> JSEP Section 5.2.3.4 discusses the implications of the =
"VoiceActivityDetection" option for the offer.=20
>=20
> One thing that is not covered is whether the "VoiceActivityDetection" =
option determines whether the client-mixer "vad" extension is set to =
"on" or "off". =20
>=20
> For example, I believe that the W3C WEBRTC WG determined that the =
client-mixer extension (support for which is required in RTP-Usage) =
should always be advertised in the offer.  RFC 6464 Section 4 gives an =
example of the a=3Dextmap line:=20
>=20
> a=3Dextmap:6 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=3Don
>=20
> So, my question is whether "vad=3Don" or "vad=3Doff" is determined by =
the "VoiceActivityDetection" option, or if not, whether there is a =
setting (e.g. "vad=3Don") that is always present in the offer.=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Roman Shpount <roman@telurix.com>
To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] JSEP: Usage of a=rtcp attribute
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--001a113fcf0e573825052928d934
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On Tue, Jan 12, 2016 at 2:34 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> Having the attribute indicates that the device is capable of rtcp on a
> port other than the rtp+1 port
>

First of all, according to  RFC 5245 rtcp SDP attribute must be present
even if RTCP is being received on  rtp+1 port. Furthermore, RFC 5245
also specifies that if RTCP is not being used, b=RS:0 and b=RR:0 must be
present. If most of the cases (especially if TURN candidates are default,
which is the recommended behavior), default ICE candidate for RTCP is rtp+1
port, which makes it SDP rtcp attribute totally redundant.

More importantly, as far as initial offer for trickle ICE is concerned, I
do no think there are any use cases which would be negatively affected by
SDP rtcp attribute not being present in the initial offer with no
candidates. We can argue regarding RFC 5245 usage of rtcp attribute, or we
can get trickle ICE to update RFC 5245 (which it does anyway) and specify
that initial offer with no candidates MUST NOT include SDP rtcp attribute
unless rtcp-mux is required.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Jan 12, 2016 at 2:34 PM, Cullen Jennings <span dir=3D"ltr">&lt=
;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</=
span> wrote:<br></div></div><div class=3D"gmail_quote"><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><br=
>
Having the attribute indicates that the device is capable of rtcp on a port=
 other than the rtp+1 port<br></blockquote><div><br></div>First of all, acc=
ording to =C2=A0RFC 5245 rtcp SDP attribute must be present even if RTCP is=
 being received on=C2=A0<font color=3D"#000000"><span style=3D"font-size:12=
.8px">=C2=A0rtp+1 port.=C2=A0Furthermore, RFC 5245 also=C2=A0specifies=C2=
=A0that if RTCP is not being used,=C2=A0</span><span style=3D"font-size:13.=
3333px;white-space:pre-wrap">b=3DRS:0 and b=3DRR:0 must be present. If most=
 of the cases (especially if TURN candidates are default, which is the reco=
mmended behavior), default ICE candidate for RTCP is rtp+1 port, which make=
s it SDP rtcp attribute totally redundant.</span></font></div><div class=3D=
"gmail_quote"><font color=3D"#000000"><span style=3D"font-size:13.3333px;wh=
ite-space:pre-wrap"><br></span></font></div><div class=3D"gmail_quote"><fon=
t color=3D"#000000"><span style=3D"font-size:13.3333px;white-space:pre-wrap=
">More importantly, as far as initial offer for trickle ICE is concerned, I=
 do no think there are any use cases which would be negatively affected by =
SDP rtcp attribute not being present in the initial offer with no candidate=
s. We can argue regarding RFC 5245 usage of rtcp attribute, or we can get t=
rickle ICE to update RFC 5245 (which it does anyway) and specify that initi=
al offer with no candidates MUST NOT include SDP rtcp attribute unless rtcp=
-mux is required.<br></span></font><div><div class=3D"gmail_signature">____=
_________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--001a113fcf0e573825052928d934--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [rtcweb] JSEP: Usage of a=rtcp attribute
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From: Sean Turner <sean@sn3rd.com>
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Date: Wed, 20 Jan 2016 17:41:30 -0500
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Subject: [rtcweb] Call for WG adoption: draft-shieh-rtcweb-ip-handling
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All,

In Yokohama, we reviewed the "IP address leakage=E2=80=9D issues and the =
sense of the room was that we should document this in its own document.  =
So to get that done =E2=80=A6 please respond to indicate whether you =
support adoption of =
https://datatracker.ietf.org/doc/draft-shieh-rtcweb-ip-handling/ as a =
working group work item and whether you will participate in the =
discussion.   If you are opposed to this draft becoming a WG document, =
please say so (and say why).  Please have your response in by February =
4th.

Thanks in advance!
T/C/S=


From nobody Wed Jan 20 14:41:44 2016
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Subject: [rtcweb] The RTCWEB WG has placed draft-shieh-rtcweb-ip-handling in state "Call For Adoption By WG Issued"
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The RTCWEB WG has placed draft-shieh-rtcweb-ip-handling in state 
Call For Adoption By WG Issued (entered by spt)

The document is available at
https://datatracker.ietf.org/doc/draft-shieh-rtcweb-ip-handling/


From nobody Wed Jan 20 14:52:15 2016
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Subject: Re: [rtcweb] Call for WG adoption: draft-shieh-rtcweb-ip-handling
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I support adoption of this document as part of the "Send Security 
Solution (draft-ietf-rtcweb-security-arch) to IESG for publication as 
Proposed Standard" milestone, although would suggest revising the 
milestone name.

I plan to participate in the discussion, and hope to have feedback on 
the current version to the mailing list by the end of next week.

/a


On 1/20/16 16:41, Sean Turner wrote:
> All,
>
> In Yokohama, we reviewed the "IP address leakage” issues and the sense of the room was that we should document this in its own document.  So to get that done … please respond to indicate whether you support adoption of https://datatracker.ietf.org/doc/draft-shieh-rtcweb-ip-handling/ as a working group work item and whether you will participate in the discussion.   If you are opposed to this draft becoming a WG document, please say so (and say why).  Please have your response in by February 4th.
>
> Thanks in advance!
> T/C/S
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Jan 21 08:39:43 2016
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: (xrblock) WGLC for draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02.txt
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--_000_9904FB1B0159DA42B0B887B7FA8119CA6BEF3155AZFFEXMB04globa_
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The xrblock WG sent to LC the Internet-Draft 'Considerations for Selecting =
RTCP Extended Report (XR) Metrics for the WebRTC Statistics API'.



Comments or questions are welcome - best on xrblock@ietf.org<mailto:xrblock=
@ietf.org>.



Thanks and Regards,



Dan




From: xrblock [mailto:xrblock-bounces@ietf.org] On Behalf Of Shida Schubert
Sent: Wednesday, January 20, 2016 6:32 AM
To: xrblock
Subject: [xrblock] WGLC for draft-ietf-xrblock-rtcweb-rtcp-xr-metrics

All;

This message starts a Working Group Last Call for the draft-ietf-xrblock-rt=
cweb-rtcp-xr-metrics-02.txt.

http://tools.ietf.org/id/draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02.txt<h=
ttps://urldefense.proofpoint.com/v2/url?u=3Dhttp-3A__tools.ietf.org_id_draf=
t-2Dietf-2Dxrblock-2Drtcweb-2Drtcp-2Dxr-2Dmetrics-2D02.txt&d=3DBQMFaQ&c=3DB=
FpWQw8bsuKpl1SgiZH64Q&r=3DI4dzGxR31OcNXCJfQzvlsiLQfucBXRucPvdrphpBsFA&m=3Dr=
oYRd0Cw-W2RAdJIn7eAn7AQ4Z7SrXV2hPXGIaXFmj0&s=3DFlCmsxln6mXHYqWYtqvo3iJXiWQN=
ClRJGMew6yqxVpY&e=3D>

Even if you have no questions, comments or concern, if you have read the dr=
aft and agree that it's ready for submission to IESG as a Standard Track, p=
lease send a message to the list indicating this.

