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

[Harald Alvestrand <[e-mail]>]

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.

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.


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