[Freeswitch-users] OPUS codec use

[Pete Kelly]

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 at 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
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> On Jan 28, 2014, at 8:27 AM, Pete Kelly <pkelly at 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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