Hi Brian Am using 1.2.13+git~20130919T142456Z~6afb17a138 It is strange because in the debug CLI it constructs some SDP (for sending back 183 early media) which looks sane: 2014-01-28 14:38:05.000169 [DEBUG] mod_sofia.c:2876 Ring SDP: v=0 o=FreeSWITCH 1390904357 1390904358 IN IP4 95.211.119.213 s=FreeSWITCH c=IN IP4 95.211.119.213 t=0 0 m=audio 15528 RTP/AVP 111 101 a=rtpmap:111 opus/48000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv then the SDP sent out is different (port is now 0 and opus is now mapped to 100). This is from the ngrep trace: . v=0. o=FreeSWITCH 1390904357 1390904358 IN IP4 95.211.119.213. s=FreeSWITCH. c=IN IP4 95.211.119.213. t=0 0. m=audio 0 RTP/AVP 100 101. a=rtpmap:100 opus/48000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. m=video 0 RTP/AVP 19. On 28 January 2014 14:38, Brian West <brian@freeswitch.org> wrote:
It works fine out of the box with WebRTC; you just allow OPUS on your codec list. Are you having a negotiation failure? What rev of FreeSWITCH are you using? -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
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On Jan 28, 2014, at 8:27 AM, Pete Kelly <pkelly@gmail.com> wrote:
Hi
I am trying to use the opus codec built into FreeSWITCH to talk to the pjsip library (running on an Android handset).
There seems to be very little docs on opus support within FreeSWITCH outside the original post saying Anthony has got it working in 2011?
Does anyone have any experience with using this codec with FreeSWITCH?
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