Obviously if you have any issues or questions please submit it to the list,=
 if you are highlighting issues suggestions to fix the issues is always hel=
pful.

The Working Group Last Call will end on February 4th.

Shida
As co-chair


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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;">The xrblock WG sent to LC the Internet-Draft &#8216;Co=
nsiderations for Selecting RTCP Extended Report (XR) Metrics for the WebRTC=
 Statistics API&#8217;. <o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;">Comments or questions are welcome &#8211; best on <a h=
ref=3D"mailto:xrblock@ietf.org">xrblock@ietf.org</a>. <o:p></o:p></span></p=
re>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;">Thanks and Regards,<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;">Dan<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Times New Roman&quot=
;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></pre>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt"><o:p>&nbsp;</o:p></=
span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> xrblock =
[<a href=3D"mailto:xrblock-bounces@ietf.org">mailto:xrblock-bounces@ietf.or=
g</a>]
<b>On Behalf Of </b>Shida Schubert<br>
<b>Sent:</b> Wednesday, January 20, 2016 6:32 AM<br>
<b>To:</b> xrblock<br>
<b>Subject:</b> [xrblock] WGLC for draft-ietf-xrblock-rtcweb-rtcp-xr-metric=
s<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">All;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This message starts a Working Group Last Call for th=
e&nbsp;draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02.txt.&nbsp;<o:p></o:p></=
p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><a href=3D"https://urldefense.proofpoint.com/v2/url?=
u=3Dhttp-3A__tools.ietf.org_id_draft-2Dietf-2Dxrblock-2Drtcweb-2Drtcp-2Dxr-=
2Dmetrics-2D02.txt&amp;d=3DBQMFaQ&amp;c=3DBFpWQw8bsuKpl1SgiZH64Q&amp;r=3DI4=
dzGxR31OcNXCJfQzvlsiLQfucBXRucPvdrphpBsFA&amp;m=3DroYRd0Cw-W2RAdJIn7eAn7AQ4=
Z7SrXV2hPXGIaXFmj0&amp;s=3DFlCmsxln6mXHYqWYtqvo3iJXiWQNClRJGMew6yqxVpY&amp;=
e=3D">http://tools.ietf.org/id/draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02=
.txt</a><o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Even if you have no questions, comments or concern, =
if you have read the draft and agree that it&#8217;s ready for submission t=
o IESG as a Standard Track, please send a message to the list indicating th=
is.&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Obviously if you have any issues or questions please=
 submit it to the list, if you are highlighting issues suggestions to fix t=
he issues is always helpful.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The Working Group Last Call will end on February 4th=
.&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Shida<o:p></o:p></p>
<p class=3D"MsoNormal">As co-chair<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_9904FB1B0159DA42B0B887B7FA8119CA6BEF3155AZFFEXMB04globa_--


From nobody Thu Jan 21 12:07:57 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Overview: Real Time Protocols for Browser-based Applications
        Author          : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-15.txt
	Pages           : 22
	Date            : 2016-01-21

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This document is an Applicability Statement - it does not itself
   specify any protocol, but specifies which other specifications WebRTC
   compliant implementations are supposed to follow.

   This document is a work item of the RTCWEB working group.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-overview-15

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-15


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Thu Jan 21 12:24:36 2016
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To: rtcweb@ietf.org
References: <EE273D0B-73DE-43F9-B06C-73CE1D856C55@sn3rd.com>
From: Harald Alvestrand <harald@alvestrand.no>
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I support adoption of this draft as the a working group item.

Den 20. jan. 2016 23:41, skrev Sean Turner:
> All,
> 
> In Yokohama, we reviewed the "IP address leakage” issues and the sense of the room was that we should document this in its own document.  So to get that done … please respond to indicate whether you support adoption of https://datatracker.ietf.org/doc/draft-shieh-rtcweb-ip-handling/ as a working group work item and whether you will participate in the discussion.   If you are opposed to this draft becoming a WG document, please say so (and say why).  Please have your response in by February 4th.
> 
> Thanks in advance!
> T/C/S
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


From nobody Thu Jan 21 12:25:55 2016
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From: Harald Alvestrand <harald@alvestrand.no>
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Date: Thu, 21 Jan 2016 21:25:47 +0100
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This is a keepalive update only. No issues have been filed on github or
mentioned where they stuck in my memory, so I have merely changed the
dates (and one reference).

Harald

Den 21. jan. 2016 21:07, skrev internet-drafts@ietf.org:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
> 
>         Title           : Overview: Real Time Protocols for Browser-based Applications
>         Author          : Harald T. Alvestrand
> 	Filename        : draft-ietf-rtcweb-overview-15.txt
> 	Pages           : 22
> 	Date            : 2016-01-21
> 
> Abstract:
>    This document gives an overview and context of a protocol suite
>    intended for use with real-time applications that can be deployed in
>    browsers - "real time communication on the Web".
> 
>    It intends to serve as a starting and coordination point to make sure
>    all the parts that are needed to achieve this goal are findable, and
>    that the parts that belong in the Internet protocol suite are fully
>    specified and on the right publication track.
> 
>    This document is an Applicability Statement - it does not itself
>    specify any protocol, but specifies which other specifications WebRTC
>    compliant implementations are supposed to follow.
> 
>    This document is a work item of the RTCWEB working group.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/
> 
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-overview-15
> 
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-15
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


From nobody Thu Jan 21 12:45:32 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Gateways
        Authors         : Harald Alvestrand
                          Uwe Rauschenbach
	Filename        : draft-ietf-rtcweb-gateways-02.txt
	Pages           : 7
	Date            : 2016-01-21

Abstract:
   This document describes interoperability considerations for a class
   of WebRTC-compatible endpoints called "WebRTC gateways", which
   interconnect between WebRTC endpoints and devices that are not WebRTC
   endpoints.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-gateways/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-gateways-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-gateways-02


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Date: Thu, 21 Jan 2016 21:47:25 +0100
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-gateways-02.txt
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This, too, is a keepalive.
Please file issues in github for desired changes.

Den 21. jan. 2016 21:45, skrev internet-drafts@ietf.org:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
> 
>         Title           : WebRTC Gateways
>         Authors         : Harald Alvestrand
>                           Uwe Rauschenbach
> 	Filename        : draft-ietf-rtcweb-gateways-02.txt
> 	Pages           : 7
> 	Date            : 2016-01-21
> 
> Abstract:
>    This document describes interoperability considerations for a class
>    of WebRTC-compatible endpoints called "WebRTC gateways", which
>    interconnect between WebRTC endpoints and devices that are not WebRTC
>    endpoints.
> 
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-gateways/
> 
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-gateways-02
> 
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-gateways-02
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


From nobody Thu Jan 21 18:49:06 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC) 
        Author          : Martin Thomson
	Filename        : draft-ietf-rtcweb-alpn-02.txt
	Pages           : 7
	Date            : 2016-01-21

Abstract:
   Application Layer Protocol Negotiation (ALPN) labels are defined for
   use in identifying Web Real-Time Communications (WebRTC) usages of
   Datagram Transport Layer Security (DTLS).  Labels are provided for
   identifying a session that uses a combination of WebRTC compatible
   media and data, and for identifying a session requiring
   confidentiality protection.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-alpn-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-alpn-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Mon Jan 25 07:56:35 2016
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-alpn-02.txt
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Just so that nobody is caught off-guard: The chairs asked Martin to =
submit this version so we could get a WGLC going on it.  Expect that =
call to come out shortly in a separate message.

spt

> On Jan 21, 2016, at 21:49, internet-drafts@ietf.org wrote:
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>        Title           : Application Layer Protocol Negotiation for =
Web Real-Time Communications (WebRTC)=20
>        Author          : Martin Thomson
> 	Filename        : draft-ietf-rtcweb-alpn-02.txt
> 	Pages           : 7
> 	Date            : 2016-01-21
>=20
> Abstract:
>   Application Layer Protocol Negotiation (ALPN) labels are defined for
>   use in identifying Web Real-Time Communications (WebRTC) usages of
>   Datagram Transport Layer Security (DTLS).  Labels are provided for
>   identifying a session that uses a combination of WebRTC compatible
>   media and data, and for identifying a session requiring
>   confidentiality protection.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-alpn-02
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-alpn-02
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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All,

