After trying and trying I found out it is a compatibility version with the browser. Using Chrome 26 I can call from jssip client but not to jssip client Using Chrom 28 I can call both from and to jssip client Using Firefox 23 I can't do anything! is there anything I can do in freeswitch to improve compatibility? any experience? On Tue, Aug 6, 2013 at 3:41 PM, Javier Menendez <menendez.garcia@gmail.com>wrote:
No luck :(
EXECUTE sofia/internal/asterisk@192.168.90.16export(nolocal:absolute_codec_string=PCMA,PCMU) 2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) (REMOTE ONLY) [absolute_codec_string]=[PCMA,PCMU] EXECUTE sofia/internal/asterisk@192.168.90.16bridge({absolute_codec_string=PCMA}sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid ;transport=ws;fs_nat=yes;fs_path=sip%3A4st0031l%40PUBLIC_IP%3A38333%3Btransport%3Dws) 2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1176 sofia/internal/ asterisk@192.168.90.16 EXPORTING[export_vars] [absolute_codec_string]=[PCMA,PCMU] to event 2013-08-06 15:37:47.312306 [DEBUG] switch_ivr_originate.c:2050 Parsing global variables 2013-08-06 15:37:47.312306 [DEBUG] switch_event.c:1615 Parsing variable [absolute_codec_string]=[PCMA] 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] 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:4420 (sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid) State Change CS_NEW -> CS_INIT 2013-08-06 15:37:47.312306 [DEBUG] switch_core_session.c:1341 Send signal sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid [BREAK] 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 2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:455 (sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid) State INIT 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:87 sofia/webrtc/sip:4st0031l@vlsoojkf29mg.invalid SOFIA INIT 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] 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] 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 2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1220 Local SDP: v=0 o=FreeSWITCH 1375775801 1375775802 IN IP4 PUBLIC_IP
s=FreeSWITCH c=IN IP4 PUBLIC_IP t=0 0 a=msid-semantic: WMS B54oKgHfL2pj5wroyvgO0c1ghIMukBaL m=audio 20466 RTP/SAVPF 8 101 13 a=rtpmap:101 telephone-event/8000 a=rtcp-mux a=rtcp:20466 IN IP4 PUBLIC_IP a=ssrc:3456367315 cname:w406axcLqrVlOdzc a=ssrc:3456367315 msid:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL a0 a=ssrc:3456367315 mslabel:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL a=ssrc:3456367315 label:B54oKgHfL2pj5wroyvgO0c1ghIMukBaLa0 a=ice-ufrag:zGQpJHvQg7bqzb2V a=ice-pwd:5SS6gUIwcuIQy7TX a=candidate:3772579260 1 udp 659136 PUBLIC_IP 20466 typ host generation 0 a=candidate:3772579260 2 udp 659136 PUBLIC_IP 20466 typ host generation 0 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro a=ptime:20 a=sendrecv
On Tue, Aug 6, 2013 at 3:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
try prepending the bridge url with {absolute_codec_string=PCMA}
On Tue, Aug 6, 2013 at 6:32 AM, Javier Menendez < menendez.garcia@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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