[Freeswitch-users] How to call B-leg without opus rtpmap ?

[Oancea, Dragos]

Yes, the rate of the telephone-event must be the rate of the audio codec.
Out of curiosity, what gateway is this ?
As much as I hate the idea, you must not offer Opus to the B-leg in this
case.

You can try this just before you bridge:
<action application="set" data="codec_string=OPUS,PCMA"/>
<action application="set" data="inbound_codec_prefs=OPUS,PCMA"/>
<action application="export" data="nolocal:absolute_codec_string=PCMA"/>

Cheers,
Dragos


On Wed, Nov 1, 2017 at 9:08 AM, Zheka Polivoda <poliv78 at yahoo.co.uk> wrote:

> Hi all
> I'm using freeswitch to call from browser(webrtc) to some sip provider.
> So, the A-keg is webrtc opus codec and then I bridge it with the sip
> gateway.
> Sip gateway supports g711 only. And it chooses wrong codec and DTMF is not
> working between them.
> When bridging to sip, freeswitch sends such SDP
>
>
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a Local SDP:
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a v=0
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a o=FreeSWITCH 1509346805 1509346806
> IN IP4 x.x.x.x
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a s=FreeSWITCH
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a c=IN IP4 x.x.x.x
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a t=0 0
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a m=audio 31568 RTP/AVP 102 9 0 8 103
> 101
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:102 opus/48000/2
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:102 useinbandfec=1;
> maxaveragebitrate=14400; maxplaybackrate=8000; sprop-maxcapturerate=8000;
> ptime=20; minptime=10; maxptim
> e=40; stereo=1
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:9 G722/8000
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:0 PCMU/8000
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:8 PCMA/8000
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:103 telephone-event/48000
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:103 0-16
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:101 telephone-event/8000
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:101 0-16
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=ptime:20
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=sendrecv
>
>
>
> Is it possible to send SDP without opus rtpmap? Because SIP gateway choose
> wrong
> telephone-event. the answer of sip gateway is
>
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a 2017-10-30 15:46:13.248384 [DEBUG]
> sofia.c:7058 Remote SDP:
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a v=0
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a o=lvp 8000 8000 IN IP4 y.y.y.y
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a s=SIP Call
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a c=IN IP4 y.y.y.y
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a t=0 0
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a m=audio 22786 RTP/AVP 8 103
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:8 PCMA/8000
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:103 telephone-event/48000
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:103 0-16
> 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=ptime:20
>
>
> I guess there must be
> a=rtpmap:101 telephone-event/8000
>
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20171101/56308f94/attachment-0001.html>