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Re: [rtcweb] draft-ietf-rtcweb-rtp-usage: CSRC in the API?

From: Emil Ivov <emcho@jitsi.org>
Date: Wed, 27 Feb 2013 20:25:11 +0200
Message-ID: <512E4F87.5000308@jitsi.org>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
CC: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
On 27.02.13, 20:16, Cullen Jennings (fluffy) wrote:
> I think this is very useful for a bunch of things in conferences.

+1

> I'd
> like to see an API so the CSRC list for outgoing RTP can be set on a
> per track basis, and the API allows the JS to get the CSRC of most
> recently received RTP packets for a given PC-Track and also be nice
> to have a callback any time the received CSRC changes.

+1 again. It would also if such an API would also the possibility to
retrieve audio levels from the RTP (Magnus already mentioned RFC6465)
so that applications can render such information.

(Like this for example: http://goo.gl/QTaqy )

Cheers,
Emil
> 
> 
> On Feb 22, 2013, at 1:19 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
> 
>> WebRTC WG,
>> 
>> As editor of the RTP usage specification in RTCWEB WG, we have had
>> a noted issue in our draft specification 
>> (https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/).
>> In Section 12.5. of version 05 (Contributing Sources) we had the
>> following:
>> 
>> (tbd: does the API need to provide the ability to add a CSRC list
>> to an outgoing packet? this is only useful if the sender is mixing 
>> content)
>> 
>> This is clearly an API question. We intended to remove it. However,
>> I like to hand it over to you in W3C to consider on the impact on
>> the API this has.
>> 
>> From my personal view point this has two aspects:
>> 
>> Exposing when a received MediaStreamTrack is actually the mix (or 
>> switch) of other MediaStreamTracks, a mix performed by a WebRTC
>> endpoint or RTP middlebox (Mixer). Applications that like to know
>> who are currently seen or audible needs this information mapping.
>> We also have specified an optional to support RTP header extension
>> (RFC6465) that provide energy levels for each contributing source.
>> If that is used, that information would be something an application
>> would like to render somehow.
>> 
>> The other aspect is when an WebRTC endpoint mixes media to produce
>> a new MediaStreamTrack, for example with the Web Audio API, then
>> one need to consider if and how the CSRC list is populated.
>> 
>> Cheers
>> 
>> Magnus Westerlund
>> 
>> ----------------------------------------------------------------------
>>
>> 
Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>>
>> 
Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079 SE-164 80
>> Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com 
>> ----------------------------------------------------------------------
>>
>>
>> 
_______________________________________________
>> rtcweb mailing list rtcweb@ietf.org 
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> _______________________________________________ rtcweb mailing list 
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> 

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Received on Wednesday, 27 February 2013 22:18:58 UTC

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