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@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@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@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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--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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

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

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http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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