<div dir="ltr"><div><div><div><div><div><div><div><div>Hi,<br><br></div>I am trying to make an outbound call to a webrtc softphone using jssip, I initiate the call from an asterisk box :<br><br></div>[Asterisk] -> [FS] ->[jssip]<br>
<br></div>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<br>
<br>2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:<br>v=0<br>o=root 451068671 451068671 IN IP4 192.168.90.16<br>s=Asterisk PBX 1.8.16.0<br>c=IN IP4 192.168.90.16<br>t=0 0<br>m=audio 16126 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br><br></div>And this is the SDP from FS wich is sent to the jssip client<br><br>2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:<br>v=0<br>
o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP<br>s=FreeSWITCH<br>c=IN IP4 PUBLIC_IP<br>t=0 0<br>a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps<br>m=audio 21772 RTP/SAVPF 8 0 101 13<br>a=rtpmap:101 telephone-event/8000<br>
a=rtcp-mux<br>a=rtcp:21772 IN IP4 PUBLIC_IP<br>a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS<br>a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0<br>a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps<br>a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0<br>
a=ice-ufrag:4BgP20qzvjuPqK7E<br>a=ice-pwd:ADcc0S2Lqc2T5cB0<br>a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host generation 0<br>a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host generation 0<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7<br>a=ptime:20<br>a=sendrecv<br><br></div>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<br>
<br> <action application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/><br><br></div>before bridge, but no luck.<br><br></div>Funny thing is that it is working if i innitiate the call from the jssip client. [jssip] -> [FS] ->[Asterisk]<br>
<br></div>Any clue?<br><div><div><div><br><br><div><br></div></div></div></div></div>