This is the working group last call for the =E2=80=9CALPN" draft =
available at https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/.  =
Please review the draft and send your comments to this list by Tuesday, =
February 9th, 2016. =20

Thanks,
C/T/S=


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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] Working group last call for draft-ietf-rtcweb-audio
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--Apple-Mail=_F879A6FF-BD73-4469-9B56-2AC573FA1EA9
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> On 19 Dec 2015, at 00:10, Cullen Jennings (fluffy) <fluffy@cisco.com =
<mailto:fluffy@cisco.com>> wrote:
>=20
>=20
>> On Dec 18, 2015, at 3:24 PM, Roman Shpount <roman@telurix.com =
<mailto:roman@telurix.com>> wrote:
>>=20
>> This applies to all DTMF tones, not A-D specifically. What I am =
saying is that WebRTC endpoints must be able to generate all DTMF tones =
including A-D and send them to legacy PSTN (gateway).
>=20
> The thing people keep asking is why? what is the use case ?
>=20
> I don=E2=80=99t feel like I have seen an answer to this.=20

More specifically - Under what circumstances will we need them where=20
the same result couldn=E2=80=99t be achieved by injecting the correct =
notes into the stream using webAudio?
It has been suggested that webaudio tone generation might not work with =
certain sets of codecs.
But surely the (few) legacy systems (coal mines for example) that use =
A-D will be using G711
for which tone generation should work well.

(I might add that tone _detection_ can also be done in webaudio so =
developers have a symmetrical solution).

Tim.

>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




--Apple-Mail=_F879A6FF-BD73-4469-9B56-2AC573FA1EA9
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 19 Dec 2015, at 00:10, Cullen Jennings =
(fluffy) &lt;<a href=3D"mailto:fluffy@cisco.com" =
class=3D"">fluffy@cisco.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D"">

<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dutf-8" =
class=3D"">

<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space;" class=3D"">
<br class=3D"">
<div class=3D"">
<blockquote type=3D"cite" class=3D"">
<div class=3D"">On Dec 18, 2015, at 3:24 PM, Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com" class=3D"">roman@telurix.com</a>&gt; =
wrote:</div>
<br class=3D"Apple-interchange-newline">
<div class=3D""><span style=3D"font-family: Helvetica; font-size: 14px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; float: =
none; display: inline !important;" class=3D"">This
 applies to all DTMF tones, not A-D specifically. What I am saying is =
that WebRTC endpoints must be able to generate all DTMF tones including =
A-D and send them to legacy PSTN (gateway).</span></div>
</blockquote>
</div>
<br class=3D"">
<div class=3D"">The thing people keep asking is why? what is the use =
case ?</div>
<div class=3D""><br class=3D"">
</div>
<div class=3D"">I don=E2=80=99t feel like I have seen an answer to =
this.&nbsp;</div></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">More specifically - Under what =
circumstances will we need them where&nbsp;</div><div class=3D"">the =
same result couldn=E2=80=99t be achieved by injecting the correct notes =
into the stream using webAudio?</div><div class=3D"">It has been =
suggested that webaudio tone generation might not work with certain sets =
of codecs.</div><div class=3D"">But surely the (few) legacy systems =
(coal mines for example) that use A-D will be using G711</div><div =
class=3D"">for which tone generation should work well.</div><div =
class=3D""><br class=3D""></div><div class=3D"">(I might add that tone =
_detection_ can also be done in webaudio so developers have a =
symmetrical solution).</div><div class=3D""><br class=3D""></div><div =
class=3D"">Tim.</div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><div style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D"">
<div class=3D""><br class=3D"">
</div>
<div class=3D""><br class=3D"">
</div>
</div>

_______________________________________________<br class=3D"">rtcweb =
mailing list<br class=3D""><a href=3D"mailto:rtcweb@ietf.org" =
class=3D"">rtcweb@ietf.org</a><br class=3D""><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a><br =
class=3D""></div></blockquote></div><br class=3D""><div =
apple-content-edited=3D"true" class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: 2; text-align: =
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: 0; "><div class=3D"">Tim Panton - =
Web/VoIP consultant and implementor</div><div class=3D""><a =
href=3D"http://www.westhawk.co.uk" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></body></html>=

--Apple-Mail=_F879A6FF-BD73-4469-9B56-2AC573FA1EA9--


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Date: Tue, 26 Jan 2016 15:26:28 +0100
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Archived-At: <http://mailarchive.ietf.org/arch/msg/rtcweb/w8mfgj2h4KqH7Hk3WegW9JAZArU>
Subject: Re: [rtcweb] Call for WG adoption: draft-shieh-rtcweb-ip-handling
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I support the adoption of this draft as WG item

On Wed, Jan 20, 2016 at 11:41 PM, Sean Turner <sean@sn3rd.com> wrote:
> All,
>
> In Yokohama, we reviewed the "IP address leakage=E2=80=9D issues and the =
sense of the room was that we should document this in its own document.  So=
 to get that done =E2=80=A6 please respond to indicate whether you support =
adoption of https://datatracker.ietf.org/doc/draft-shieh-rtcweb-ip-handling=
/ as a working group work item and whether you will participate in the disc=
ussion.   If you are opposed to this draft becoming a WG document, please s=
ay so (and say why).  Please have your response in by February 4th.
>
> Thanks in advance!
> T/C/S
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Victor Pascual =C3=81vila


From nobody Tue Jan 26 11:51:32 2016
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-return-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Recursively Encapsulated TURN (RETURN) for Connectivity and Privacy in WebRTC
        Authors         : Benjamin M. Schwartz
                          Justin Uberti
	Filename        : draft-ietf-rtcweb-return-01.txt
	Pages           : 18
	Date            : 2016-01-26

Abstract:
   In the context of WebRTC, the concept of a local TURN proxy has been
   suggested, but not reviewed in detail.  WebRTC applications are
   already using TURN to enhance connectivity and privacy.  This
   document explains how local TURN proxies and WebRTC applications can
   work together.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-return-01

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-return-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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From: Roman Shpount <roman@telurix.com>
To: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Working group last call for draft-ietf-rtcweb-audio
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--089e011602a8467b10052a42218e
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On Tue, Jan 26, 2016 at 7:15 AM, Tim Panton <tim@phonefromhere.com> wrote:

>
> More specifically - Under what circumstances will we need them where
> the same result couldn=E2=80=99t be achieved by injecting the correct not=
es into
> the stream using webAudio?
> It has been suggested that webaudio tone generation might not work with
> certain sets of codecs.
> But surely the (few) legacy systems (coal mines for example) that use A-D
> will be using G711
> for which tone generation should work well.
>
> (I might add that tone _detection_ can also be done in webaudio so
> developers have a symmetrical solution).
>

Any tone generated using inband audio will be less reliable then
telephone-events tone. For telephone-events 3 or more packets in a row need
to be lost for the tone to be lost or interrupted. For inband loosing a
single packet can break tone in two distinct tones. I am not even talking
about all the extra work and interop required to properly implement tone
generation. To generate all DTMF tone telephone-events you need no extra
code in the browser you need no additional code.

Finally, if you want WebRTC to be backwards compatible with RFC 2833 (which
is the most common implementation of telephone-events in PSTN universe),
then you will need to deal with this
https://tools.ietf.org/html/rfc2833#section-3.3:

   A compliant implementation MUST support the events listed in Table 1
   with the exception of "flash". If it uses some other, out-of-band
   mechanism for signaling line conditions, it does not have to
   implement the other events.


Here is table 1 (https://tools.ietf.org/html/rfc2833#section-3.10):

   Table 1 summarizes the DTMF-related named events within the
   telephone-event payload format.

