Re: [rtcweb] JSEP: Issues with a=ssrc and RTP payload type switching
Harald Alvestrand <harald@alvestrand.no> Mon, 08 June 2015 15:13 UTC
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Date: Mon, 08 Jun 2015 17:09:07 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, Simon Perreault <sperreault@jive.com>, Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] JSEP: Issues with a=ssrc and RTP payload type switching
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On 06/08/2015 04:43 PM, Magnus Westerlund wrote: > Simon Perreault skrev den 2015-06-08 16:15: >> Le 2015-06-08 03:36, Magnus Westerlund a écrit : >>> Harald Alvestrand skrev den 2015-06-08 11:19: >>>> Den 08. juni 2015 10:48, skrev Magnus Westerlund: >>> >>>>> However, this can't work as RFC 7160 says in Section 4.1: >>>>> >>>>> An RTP Sender with RTCP turned on MUST use a different SSRC >>>>> for each >>>>> different clock rate. An RTCP BYE MUST be sent and a new >>>>> SSRC MUST >>>>> be used if the clock rate switches back to a value already >>>>> seen in >>>>> the RTP stream. >>>>> >>>>> Note the second sentence. One would still have to do a new round of >>>>> signalling to prime a new SSRC for the previous rate after having >>>>> switched to be able to switch back to that Payload Type. >>>> >>>> Hm. I missed that when reviewing the clock-rate doc. >>>> So that means one uses up an SSRC for every clock rate switch. That >>>> seems silly; is it possible that this should be considered an erratum >>>> for RFC 7160? >>> >>> No, it is highly intentional as I remember the discussion. >> >> Can you please explain, for those of us with short memories? :) > > I went and checked this. This formulation has been present since the > first versions, and no one has questioned this RFC 2119 worded > statement. Therefore I consider there to be consensus behind it and > being intentional. > > The motivation I see for this statement is the following. First of all > I think people have been against considering having stand bye SSRCs > that are really likely to never be used. I think DTMF is one of these > exceptions. Having stand by SSRC also have an overhead cost, primarily > in the result of the number of SSRCs present in the session and their > consumption of their share of the RTCP bandwidth resources. We also > have the issue of correctly associating the SSRC to a media source, > now that there are more than one. > > Based on that, (with the assumption that you do PT switching rarely) > the lower cost is to drop the SSRC after its usage. From that follows > that one stop using it, and therefore sends BYE, then you really > should not revive a dead SSRC. Thus, need to create a new one. Simply > codifying this expectation in the RFC. This has another implication: When doing PT switching for comfort noise, you MUST have a CN with the same clock rate as your normal audio codec. CN is exactly the type of PT-switching that, if it occurs at all, occurs very frequently.
- [rtcweb] JSEP: Issues with a=ssrc and RTP payload… Magnus Westerlund
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Bernard Aboba
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Harald Alvestrand
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Roman Shpount
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Magnus Westerlund
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Harald Alvestrand
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Magnus Westerlund
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Simon Perreault
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Magnus Westerlund
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Simon Perreault
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Harald Alvestrand
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Simon Perreault
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Harald Alvestrand
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Magnus Westerlund
- Re: [rtcweb] JSEP: Issues with a=ssrc and RTP pay… Christer Holmberg