Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01

> I don't think this points to a need to push AMR into the web browsers,
> mind you. To take Justin's earlier point a few steps further: if we're
> going on number of shipping units, the mere existence of Chrome and
> Firefox's WebRTC implementation means that the number of deployed Opus
> codecs far overwhelms the number of deployed AMR codecs. Appeals to the
> size of codec deployment would more reasonably reach the conclusion that
> Opus should be MTI for the next 3GPP release. ;-)

You are welcome to create the relevant work item in 3GPP SA4 if you think it appropriate.

That of course will not solve the need to transcode with all the phones prior to the release where that work item is adopted.

Keith

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Adam Roach
> Sent: 15 March 2013 12:42
> To: Andrew Allen
> Cc: rtcweb@ietf.org; xavier.marjou@orange.com
> Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-
> codecs-for-interop-01
> 
> On 3/14/13 17:09, Andrew Allen wrote:
> > Koen is right that there are many more obstacles to RTCweb and legacy
> network interop than just a common codec. First there will need to be
> RTCweb signaling gateways to interface between the RTCweb signaling and
> the legacy networks (SIP, H.323 etc) and there will need to be in place
> mechanisms for peering, federation and address resolution between networks
> as well as service agreements in place between the players.
> 
> 
> We had this mostly working at the SIPit in Boston, using the SIP over
> Websockets spec. The only reason we weren't actually setting calls up
> was that we couldn't find any clients there that supported both ICE and
> DTLS-SRTP. What's neat is that this isn't even really "gatewaying" in
> the way that you mean it -- all you need is a SIP proxy that supports
> one more (easy-to-implement) transport protocol.
> 
> And once you hit that proxy, all of the peering, federation, and address
> resolution solutions developed for SIP apply.
> 
> I don't think this points to a need to push AMR into the web browsers,
> mind you. To take Justin's earlier point a few steps further: if we're
> going on number of shipping units, the mere existence of Chrome and
> Firefox's WebRTC implementation means that the number of deployed Opus
> codecs far overwhelms the number of deployed AMR codecs. Appeals to the
> size of codec deployment would more reasonably reach the conclusion that
> Opus should be MTI for the next 3GPP release. ;-)
> 
> /a
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