                     Event  encoding (decimal)
                     _________________________
                     0--9                0--9
                     *                     10
                     #                     11
                     A--D              12--15
                     Flash                 16

                     Table 1: DTMF named events


So, all DTMF digits 0-9, *, #, A-D are required by RFC 2833.

So, if you want to break backwards compatibility please explain your
motivation. These tones (as well as all the other telephone-events) are
only needed for backwards compatibility. Implemented all or some of them
requires the same amount of effort. Not implementing all of them will make
life harder for the few people who use them.

Finally, I have wrote several other things regarding DTMF tone generation
which are considerably more important. Does everyone agree with the
following or no one cares:

Generated events MUST have duration of no more than 6000 ms and no less
than 40 ms with the recommended default duration of 100 ms for each tone.
The gap between events MUST be no less then 30 ms with the recommended
default duration of 70 ms. Event SHOULD start on a regular audio packet
border and event and gap duration SHOULD be rounded up to the next regular
audio packet border.

During the time events are generated no audio SHOULD be sent for the same
audio stream. When gaps between events are generated, silence
and not the background audio SHOULD be sent using regular audio encoding.

If multiple audio sampling rates are supported, audio/telephone-event
payload
SHOULD be present for each supported sampling rate. Endpoints SHOULD use
audio/telephone-event format parameters during the offer/answer to
indicate which events are supported.

Receivers MUST be able to consume any audio/telephone-event events
in such a way that it will not generate audio artifacts, jitter buffer
adjustments, payload mismatches, or invalid RTCP statistics calculation.
Receivers MAY generate audio corresponding to the received events
but are also allowed to discard them in a manner that does not affect
regular audio processing.

Regards,
_____________
Roman Shpount

--089e011602a8467b10052a42218e
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Jan 26, 2016 at 7:15 AM, Tim Panton <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:tim@phonefromhere.com" target=3D"_blank">tim@phonefromhere.co=
m</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-widt=
h:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-le=
ft:1ex"><div style=3D"word-wrap:break-word"><div><div><div class=3D"h5"><di=
v><br></div></div></div><div>More specifically - Under what circumstances w=
ill we need them where=C2=A0</div><div>the same result couldn=E2=80=99t be =
achieved by injecting the correct notes into the stream using webAudio?</di=
v><div>It has been suggested that webaudio tone generation might not work w=
ith certain sets of codecs.</div><div>But surely the (few) legacy systems (=
coal mines for example) that use A-D will be using G711</div><div>for which=
 tone generation should work well.</div><div><br></div><div>(I might add th=
at tone _detection_ can also be done in webaudio so developers have a symme=
trical solution).</div></div></div></blockquote><div><br></div><div>Any ton=
e generated using inband audio will be less reliable then telephone-events =
tone. For telephone-events 3 or more packets in a row need to be lost for t=
he tone to be lost or interrupted. For inband loosing a single packet can b=
reak tone in two distinct tones. I am not even talking about all the extra =
work and interop required to properly implement tone generation. To generat=
e all DTMF tone telephone-events you need no extra code in the browser you =
need no additional code.</div><div><br></div><div>Finally, if you want WebR=
TC to be backwards compatible with RFC 2833 (which is the most common imple=
mentation of telephone-events in PSTN universe), then you will need to deal=
 with this=C2=A0<a href=3D"https://tools.ietf.org/html/rfc2833#section-3.3"=
>https://tools.ietf.org/html/rfc2833#section-3.3</a>:</div><div><br></div><=
div><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bott=
om:0px;color:rgb(0,0,0)">   A compliant implementation MUST support the eve=
nts listed in Table 1
   with the exception of &quot;flash&quot;. If it uses some other, out-of-b=
and
   mechanism for signaling line conditions, it does not have to
   implement the other events.</pre><pre class=3D"" style=3D"font-size:13.3=
333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre>Here is =
table 1 (<a href=3D"https://tools.ietf.org/html/rfc2833#section-3.10">https=
://tools.ietf.org/html/rfc2833#section-3.10</a>):</div><div><br></div><div>=
<pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0=
px;color:rgb(0,0,0)">   Table 1 summarizes the DTMF-related named events wi=
thin the
   telephone-event payload format.

                     Event  encoding (decimal)
                     _________________________
                     0--9                0--9
                     *                     10
                     #                     11
                     A--D              12--15
                     Flash                 16

                     Table 1: DTMF named events</pre><pre class=3D"" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">=
<br></pre>So, all DTMF digits 0-9, *, #, A-D are required by RFC 2833.</div=
><div><br></div><div>So, if you want to break backwards compatibility pleas=
e explain your motivation. These tones (as well as all the other telephone-=
events) are only needed for backwards compatibility. Implemented all or som=
e of them requires the same amount of effort. Not implementing all of them =
will make life harder for the few people who use them.</div><div><br></div>=
<div>Finally, I have wrote several other things regarding DTMF tone generat=
ion which are considerably more important. Does everyone agree with the fol=
lowing or no one cares:</div><div><br></div><div><div style=3D"color:rgb(0,=
0,0);font-size:12.8px"><font face=3D"monospace, monospace">Generated events=
 MUST have duration of no more than 6000 ms and no less</font></div><div st=
yle=3D"color:rgb(0,0,0);font-size:12.8px"><font face=3D"monospace, monospac=
e">than 40 ms with the recommended default duration of 100 ms for each tone=
.=C2=A0</font></div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><font =
face=3D"monospace, monospace">The gap between events MUST be no less then 3=
0 ms with the recommended</font></div><div style=3D"color:rgb(0,0,0);font-s=
ize:12.8px"><font face=3D"monospace, monospace">default duration of 70 ms. =
Event SHOULD start on a regular audio packet</font></div><div style=3D"colo=
r:rgb(0,0,0);font-size:12.8px"><font face=3D"monospace, monospace">border a=
nd event and gap duration SHOULD be rounded up to the next regular</font></=
div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><font face=3D"monospac=
e, monospace">audio=C2=A0</font><span style=3D"font-family:monospace,monosp=
ace">packet border.</span></div><div style=3D"color:rgb(0,0,0);font-size:12=
.8px"><font face=3D"monospace, monospace"><br></font></div><div style=3D"co=
lor:rgb(0,0,0);font-size:12.8px"><font face=3D"monospace, monospace">During=
 the time events are generated no audio SHOULD be sent for the same</font><=
/div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><font face=3D"monospa=
ce, monospace">audio stream. When gaps between events are generated, silenc=
e</font></div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><span style=
=3D"font-family:monospace,monospace">and not the background audio SHOULD be=
 sent using regular audio encoding.</span><br></div><div style=3D"color:rgb=
(0,0,0);font-size:12.8px"><font face=3D"monospace, monospace"><br></font></=
div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><font face=3D"monospac=
e, monospace">If multiple audio sampling rates are supported, audio/telepho=
ne-event payload</font></div><div style=3D"color:rgb(0,0,0);font-size:12.8p=
x"><font face=3D"monospace, monospace">SHOULD be present for each supported=
 sampling rate. Endpoints SHOULD use</font></div><div style=3D"color:rgb(0,=
0,0);font-size:12.8px"><font face=3D"monospace, monospace">audio/telephone-=
event format parameters during the offer/answer to=C2=A0</font></div><div s=
tyle=3D"color:rgb(0,0,0);font-size:12.8px"><font face=3D"monospace, monospa=
ce">indicate which events are supported.</font></div><div style=3D"color:rg=
b(0,0,0);font-size:12.8px"><font face=3D"monospace, monospace"><br></font><=
/div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><font face=3D"monospa=
ce, monospace">Receivers MUST be able to consume any audio/telephone-event =
events</font></div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><font f=
ace=3D"monospace, monospace">in such a way that it will not generate audio =
artifacts, jitter buffer</font></div><div style=3D"color:rgb(0,0,0);font-si=
ze:12.8px"><font face=3D"monospace, monospace">adjustments, payload mismatc=
hes, or invalid RTCP statistics calculation.</font></div><div style=3D"colo=
r:rgb(0,0,0);font-size:12.8px"><font face=3D"monospace, monospace">Receiver=
s MAY generate audio=C2=A0</font><span style=3D"font-family:monospace,monos=
pace">corresponding to the received events=C2=A0</span></div><div style=3D"=
color:rgb(0,0,0);font-size:12.8px"><span style=3D"font-family:monospace,mon=
ospace">but are also allowed to discard them=C2=A0</span><span style=3D"fon=
t-family:monospace,monospace">in a manner that does not affect=C2=A0</span>=
</div><div style=3D"color:rgb(0,0,0);font-size:12.8px"><span style=3D"font-=
family:monospace,monospace">regular audio processing.</span></div></div><di=
v style=3D"color:rgb(0,0,0);font-size:12.8px"><span style=3D"font-family:mo=
nospace,monospace"><br></span></div><div style=3D"color:rgb(0,0,0);font-siz=
e:12.8px"><span style=3D"font-family:monospace,monospace">Regards,</span></=
div><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div=
></div><div>=C2=A0</div></div></div></div>

