[asterisk-dev] "telephone-event" at rates other than 8000?

[Stephen Davies]

Hi,

Reviewing rtp_engine.c it appears that we only support telephone-event rtp
with a sample rate of 8000?

JSSIP using Opus offers Opus as "opus/48000/2" and then (by necessity, I
think), telephone-event/48000.

EG (this is a JSSIP using WebRTC behind a Freeswitch system):

   v=0
   o=FreeSWITCH 1496895595 1496895596 IN IP4 x.y.250.156
   s=FreeSWITCH
   c=IN IP4 x.y.250.156
   t=0 0
   m=audio 28302 RTP/AVP 102 101
   a=rtpmap:102 opus/48000/2
   a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000;
maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
   a=rtpmap:101 telephone-event/48000
   a=fmtp:101 0-16
   a=ptime:20

Asterisk (13) responds with:

   v=0
   o=root 615288785 615288785 IN IP4 x.y.250.132
   s=Telviva
   c=IN IP4 x.y.250.132
   t=0 0
   m=audio 11824 RTP/AVP 102
   a=rtpmap:102 opus/48000/2
   a=fmtp:102 maxaveragebitrate=30000;useinbandfec=1
   a=ptime:20
   a=maxptime:60
   a=sendrecv

So drops the telephone-event.

In rtp_engine.c there is only:

set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);

Has this come up before?

Can any other developer point me as to where I'd need to look to try to add
48000 too?

Thanks,
Steve
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