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
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
