Re: [rtcweb] Number of samples (ptime) to be supported by required codecs (draft-ietf-rtcweb-audio-05)

[Magnus Westerlund <[e-mail]>]

On 2014-02-19 19:11, Harald Alvestrand wrote:
> On 02/19/2014 09:16 AM, Magnus Westerlund wrote:
>> On 2014-02-18 23:58, Harald Alvestrand wrote:
>>> I may be a little simple-minded, but if we have a recommendation that
>>> receivers MUST be able to receive all packetizations of G.711 and OPUS
>>> up to 200 ms per packet, and that receivers should signal this by
>>> sending "a=maxptime:200" in their SDP, what situations exist where
>>> interoperability is not going to be achieved?
>> Interoperability is going to be achieve in the direction towards this
>> endpoint. The question is if we achieve interoperability in the
>> direction from that endpoint.
> 
> So, given that there is nothing in the G.711 specification about which
> sample sizes a G.711 receiver has to support - the right approach seems
> to be to amend the G.711 MIME registration with this information.

Yes, it would for the future. But considering the wide-spread deployment
of this payload format I don't see that having any short term effect on
the interoperability.

Also, the recommendations are actually context dependent. Thus, it is
not obvious that a single recommendation is suitable.

> 
> This is not a WebRTC issue; it is an absence of specification for the
> non-WebRTC devices that use G.711. The RTCWEB specifications can
> reasonably be expected to point to existing information about this
> issue; it is not reasonable (in my mind) that the RTCWEB WG decide what
> these values should be.
> 

I would argue for this consideration based on that this is done in the
WebRTC context, not any other context.

Cheers

Magnus Westerlund

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