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@192.168.90.16 export(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@192.168.90.16 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)
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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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