[Freeswitch-users] Invalid codec CN is tearing down the call request

Ken Rice krice at freeswitch.org
Thu Dec 11 19:12:00 MSK 2014


CN is comfort noise... Your cisco is sending it, and freeswitch is passing
it thru because you have told it to use proxy media mode...


On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Your version is 3 months old. Can you reproduce this on the latest git head?
> 
> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>> Hello,
>> 
>> I am running FreeSWITCH Version
>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>> problem with some call requests setup.
>> Here is one of them:
>> 
>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>> sofia/outside_1/20049112223344 []
>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal
>> sofia/outside_1/20049112223344 [BREAK]
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>> (sofia/outside_1/20049112223344) State INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>> sofia/outside_1/20049112223344 SOFIA INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>> sofia/outside_1/20049112223344 Patched SDP
>>  ---
>>  v=0
>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>  s=SIP Call
>>  c=IN IP4 10.10.10.12
>>  t=0 0
>>  m=audio 24782 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.12
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>> 
>>  +++
>>  v=0
>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 19690 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>> 
>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>> 2014-09-17 23:11:20Z 64bit
>>  Local SDP:
>>  v=0
>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 19690 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>> 
>> sending out SDP exactly as shown above.
>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183 w/SDP
>>  
>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>  v=0
>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>  s=Dialogic-SIP
>>  c=IN IP4 10.10.10.198
>>  t=0 0
>>  m=audio 10024 RTP/AVP 19 18
>>  a=rtpmap:19 CN/8000
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=silenceSupp:off - - - -
>>  a=ptime:20
>> 
>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>> sofia/outside_1/20049112223344 Patched SDP
>>  ---
>>  v=0
>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 29492 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>> 
>>  +++
>>  v=0
>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 29492 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>> 
>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 Invalid codec CN!
>> 2014-12-11 14:29:12.214608 [ERR] switch_core_media.c:2294 Can't load codec?
>> 2014-12-11 14:29:12.214608 [NOTICE] switch_core_media.c:2295 Hangup
>> sofia/outside_1/20049112223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>> sofia/outside_1/20049112223344 [KILL]
>> 
>> and Freeswitch is terminating the call by using CANCEL.
>> 
>> My modules.conf.xml config part looks like following:
>>     <!-- Codec Interfaces -->
>>     <load module="mod_spandsp"/>
>>     <load module="mod_g723_1"/>
>>     <load module="mod_g729"/>
>>     <!--load module="mod_amr"/-->
>>     <!--<load module="mod_ilbc"/>-->
>>     <load module="mod_h26x"/>
>>     <!--load module="mod_vp8"/-->
>>     <!--load module="mod_b64"/-->
>>     <!--<load module="mod_siren"/>-->
>>     <!--<load module="mod_isac"/>-->
>>     <!--<load module="mod_celt"/>-->
>>     <!--load module="mod_opus"/-->
>> 
>> Is it a problem with CN definition?
>> Could you please lead me where the problem is located?
>> 
>> Thanks in advance.
>> Mac.
>> 
>> _________________________________________________________________________
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>> 
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> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
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> 
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-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
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irc.freenode.net #freeswitch
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