--089e011602a8467b10052a42218e--


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To: Roman Shpount <roman@telurix.com>, Tim Panton <tim@phonefromhere.com>
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

Hi Roman,

On 01/26/2016 02:59 PM, Roman Shpount wrote:
> Finally, I have wrote several other things regarding DTMF tone 
> generation which are considerably more important. Does everyone
> agree with the following or no one cares:

The text below is what you propose for the rtcweb-audio draft? I
personally have no opinion on these events (see comment inline for
clarification though) so I'm happy to include that text if everyone
else is OK with it.

> Generated events MUST have duration of no more than 6000 ms and no
> less than 40 ms with the recommended default duration of 100 ms for
> each tone. The gap between events MUST be no less then 30 ms with
> the recommended default duration of 70 ms. Event SHOULD start on a
> regular audio packet border and event and gap duration SHOULD be
> rounded up to the next regular audio packet border.
> 
> During the time events are generated no audio SHOULD be sent for
> the same audio stream. When gaps between events are generated,
> silence and not the background audio SHOULD be sent using regular
> audio encoding.

I think this would require a bit of clarification. What does "no
audio" mean here? Silence or no codec packets at all (in that case
what about PLC)? Also, keep in mind that any silence may not be
perfectly aligned with theoretical frame boundaries, due to look-ahead.

> If multiple audio sampling rates are supported,
> audio/telephone-event payload SHOULD be present for each supported
> sampling rate. Endpoints SHOULD use audio/telephone-event format
> parameters during the offer/answer to indicate which events are
> supported.
> 
> Receivers MUST be able to consume any audio/telephone-event events 
> in such a way that it will not generate audio artifacts, jitter
> buffer adjustments, payload mismatches, or invalid RTCP statistics
> calculation. Receivers MAY generate audio corresponding to the
> received events but are also allowed to discard them in a manner
> that does not affect regular audio processing.

Cheers,

	Jean-Marc
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From: Roman Shpount <roman@telurix.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>
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Subject: Re: [rtcweb] Working group last call for draft-ietf-rtcweb-audio
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On Tue, Jan 26, 2016 at 3:26 PM, Jean-Marc Valin <jmvalin@mozilla.com>
wrote:

> The text below is what you propose for the rtcweb-audio draft? I
> personally have no opinion on these events (see comment inline for
> clarification though) so I'm happy to include that text if everyone
> else is OK with it.
>

Yes, this is the text I am proposing for rtcweb-audio draft.

> Generated events MUST have duration of no more than 6000 ms and no
> > less than 40 ms with the recommended default duration of 100 ms for
> > each tone. The gap between events MUST be no less then 30 ms with
> > the recommended default duration of 70 ms. Event SHOULD start on a
> > regular audio packet border and event and gap duration SHOULD be
> > rounded up to the next regular audio packet border.
> >
> > During the time events are generated no audio SHOULD be sent for
> > the same audio stream. When gaps between events are generated,
> > silence and not the background audio SHOULD be sent using regular
> > audio encoding.
>
> I think this would require a bit of clarification. What does "no
> audio" mean here? Silence or no codec packets at all (in that case
> what about PLC)? Also, keep in mind that any silence may not be
> perfectly aligned with theoretical frame boundaries, due to look-ahead.
>
>
First of all I was trying to copy all the normative tone duration
information from W3C document to rtcweb-audio draft.

Second, as far as "no audio" is concerned, what I mean here is that RFC4733
telephone-events are alternative audio encoding similar to CN (see
https://tools.ietf.org/html/rfc4733#section-2.5.2.2). For RTP timestamp
interval covered by telephone-event no non-event audio packets, such as
Opus or G.711, should be present since they are redundant and cause interop
problems. PLC does not apply here since remote gateways is supposed to
generate audio based on the information in telephone-events packets. This
is essentially the same requirement that end point should not send packets
with two different audio encodings (i.e. send Opus and G.711 RTP packets)
at the same time. This can work but can also cause interop problems.

Packet boundaries of audio encoded using codecs such as G.711 or Opus
should create a continuous stream as far as RTP timestamps are concerned
with RFC 4733 telephone event packets unless discontinuous RTP is used. It
should be possible to align decoded audio and telephone-events based on RTP
timestamps.

I also wanted to provide clarification for
https://tools.ietf.org/html/rfc4733#section-2.5.1.2. This section says:

   A source has wide latitude as to how often it sends event updates.  A
   natural interval is the spacing between non-event audio packets.
   (Recall that a single RTP packet can contain multiple audio frames
   for frame-based codecs and that the packet interval can vary during a
   session.)  Alternatively, a source MAY decide to use a different
   spacing for event updates, with a value of 50 ms RECOMMENDED.


I am proposing that WebRTC should recommend that sending telephone-event
packet at natural interval between non-event audio packets is
strongly preferred.

Also, blocking input audio for a short period of time after
telephone-events are transmitted prevents a duplicate DTMF digit audio from
being present in the encoded audio stream which some times happen due to
microphone picking up audio feedback tones played by some DTMF keypad
implementations. This also improves DTMF tone recognition by a receiving
IVR system after audio stream passes through PSTN gateway since having
background audio as a separator between DTMF digits can negatively affect
some tone detection algorithms.

Please let me know if you need more information.

