[Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION

Anthony Minessale anthony.minessale at gmail.com
Tue Aug 6 17:26:50 MSD 2013


try prepending the bridge url with {absolute_codec_string=PCMA}


On Tue, Aug 6, 2013 at 6:32 AM, Javier Menendez
<menendez.garcia at gmail.com>wrote:

> Hi,
>
> I am trying to make an outbound call to a webrtc softphone using jssip, I
> initiate the call from an asterisk box :
>
> [Asterisk] -> [FS] ->[jssip]
>
> I always get an  INCOMPATIBLE_DESTINATION error, looking at the trace logs
> I found out that the problem is the codec negotiation but I can not make it
> work, AFAIK the call should use the ALAW codec as it is compatible with all
> legs involved. this is the asterisk SDP
>
> 2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:
> v=0
> o=root 451068671 451068671 IN IP4 192.168.90.16
> s=Asterisk PBX 1.8.16.0
> c=IN IP4 192.168.90.16
> t=0 0
> m=audio 16126 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> And this is the SDP from FS wich is sent to the jssip client
>
> 2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:
> v=0
> o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP
> s=FreeSWITCH
> c=IN IP4 PUBLIC_IP
> t=0 0
> a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps
> m=audio 21772 RTP/SAVPF 8 0 101 13
> a=rtpmap:101 telephone-event/8000
> a=rtcp-mux
> a=rtcp:21772 IN IP4 PUBLIC_IP
> a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS
> a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0
> a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps
> a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0
> a=ice-ufrag:4BgP20qzvjuPqK7E
> a=ice-pwd:ADcc0S2Lqc2T5cB0
> a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host
> generation 0
> a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host
> generation 0
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7
> a=ptime:20
> a=sendrecv
>
> Shouldn't it include the PCMA codec? I hace tried to enable it in
> configuration but it doesn't work, also tried to set it with
>
>   <action application="export"
> data="nolocal:absolute_codec_string=PCMA,PCMU"/>
>
> before bridge, but no luck.
>
> Funny thing is that it is working if i innitiate the call from the jssip
> client. [jssip] -> [FS] ->[Asterisk]
>
> Any clue?
>
>
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>


-- 
Anthony Minessale II

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