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

I agree with Haralds comments on this.

I do not see any point in writing RTCWEB specific requirements on this.

Keith 

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of 
> Magnus Westerlund
> Sent: 25 February 2014 10:24
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Number of samples (ptime) to be 
> supported by required codecs (draft-ietf-rtcweb-audio-05)
> 
> 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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