Regards,
_____________
Roman Shpount

--047d7b10ca098b963b052a434572
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Jan 26, 2016 at 3:26 PM, Jean-Marc Valin <span dir=3D"ltr">&lt=
;<a href=3D"mailto:jmvalin@mozilla.com" target=3D"_blank">jmvalin@mozilla.c=
om</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wid=
th:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-l=
eft:1ex">The text below is what you propose for the rtcweb-audio draft? I<b=
r>
personally have no opinion on these events (see comment inline for<br>
clarification though) so I&#39;m happy to include that text if everyone<br>
else is OK with it.<br></blockquote><div><br></div><div>Yes, this is the te=
xt I am proposing for rtcweb-audio draft.=C2=A0</div><div><br></div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex"><span class=3D"">&gt; Generated events MUST have duration of no m=
ore than 6000 ms and no<br>
&gt; less than 40 ms with the recommended default duration of 100 ms for<br=
>
&gt; each tone. The gap between events MUST be no less then 30 ms with<br>
&gt; the recommended default duration of 70 ms. Event SHOULD start on a<br>
&gt; regular audio packet border and event and gap duration SHOULD be<br>
&gt; rounded up to the next regular audio packet border.<br>
&gt;<br>
&gt; During the time events are generated no audio SHOULD be sent for<br>
&gt; the same audio stream. When gaps between events are generated,<br>
&gt; silence and not the background audio SHOULD be sent using regular<br>
&gt; audio encoding.<br>
<br>
</span>I think this would require a bit of clarification. What does &quot;n=
o<br>
audio&quot; mean here? Silence or no codec packets at all (in that case<br>
what about PLC)? Also, keep in mind that any silence may not be<br>
perfectly aligned with theoretical frame boundaries, due to look-ahead.<br>
<span class=3D""><br></span></blockquote><div><br></div><div>First of all I=
 was trying to copy all the normative tone duration information from W3C do=
cument to rtcweb-audio draft.</div><div><br></div><div>Second, as far as &q=
uot;no audio&quot; is concerned, what I mean here is that RFC4733 telephone=
-events are alternative audio encoding similar to CN (see =C2=A0<a href=3D"=
https://tools.ietf.org/html/rfc4733#section-2.5.2.2">https://tools.ietf.org=
/html/rfc4733#section-2.5.2.2</a>). For RTP timestamp interval covered by t=
elephone-event no non-event audio packets, such as Opus or G.711, should be=
 present since they are redundant and cause interop problems. PLC does not =
apply here since remote gateways is supposed to generate audio based on the=
 information in telephone-events packets. This is essentially the same requ=
irement that end point should not send packets with two different audio enc=
odings (i.e. send Opus and G.711 RTP packets) at the same time. This can wo=
rk but can also cause interop problems.=C2=A0</div><div><br></div><div>Pack=
et boundaries of audio encoded using codecs such as G.711 or Opus should cr=
eate a continuous stream as far as RTP timestamps are concerned with RFC 47=
33 telephone event packets unless discontinuous RTP is used. It should be p=
ossible to align decoded audio and telephone-events based on RTP timestamps=
.</div><div><br></div><div>I also wanted to provide clarification for <a hr=
ef=3D"https://tools.ietf.org/html/rfc4733#section-2.5.1.2">https://tools.ie=
tf.org/html/rfc4733#section-2.5.1.2</a>. This section says:</div><div><br><=
/div><div><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margi=
n-bottom:0px;color:rgb(0,0,0)">   A source has wide latitude as to how ofte=
n it sends event updates.  A
   natural interval is the spacing between non-event audio packets.
   (Recall that a single RTP packet can contain multiple audio frames
   for frame-based codecs and that the packet interval can vary during a
   session.)  Alternatively, a source MAY decide to use a different
   spacing for event updates, with a value of 50 ms RECOMMENDED.
</pre></div><div><br></div><div>I am proposing that WebRTC should recommend=
 that sending telephone-event packet at natural interval between n<font col=
or=3D"#000000"><span style=3D"font-size:13.3333px">on-event audio packets i=
s strongly=C2=A0preferred.</span></font></div><div><span style=3D"color:rgb=
(0,0,0);font-size:13.3333px"><br></span></div><div><span style=3D"color:rgb=
(0,0,0);font-size:13.3333px">Also, blocking input audio for a short period =
of time after telephone-events are transmitted prevents a duplicate DTMF di=
git audio from being present in the encoded audio stream which some times h=
appen due to microphone picking up audio feedback tones played by some DTMF=
 keypad implementations. This also improves DTMF tone recognition by a rece=
iving IVR system after audio stream passes through PSTN gateway since havin=
g background audio as a separator between DTMF digits can negatively affect=
 some tone detection algorithms.</span></div><div><span style=3D"color:rgb(=
0,0,0);font-size:13.3333px"><br></span></div><div><span style=3D"color:rgb(=
0,0,0);font-size:13.3333px">Please let me know if you need more information=
.</span></div><div><span style=3D"color:rgb(0,0,0);font-size:13.3333px"><br=
></span></div><div><span style=3D"color:rgb(0,0,0);font-size:13.3333px">Reg=
ards,</span></div><div><div class=3D"gmail_signature">_____________<br>Roma=
n Shpount</div></div><div>=C2=A0</div></div></div></div>

--047d7b10ca098b963b052a434572--


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To: Roman Shpount <roman@telurix.com>
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From: Jean-Marc Valin <jmvalin@mozilla.com>
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Subject: Re: [rtcweb] Working group last call for draft-ietf-rtcweb-audio
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Hash: SHA256

On 01/26/2016 04:21 PM, Roman Shpount wrote:
> Second, as far as "no audio" is concerned, what I mean here is
> that RFC4733 telephone-events are alternative audio encoding
> similar to CN (see
> https://tools.ietf.org/html/rfc4733#section-2.5.2.2). For RTP 
> timestamp interval covered by telephone-event no non-event audio 
> packets, such as Opus or G.711, should be present since they are 
> redundant and cause interop problems. PLC does not apply here
> since remote gateways is supposed to generate audio based on the
> information in telephone-events packets. This is essentially the
> same requirement that end point should not send packets with two
> different audio encodings (i.e. send Opus and G.711 RTP packets) at
> the same time. This can work but can also cause interop problems.

The problem I see is that unlike G.711, modern codecs (not just Opus)
aren't stateless. It's not a simple thing to just stop and restart
transmission during telephony events -- especially if you want to
avoid glitches and want all implementations to behave the same. You
need to consider look-ahead (6 ms for Opus, with 5-10 ms being
typical), but also the "memory" from the previous frame.

> Packet boundaries of audio encoded using codecs such as G.711 or
> Opus should create a continuous stream as far as RTP timestamps are
> concerned with RFC 4733 telephone event packets unless
> discontinuous RTP is used. It should be possible to align decoded
> audio and telephone-events based on RTP timestamps.

What do you mean "create a continuous stream"? I thought you wanted
the codec to stop transmitting.

	Jean-Marc
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To: Jean-Marc Valin <jmvalin@mozilla.com>
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--001a11346410d01769052a54565e
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On Wed, Jan 27, 2016 at 12:17 PM, Jean-Marc Valin <jmvalin@mozilla.com>
wrote:

> The problem I see is that unlike G.711, modern codecs (not just Opus)
> aren't stateless. It's not a simple thing to just stop and restart
> transmission during telephony events -- especially if you want to
> avoid glitches and want all implementations to behave the same. You
> need to consider look-ahead (6 ms for Opus, with 5-10 ms being
> typical), but also the "memory" from the previous frame.
>

In this regard Opus is no different then G.729 which was used with
telephone-events for ages. Sending a telephone-event packet is an audio
codec switch. Its behavior should be no different then switch from Opus to
PCMU or from Opus to CN and back. I assume statefull codec such as Opus
should reset its state before starting to send audio after such switch
occurs. Is this correct? From what I know this was never defined anywhere
and it might be a good idea from the codec designers to specify what is the
best behavior for codec switches from telephone-event or CN. Since most of
codecs converge very quickly I see this implemented equally often with or
without codec state reset. Also, since telephone-events are normally
followed by a period of silence (intertone gap), codec state plays much
smaller role in practice.

> Packet boundaries of audio encoded using codecs such as G.711 or
> > Opus should create a continuous stream as far as RTP timestamps are
> > concerned with RFC 4733 telephone event packets unless
> > discontinuous RTP is used. It should be possible to align decoded
> > audio and telephone-events based on RTP timestamps.
>
> What do you mean "create a continuous stream"? I thought you wanted
> the codec to stop transmitting.
>

Yes, I do want the code to stop transmitting. I want the clean codec switch
from non-event audio codec to telephone-event and back. What I mean here is
if you have PCMU packet with timestamp 1000 and duration 160 the following
telephone-event start timestamp should be 1160. The transition in other
direction should also happen without the timestamp gap so that if
telephone-event start timestamp is 1160 and the total tone duration is 800
then next non event audio packet should start with timestamp 1960.
Furthermore, I like to recommend to switch from non-event to
telephone-event and back on the regular packet boundary. This means no
partial audio packets (17 ms audio packet before telephone event) and no
gaps or overlaps after telephone-event (no telephone events with duration
117 ms which causes subsequent audio packet either start before the end of
the telephone-event or with a 3 ms gap after the event or exactly after the
telephone event but with 3 ms offset in comparison with audio packets
before telephone events).

Sorry for spending so much time on telephone-event generation but equipment
generating broken telephone-event streams and then causing havoc in the
phone network is something that is something that wasted couple of weeks of
my time this month alone. I simply want to provide the list of
recommendations for rtcweb which will cause the least amount of interop
trouble in the future.

