[Freeswitch-users] INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen

Danny Iland iland at cs.ucsb.edu
Thu Oct 17 22:36:34 MSD 2013


Giovanni,

I know the Skype API takes a TCP stream of SL 16. In order to force the
correct codecs, I inserted setting the following in the dialplan just
before the bridge.

    <action application="export"
data="nolocal:absolute_codec_string=L16,SL16,GSM"/>

However call establishment still fails with INCOMPATIBLE_DESTINATION after
comparing GSM and L16:

2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare
[GSM:3:8000:20:13200]/[L16:70:8000:20:128000]

Any suggestions on the contents of global_codec_prefs,
outbound_codec_prefs, or nolocal:absolute_codec_string? I suspect I need to
define something like L16 at 13200h but when I tried this, the L16 offering
did not change.

Full log snippet follows

Thanks,
Danny Iland

2013-10-17 18:22:25.252453 [NOTICE] mod_skypopen.c:2471 Channel
[skypopen/skype102/echo123] has been answered
2013-10-17 18:22:25.252453 [DEBUG] switch_channel.c:3594
(skypopen/skype102/echo123) Callstate Change RINGING -> ACTIVE
2013-10-17 18:22:25.252453 [DEBUG] mod_skypopen.c:1232
[543dc3c|22dd4bf] [DEBUG_SKYPE  1232 ][skype102       ][UP,INPROGRS]
MSG_ID=41
2013-10-17 18:22:25.252453 [DEBUG] mod_skypopen.c:2479
[543dc3c|22dd4bf] [DEBUG_SKYPE  2479 ][skype102       ][UP,INPROGRS]
outbound_channel_answered!
2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare
[GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]
2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare
[GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]
2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare
[GSM:3:8000:20:13200]/[L16:70:8000:20:128000]
2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5370 No 2833 in SDP.
 Disable 2833 dtmf and switch to INFO
2013-10-17 18:22:25.272510 [DEBUG] switch_core_session.c:858 Send signal
sofia/internal/[me]@[ip] [BREAK]
2013-10-17 18:22:25.272510 [NOTICE] switch_channel.c:3633 Hangup
sofia/internal//[me]@[ip]  [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
2013-10-17 18:22:25.272510 [DEBUG] switch_channel.c:3134 Send signal
sofia/internal//[me]@[ip]  [KILL]
2013-10-17 18:22:25.272510 [DEBUG] switch_core_session.c:1338 Send signal
sofia/internal//[me]@[ip]  [BREAK]
2013-10-17 18:22:25.272510 [DEBUG] switch_ivr_originate.c:3450
sofia/internal//[me]@[ip]  Media Establishment Failed.
2013-10-17 18:22:25.272510 [NOTICE] switch_ivr_originate.c:3452 Hangup
skypopen/skype102/echo123 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]



On Thu, Oct 17, 2013 at 1:00 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> mod_skypopen uses signed raw linear 16 bit (eg: SL16) natively, but you
> seem to force it to use PCMU or PCMA (eg: g711).
>
> FreeSWITCH will automatically do the conversion for you (from whatever
> your channel uses) when bridging to skypopen if you don't force it.
>
> -giovanni
>
>
> On Wed, Oct 16, 2013 at 8:04 PM, Danny Iland <iland at cs.ucsb.edu> wrote:
>
>> I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile
>> phones to VoIP providers.
>>
>> When bridging a call from a cellular handset to a mod_skypopen interface,
>> I get the following error as soon as the Skype call is answered. The
>> handset then hangs up. It appears the problem lies in the 'audio codec
>> compare' phase.
>>
>> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal
>> sofia/internal/IMSI[me]@[ip] [BREAK]
>> 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel
>> [skypopen/skype101/echo123] has been answered
>> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594
>> (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE
>> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232
>> [543dc3c|22dd4bf] [DEBUG_SKYPE  1232 ][skype101       ][UP,INPROGRS]
>> MSG_ID=41
>> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479
>> [543dc3c|22dd4bf] [DEBUG_SKYPE  2479 ][skype101       ][UP,INPROGRS]
>> outbound_channel_answered!
>> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172
>>  [543dc3c|22dd4bf] [DEBUG_SKYPE  1172 ][skype101       ][UP,INPROGRS]
>> ACCEPTED here you send me 32776
>> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177
>>  [543dc3c|22dd4bf] [DEBUG_SKYPE  1177 ][skype101       ][UP,INPROGRS] 4
>> SO_RCVBUF is 5120, size is 4
>> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181
>>  [543dc3c|22dd4bf] [DEBUG_SKYPE  1181 ][skype101       ][UP,INPROGRS] 4
>> SO_SNDBUF is 5120, size is 4
>> 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207
>> [543dc3c|22dd4bf] [DEBUG_SKYPE  207  ][skype101       ][UP,INPROGRS]
>> READING: |||ALTER CALL 32 SET_INPUT PORT="32776"|||
>> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]
>> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]
>> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP.
>>  Disable 2833 dtmf and switch to INFO
>> 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal
>> sofia/internal/IMSI[me]@[ip] [BREAK]
>> 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup
>> sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>
>> Any advice on forcing a transcode or otherwise having the Audio Codec
>> Compare result in a successful bridge is appreciated.
>>
>> Thanks,
>> Danny Iland
>>
>>
>>
>>
>> _________________________________________________________________________
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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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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