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

[Harald Alvestrand <[e-mail]>]

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?

I mean - if a G.711 receiver is able to handle any packet size thrown at
it, why do we need to put requirements on the sender (apart from "don't
send bigger packets than the maxptime")? For interoperability with
non-WebRTC endpoints?

Of course, I do have a question, now that all the ptime and maxtime
definitions have been put on the table:

Is it true that ptime and maxptime only specify the SDP sender's
declaration of his desires for incoming packet streams? That is, that
there is nowhere specified what the sender intends to send? I think that
is true for ptime in RFC 3264 offer/answer, and would assume that the
same is true for maxptime.