Re: [MMUSIC] Telephone-event and multiple clock rates

Magnus Westerlund <magnus.westerlund@ericsson.com> Mon, 27 January 2014 08:47 UTC

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Date: Mon, 27 Jan 2014 09:47:18 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Adam Roach <adam@nostrum.com>, "Stach, Thomas" <thomas.stach@unify.com>
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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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On 2014-01-24 16:59, Adam Roach wrote:
> On 1/24/14 04:29, Magnus Westerlund wrote:
>> My understanding for the reasoning behind this definition in the RFC is
>> due to that the events is supposed to be connected to the same time-line
>> as the audio. Thus it ended up using the same SSRC. To avoid timestamp
>> rate switching which we know has issues
>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-rates/)
>>
>> it was mandated to provide one PT for each rate negotiated for the audio
>> payload types.
> 
> To make sure I understand what you're saying correctly: you are
> asserting that having four different clock rates for voice codecs
> requires us to use four different PTs for telephone-event if we're going
> to have them on the same SSRC. Is that correct?

Yes!

All to avoid the issue that with rate switching within an SSRC. This is
both potential loss of time synchronization as well as degrading RTP
timestamp based RTCP measurements. And the above draft (soon RFC) does
require new implementations to change their SSRC when doing
rate-switching. That clearly have signaling implications in the context
of WebRTC.

Cheers

Magnus Westerlund

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