Regards,
_____________
Roman Shpount

--001a11346410d01769052a54565e
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Wed, Jan 27, 2016 at 12:17 PM, Jean-Marc Valin <span dir=3D"ltr">&l=
t;<a href=3D"mailto:jmvalin@mozilla.com" target=3D"_blank">jmvalin@mozilla.=
com</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wi=
dth:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-=
left:1ex">The problem I see is that unlike G.711, modern codecs (not just O=
pus)<br>
aren&#39;t stateless. It&#39;s not a simple thing to just stop and restart<=
br>
transmission during telephony events -- especially if you want to<br>
avoid glitches and want all implementations to behave the same. You<br>
need to consider look-ahead (6 ms for Opus, with 5-10 ms being<br>
typical), but also the &quot;memory&quot; from the previous frame.<br></blo=
ckquote><div><br></div><div>In this regard Opus is no different then G.729 =
which was used with telephone-events for ages. Sending a telephone-event pa=
cket is an audio codec switch. Its behavior should be no different then swi=
tch from Opus to PCMU or from Opus to CN and back. I assume statefull codec=
 such as Opus should reset its state before starting to send audio after su=
ch switch occurs. Is this correct? From what I know this was never defined =
anywhere and it might be a good idea from the codec designers to specify wh=
at is the best behavior for codec switches from telephone-event or CN. Sinc=
e most of codecs converge very quickly I see this implemented equally often=
 with or without codec state reset. Also, since telephone-events are normal=
ly followed by a period of silence (intertone gap), codec state plays much =
smaller role in practice.</div><div><br></div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-co=
lor:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span class=
=3D"">&gt; Packet boundaries of audio encoded using codecs such as G.711 or=
<br>
&gt; Opus should create a continuous stream as far as RTP timestamps are<br=
>
&gt; concerned with RFC 4733 telephone event packets unless<br>
&gt; discontinuous RTP is used. It should be possible to align decoded<br>
&gt; audio and telephone-events based on RTP timestamps.<br>
<br>
</span>What do you mean &quot;create a continuous stream&quot;? I thought y=
ou wanted<br>
the codec to stop transmitting.<br></blockquote><div><br></div><div>Yes, I =
do want the code to stop transmitting. I want the clean codec switch from n=
on-event audio codec to telephone-event and back. What I mean here is if yo=
u have PCMU packet with timestamp 1000 and duration 160 the following telep=
hone-event start timestamp should be 1160. The transition in other directio=
n should also happen without the timestamp gap so that if telephone-event s=
tart timestamp is 1160 and the total tone duration is 800 then next non eve=
nt audio packet should start with timestamp 1960. Furthermore, I like to re=
commend to switch from non-event to telephone-event and back on the regular=
 packet boundary. This means no partial audio packets (17 ms audio packet b=
efore telephone event) and no gaps or overlaps after telephone-event (no te=
lephone events with duration 117 ms which causes subsequent audio packet ei=
ther start before the end of the telephone-event or with a 3 ms gap after t=
he event or exactly after the telephone event but with 3 ms offset in compa=
rison with audio packets before telephone events).</div><div><br></div><div=
>Sorry for spending so much time on telephone-event generation but equipmen=
t generating broken telephone-event streams and then causing havoc in the p=
hone network is something that is something that wasted couple of weeks of =
my time this month alone. I simply want to provide the list of recommendati=
ons for rtcweb which will cause the least amount of interop trouble in the =
future.</div><div><br></div><div>Regards,</div><div><div class=3D"gmail_sig=
nature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></=
div></div>

--001a11346410d01769052a54565e--


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From: Alissa Cooper <alissa@cooperw.in>
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Subject: [rtcweb] AD evaluation: draft-ietf-rtcweb-audio-09
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I have reviewed this document in preparation for IETF LC. The document =
is loose with terminology =E2=80=94 =E2=80=9CWebRTC endpoint =
implementations,=E2=80=9D =E2=80=9CWebRTC endpoints,=E2=80=9D =E2=80=9Cthe=
 browser,=E2=80=9D "WebRTC compatible endpoints,=E2=80=9D etc. I think =
this document needs to reference draft-ietf-rtcweb-overview and =
consistently use the terminology defined there (or, if not, explain why =
not). Once those changes are made, I believe this document will be ready =
for last call.

Alissa=

--Apple-Mail=_822DFCE6-D04E-4CE8-92B3-217FECEBCA79
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D"">I have reviewed this document in preparation for IETF LC. =
<span style=3D"widows: 1;" class=3D"">The document is loose with =
terminology =E2=80=94 =E2=80=9CWebRTC endpoint implementations,=E2=80=9D =
=E2=80=9CWebRTC endpoints,=E2=80=9D =E2=80=9Cthe browser,=E2=80=9D =
"</span><span style=3D"widows: 1;" class=3D""><font size=3D"2" =
class=3D"">WebRTC compatible endpoints,=E2=80=9D&nbsp;</font></span><span =
style=3D"widows: 1;" class=3D"">etc. I think this document needs to =
reference&nbsp;draft-ietf-rtcweb-overview and consistently use the =
terminology defined there (or, if not, explain why not). Once those =
changes are made, I believe this document will be ready for last =
call.</span><div class=3D""><span style=3D"widows: 1;" class=3D""><br =
class=3D""></span></div><div class=3D""><span style=3D"widows: 1;" =
class=3D"">Alissa</span></div></body></html>=

--Apple-Mail=_822DFCE6-D04E-4CE8-92B3-217FECEBCA79--


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Subject: [rtcweb] AD evaluation: draft-ietf-rtcweb-audio-codecs-for-interop-04
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I have reviewed this document in preparation for IETF LC. I have a few =
substantive comments that need to be resolved for IETF LC. I=E2=80=99ve =
also included some nits below that should be resolved together with LC =
comments.

Substantive comments:

(1) Typically when a long author list gets shifted to the =
Acknowledgments section the remaining name at the top get marked as an =
editor. That seems appropriate in this case.

(2) The document uses the term =E2=80=9CWebRTC client=E2=80=9D in a few =
places. I would suggest referencing the terminology defined in =
draft-ietf-rtcweb-overview regarding endpoints/devices/browsers and =
consistently using the terminology defined there (or, if not, explain =
why not).

(3) The claims throughout Section 4 about the number of terminals =
affected sound like marketing. Wanted to check if the WG is confident =
that these claims need to be in the document.

(4) Section 4.4 seems superfluous and should be deleted.


Nits:

=3D Section 1 =3D

s/to include in the offer other suitable audio codecs/to include in the =
offer other suitable audio codecs beyond those that are mandatory to =
implement/

=3D Section 5 =3D

s/to also report more specifically/to also refer more specifically/


From nobody Thu Jan 28 03:08:41 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Transports for WebRTC
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-11.txt
	Pages           : 16
	Date            : 2016-01-28

Abstract:
   This document describes the data transport protocols used by WebRTC,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-transports-11

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-11


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Thu Jan 28 03:16:52 2016
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From: Harald Alvestrand <harald@alvestrand.no>
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The main change in this version is to define the terms "data flow" and
"media flow", which are referenced by the -qos draft that is about to be
submitted for publication.

This version closes both open issues I had in github
(https://github.com/rtcweb-wg/rtcweb-transport/issues).

Harald

Den 28. jan. 2016 12:08, skrev internet-drafts@ietf.org:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
> 
>         Title           : Transports for WebRTC
>         Author          : Harald Alvestrand
> 	Filename        : draft-ietf-rtcweb-transports-11.txt
> 	Pages           : 16
> 	Date            : 2016-01-28
> 
> Abstract:
>    This document describes the data transport protocols used by WebRTC,
>    including the protocols used for interaction with intermediate boxes
>    such as firewalls, relays and NAT boxes.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/
> 
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-transports-11
> 
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-11
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 


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Subject: [rtcweb] Plan B and SDP with different payload numbers pointing to the same codec
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Hi,

In Plan B [*] if a SFU receives VP8 tracks from 3 peers, using each
peer a different
SSRC and payload type/number: Can the SFU generate a SDP containing a
single m=3Dvideo line with 3 different payload numbers pointing to the
same VP8 codec?

