when i add opus to the incoming packet, the call goes through, although in a pass through mode, of course, no transcoding.
asterisk sends this m=audio 19112 RTP/AVP 111 101. a=rtpmap:111 opus/48000/2. a=maxptime:60. a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. sems receive and sends this out m=audio 1028 RTP/AVP 111 101 112 0. a=rtpmap:111 opus/48000/2. a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:112 opus/48000. a=rtpmap:0 pcmu/8000. a=sendrecv. a=maxptime:60. a=silenceSupp:off - - - -. a=ptime:20. notice it still adds opus/48000. Kelvin Chua On Wed, Sep 17, 2014 at 4:51 PM, Juha Heinanen <[email protected]> wrote: > Kelvin Chua writes: > >> how come i have a different experience? lol > > include also opus in the incoming request and see if it makes a > difference. > > -- juha _______________________________________________ Sems mailing list [email protected] http://lists.iptel.org/mailman/listinfo/sems
