[Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION

Tue Aug 6 21:14:10 MSD 2013

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 at gmail.com>wrote:

> No luck :(
>
> EXECUTE sofia/internal/asterisk at 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 at 192.168.90.16bridge({absolute_codec_string=PCMA}sofia/webrtc/sip:4st0031l at 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 at 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 at vlsoojkf29mg.invalid[617a9f0e-fe9d-11e2-b1fd-b761e46dca72]
> 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:4420
> (sofia/webrtc/sip:4st0031l at 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 at vlsoojkf29mg.invalid [BREAK]
> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:416
> (sofia/webrtc/sip:4st0031l at 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 at vlsoojkf29mg.invalid) State INIT
> 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:87
> sofia/webrtc/sip:4st0031l at 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 at PUBLIC_IP:38333;transport=ws
> Setting proxy route to sofia/webrtc/sip:4st0031l at 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 at 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 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?
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
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>>
>> AIM: anthm
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>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>
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