Re: [rtcweb] A problem with both A and B

Christer Holmberg <christer.holmberg@ericsson.com> Tue, 21 May 2013 07:20 UTC

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] A problem with both A and B
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Date: Tue, 21 May 2013 07:20:18 +0000
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Subject: Re: [rtcweb] A problem with both A and B
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Hi,

>> I don't think it's a BUNDLE issue whether adding/removing of streams require an O/A exchange or not. It's an SDP, and SDP O/A, issue.
>>
>> BUNDLE is about sharing a 5-tuple.
>
> I don't agree.
>
> RTP is already able to support multiple RTP streams sharing an RTP session (and 5-tuple).
>
> What BUNDLE is about is describing that in SDP.
> And that includes how SDP O/A works.

Multiple RTP streams can already share an RTP session today, using multiple SSRCs per m- line :)

If adding a new stream means adding a new m- line, then an SDP O/A is obviously needed - bundle or no bundle.

If adding a new stream means adding a new SSRC, within an existing m-, then it is also an SDP O/A issue - bundle or no bundle.

Regards,

Christer



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
> Behalf Of Martin Thomson
> Sent: 17. toukokuuta 2013 23:50
> To: Bernard Aboba
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] A problem with both A and B
>
>> "Dale R. Worley" <worley@ariadne.com> wrote:
>>     DES F11  It must be possible to add and remove one way video flows
>>        within the bundle without requiring an additional offer/answer
>>        cycle.
>
> I'm not sure why this speaks specifically about video...
>
> On 17 May 2013 13:21, Bernard Aboba <bernard_aboba@hotmail.com> wrote:
>> It is important. In fact, I would argue it is critical for congestion control (e.g. removal of simulcast or layered streams by the sender should not require an O/A exchange).
>
> This is different I think.  Responding to congestion only requires 
> that it be possible to stop sending a given stream.  (Or maybe crank 
> down resolution or frame rate.)
>
> This item talks about addition.  That's an entirely different proposition.  As long as we assume constraints (you don't need to re-ICE a new transport for the stream, you don't use new packet types and codec profiles), this could be possible.  But to get WebRTC to support this requires API changes.
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