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Re: [rtcweb] How to multiplex between peers

Bala Pitchandi <Bala@vidyo.com> Wed, 20 July 2011 20:57 UTC

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From: Bala Pitchandi <Bala@vidyo.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>, Justin Uberti <juberti@google.com>
Date: Wed, 20 Jul 2011 16:57:27 -0400
Thread-Topic: [rtcweb] How to multiplex between peers
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] How to multiplex between peers
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+1 for sending audio & video (and any other type of media, for that matter) on a single RTP session. With RTCP mux, that means all media can be sent on a single UDP port. 

-- Bala


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Matthew Kaufman
Sent: Wednesday, July 20, 2011 3:51 PM
To: Justin Uberti
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] How to multiplex between peers


On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:

> 
> This is where I've ended up on this topic. We can easily multiplex multiple RTP sources (of the same type) over a single RTP session using SSRC. We can also mux RTCP over the RTP session using RTCP mux. So, for an arbitrary video call, we have just 2 RTP sessions/NAT bindings. 
> 
> Is it worth going the extra mile to get down to 1 in v1.0, given the lack of consensus that exists right now? Is there even a compelling argument to do so?

Yes and yes.

I really can't understand why, if we can multiplex 3 totally different types of video streams over the same RTP session using SSRC (with wildly different bit-rates, inter-packet times, etc.) we can't also multiplex audio and video. Not a single argument that has been presented has convinced me, and getting from 2 to 1 is a *50%* reduction in NAT port utilization. (And a significant reduction in the number of "strange" failure modes, where one traversal worked and the other didn't.)

Matthew Kaufman

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