[Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION

Wed Aug 14 01:02:00 MSD 2013

JSSIP does not work with Firefox yet.

On Tue, Aug 6, 2013 at 12:14 PM, Javier Menendez
<menendez.garcia at gmail.com>wrote:

> 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
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
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>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> 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
>>> FreeSWITCH-users at lists.freeswitch.org
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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
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

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