<div dir="ltr"><div><div><div><div>After trying and trying I found out it is a compatibility version with the browser.<br><br></div>Using Chrome 26 I can call from jssip client but not to jssip client<br></div>Using Chrom 28 I can call both from and to jssip client<br>
</div>Using Firefox 23 I can't do anything!<br><br></div>is there anything I can do in freeswitch to improve compatibility? any experience?<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Aug 6, 2013 at 3:41 PM, Javier Menendez <span dir="ltr"><<a href="mailto:menendez.garcia@gmail.com" target="_blank">menendez.garcia@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">No luck :(<br><div><br>EXECUTE sofia/internal/<a href="mailto:asterisk@192.168.90.16" target="_blank">asterisk@192.168.90.16</a> export(nolocal:absolute_codec_string=PCMA,PCMU)<br>
2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) (REMOTE ONLY) [absolute_codec_string]=[PCMA,PCMU]<br>
EXECUTE sofia/internal/<a href="mailto:asterisk@192.168.90.16" target="_blank">asterisk@192.168.90.16</a> bridge({absolute_codec_string=PCMA}sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid;transport=ws;fs_nat=yes;fs_path=sip%3A4st0031l%40PUBLIC_IP%3A38333%3Btransport%3Dws)<br>
2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1176 sofia/internal/<a href="mailto:asterisk@192.168.90.16" target="_blank">asterisk@192.168.90.16</a> EXPORTING[export_vars] [absolute_codec_string]=[PCMA,PCMU] to event<br>
2013-08-06 15:37:47.312306 [DEBUG] switch_ivr_originate.c:2050 Parsing global variables<br>
2013-08-06 15:37:47.312306 [DEBUG] switch_event.c:1615 Parsing variable [absolute_codec_string]=[PCMA]<br>2013-08-06 15:37:47.312306 [NOTICE] switch_channel.c:1030 New Channel sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid [617a9f0e-fe9d-11e2-b1fd-b761e46dca72]<br>
2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:4420 (sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid) State Change CS_NEW -> CS_INIT<br>2013-08-06 15:37:47.312306 [DEBUG] switch_core_session.c:1341 Send signal sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid [BREAK]<br>
2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:416 (sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid) Running State Change CS_INIT<br>2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:455 (sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid) State INIT<br>
2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:87 sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid SOFIA INIT<br>2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro]<br>
2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:wUFqlClOJwLZaHzE+QlEQfVQEx979Gx4e7BCbl56]<br>2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1191 sip:4st0031l@PUBLIC_IP:38333;transport=ws Setting proxy route to sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid<br>
2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1220 Local SDP:<br>v=0<br>o=FreeSWITCH 1375775801 1375775802 IN IP4 PUBLIC_IP<div class="im"><br>s=FreeSWITCH<br>c=IN IP4 PUBLIC_IP<br>t=0 0<br></div>a=msid-semantic: WMS B54oKgHfL2pj5wroyvgO0c1ghIMukBaL<br>
m=audio 20466 RTP/SAVPF 8 101 13<br>a=rtpmap:101 telephone-event/8000<br>a=rtcp-mux<br>a=rtcp:20466 IN IP4 PUBLIC_IP<br>a=ssrc:3456367315 cname:w406axcLqrVlOdzc<br>a=ssrc:3456367315 msid:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL a0<br>
a=ssrc:3456367315 mslabel:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL<br>a=ssrc:3456367315 label:B54oKgHfL2pj5wroyvgO0c1ghIMukBaLa0<br>a=ice-ufrag:zGQpJHvQg7bqzb2V<br>a=ice-pwd:5SS6gUIwcuIQy7TX<br>a=candidate:3772579260 1 udp 659136 PUBLIC_IP 20466 typ host generation 0<br>
a=candidate:3772579260 2 udp 659136 PUBLIC_IP 20466 typ host generation 0<br>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro<br>a=ptime:20<br>a=sendrecv<br><br></div></div><div class="HOEnZb">
<div class="h5"><div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Aug 6, 2013 at 3:26 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">try prepending the bridge url with {absolute_codec_string=PCMA} </div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Tue, Aug 6, 2013 at 6:32 AM, Javier Menendez <span dir="ltr"><<a href="mailto:menendez.garcia@gmail.com" target="_blank">menendez.garcia@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><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>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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