The draft seems to assume that the SFU/server will translate/override
the payloadType of the incoming sources and make them the same.

Thanks a lot.


[*] https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00

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


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From: Adam Roach <adam@nostrum.com>
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Subject: Re: [rtcweb] Plan B and SDP with different payload numbers pointing to the same codec
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Um... the working group decided not to move forward with this mechanism. 
I'm not sure it was ever fully formed enough to be able answer your 
question.

Put another way: you're implementing a non-standard solution. 
Interoperability is a matter of between you and other people wanting to 
implement your proprietary approach.

/a

On 1/28/16 14:05, Iñaki Baz Castillo wrote:
> Hi,
>
> In Plan B [*] if a SFU receives VP8 tracks from 3 peers, using each
> peer a different
> SSRC and payload type/number: Can the SFU generate a SDP containing a
> single m=video line with 3 different payload numbers pointing to the
> same VP8 codec?
>
> The draft seems to assume that the SFU/server will translate/override
> the payloadType of the incoming sources and make them the same.
>
> Thanks a lot.
>
>
> [*] https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00
>


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2016-01-28 22:28 GMT+01:00 Adam Roach <adam@nostrum.com>:
> Um... the working group decided not to move forward with this mechanism. =
I'm
> not sure it was ever fully formed enough to be able answer your question.
>
> Put another way: you're implementing a non-standard solution.
> Interoperability is a matter of between you and other people wanting to
> implement your proprietary approach.

Hi Adam,

I don't want Plan A, Plan B or Unified-Plan. But... SDP + Plan B is
what current Google's libwebrtc implements so I need to interop with
it.

Said that, I agree that Plan B is an abandoned draft so I shouldn't
expect a specific response to my question.

Thanks a lot.

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


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This isn't the right form to talk about Plan B.  Perhaps
discuss-webrtc@googlegroups.com would be.

On Thu, Jan 28, 2016 at 1:34 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> 2016-01-28 22:28 GMT+01:00 Adam Roach <adam@nostrum.com>:
> > Um... the working group decided not to move forward with this mechanism=
.
> I'm
> > not sure it was ever fully formed enough to be able answer your questio=
n.
> >
> > Put another way: you're implementing a non-standard solution.
> > Interoperability is a matter of between you and other people wanting to
> > implement your proprietary approach.
>
> Hi Adam,
>
> I don't want Plan A, Plan B or Unified-Plan. But... SDP + Plan B is
> what current Google's libwebrtc implements so I need to interop with
> it.
>
> Said that, I agree that Plan B is an abandoned draft so I shouldn't
> expect a specific response to my question.
>
> Thanks a lot.
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">This isn&#39;t the right form to talk about Plan B.=C2=
=A0 Perhaps <a href=3D"mailto:discuss-webrtc@googlegroups.com" target=3D"_b=
lank">discuss-webrtc@googlegroups.com</a>=C2=A0would be.</div><div class=3D=
"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Jan 28, 2016 at 1:34 P=
M, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@alia=
x.net" target=3D"_blank">ibc@aliax.net</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"><span>2016-01-28 22:28 GMT+01:00 Adam Roach &lt;<a href=
=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt;:<br=
>
&gt; Um... the working group decided not to move forward with this mechanis=
m. I&#39;m<br>
&gt; not sure it was ever fully formed enough to be able answer your questi=
on.<br>
&gt;<br>
&gt; Put another way: you&#39;re implementing a non-standard solution.<br>
&gt; Interoperability is a matter of between you and other people wanting t=
o<br>
&gt; implement your proprietary approach.<br>
<br>
</span>Hi Adam,<br>
<br>
I don&#39;t want Plan A, Plan B or Unified-Plan. But... SDP + Plan B is<br>
what current Google&#39;s libwebrtc implements so I need to interop with<br=
>
it.<br>
<br>
Said that, I agree that Plan B is an abandoned draft so I shouldn&#39;t<br>
expect a specific response to my question.<br>
<br>
Thanks a lot.<br>
<span><br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<br>
<br>
</span><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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--001a113d616aa52bf1052a6bc861--


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From: Ted Hardie <ted.ietf@gmail.com>
Date: Fri, 29 Jan 2016 11:10:40 -0800
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Subject: [rtcweb] Doodle for Virtual Interim timing
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--001a114a7c6eb0e861052a7dcb04
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Howdy,

The chairs have heard from the JSEP editor team that there is likely be to
a new version of JSEP in early march.  In order to make as much as progress
as possible, we'd like to consider a two-hour virtual interim just after
that to discuss the changes and look any further issues that need
discussion at IETF 95.  Below is a doodle poll for approximate timing:

http://doodle.com/poll/tn2uavay8bhq4iwe


Please fill it out if you would consider attending, and let the chairs know
if there are concerns,

thanks,

Sean, Cullen, Ted

--001a114a7c6eb0e861052a7dcb04
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:garamond=
,serif;font-size:small">Howdy,<br><br></div><div class=3D"gmail_default" st=
yle=3D"font-family:garamond,serif;font-size:small">The chairs have heard fr=
om the JSEP editor team that there is likely be to a new version of JSEP in=
 early march.=C2=A0 In order to make as much as progress as possible, we&#3=
9;d like to consider a two-hour virtual interim just after that to discuss =
the changes and look any further issues that need discussion at IETF 95.=C2=
=A0 Below is a doodle poll for approximate timing:<br><br><a href=3D"http:/=
/doodle.com/poll/tn2uavay8bhq4iwe" rel=3D"noreferrer" target=3D"_blank">htt=
p://doodle.com/poll/tn2uavay8bhq4iwe</a><br>
<br><br></div><div class=3D"gmail_default" style=3D"font-family:garamond,se=
rif;font-size:small">Please fill it out if you would consider attending, an=
d let the chairs know if there are concerns,<br><br></div><div class=3D"gma=
il_default" style=3D"font-family:garamond,serif;font-size:small">thanks,<br=
><br></div><div class=3D"gmail_default" style=3D"font-family:garamond,serif=
;font-size:small">Sean, Cullen, Ted<br></div></div>

--001a114a7c6eb0e861052a7dcb04--


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Subject: Re: [rtcweb] Doodle for Virtual Interim timing
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On 30 January 2016 at 06:10, Ted Hardie <ted.ietf@gmail.com> wrote:
> http://doodle.com/poll/tn2uavay8bhq4iwe


I assume that in the absence of time zone information, this is Pacific Time. No?


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--001a11416fbc512957052aaaef92
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That was my assumption looking at the offered slots :)

On Sunday, January 31, 2016, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 30 January 2016 at 06:10, Ted Hardie <ted.ietf@gmail.com <javascript:;>>
> wrote:
> > http://doodle.com/poll/tn2uavay8bhq4iwe
>
>
> I assume that in the absence of time zone information, this is Pacific
> Time. No?
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <javascript:;>
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11416fbc512957052aaaef92
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That was my assumption looking at the offered slots :)<br><br>On Sunday, Ja=
nuary 31, 2016, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.c=
om">martin.thomson@gmail.com</a>&gt; wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">On 30 January 2016 at 06:10, Ted Hardie &lt;<a href=3D"javascript:;" onc=
lick=3D"_e(event, &#39;cvml&#39;, &#39;ted.ietf@gmail.com&#39;)">ted.ietf@g=
mail.com</a>&gt; wrote:<br>
&gt; <a href=3D"http://doodle.com/poll/tn2uavay8bhq4iwe" target=3D"_blank">=
http://doodle.com/poll/tn2uavay8bhq4iwe</a><br>
<br>
<br>
I assume that in the absence of time zone information, this is Pacific Time=
. No?<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;rtcweb@i=
etf.org&#39;)">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>

--001a11416fbc512957052aaaef92--

