mod_conference wrong codec for auto_outcall b-leg
Hi everyone, I got a strange problem with mod_conference and auto_outcall taking the wrong codec for b-leg. For what I can see from the log everything seems fine. The a-leg calls with the following SDP: 2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP: v=0 o=- 3556961565 3556961565 IN IP4 10.20.30.125 s=pjmedia t=0 0 m=audio 4000 RTP/AVP 99 101 c=IN IP4 10.20.30.125 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 10.20.30.125 And the SILK/24000 codec is accepted according to the log file: 2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/1000004@10.20.30.240 SILK/24000 20 ms 480 samples 40000 bits 2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111 sofia/internal/1000004@10.20.30.240 Original read codec set to SILK:120 2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP sofia/internal/1000004@10.20.30.240: v=0 o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 21622 RTP/AVP 99 101 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv The conference get's created and activates the read/write codecs for the first member: 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@24000hz 1 channel 20ms 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms After that he's trying to do the auto_outcall and that's where things get weird: EXECUTE sofia/internal/1000004@freeswitch.aws.hosts.corpex.de conference_set_auto_outcall(sofia/internal/sip:1000003@10.20.30.108:63741;transport=UDP;ob) 2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel sofia/internal/sip:1000003@10.20.30.108:63741 [0236d005-1ff0-4f13-9370-bf7736087dd0] 2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP: v=0 o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 20318 RTP/AVP 98 0 8 101 13 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1; usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is configured like this: outbound_codec_prefs=silk@24000,silk@16000,silk@8000,PCMU,PCMA,GSM Thanks for your help, Oliver
maybe try to set http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix ? -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, September 18, 2012 at 9:22 PM, Oliver Saggau | Sysvision GmbH wrote:
Hi everyone, I got a strange problem with mod_conference and auto_outcall taking the wrong codec for b-leg. For what I can see from the log everything seems fine. The a-leg calls with the following SDP:
2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP: v=0 o=- 3556961565 3556961565 IN IP4 10.20.30.125 s=pjmedia t=0 0 m=audio 4000 RTP/AVP 99 101 c=IN IP4 10.20.30.125 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 10.20.30.125
And the SILK/24000 codec is accepted according to the log file: 2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/1000004@10.20.30.240 (mailto:1000004@10.20.30.240) SILK/24000 20 ms 480 samples 40000 bits 2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111 sofia/internal/1000004@10.20.30.240 (mailto:1000004@10.20.30.240) Original read codec set to SILK:120 2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP sofia/internal/1000004@10.20.30.240 (mailto:1000004@10.20.30.240): v=0 o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 21622 RTP/AVP 99 101 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
The conference get's created and activates the read/write codecs for the first member: 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@24000hz 1 channel 20ms 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms
After that he's trying to do the auto_outcall and that's where things get weird: EXECUTE sofia/internal/1000004@freeswitch.aws.hosts.corpex.de (mailto:1000004@freeswitch.aws.hosts.corpex.de) conference_set_auto_outcall(sofia/internal/sip:1000003@10.20.30.108 (mailto:1000003@10.20.30.108):63741;transport=UDP;ob) 2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel sofia/internal/sip:1000003@10.20.30.108 (mailto:1000003@10.20.30.108):63741 [0236d005-1ff0-4f13-9370-bf7736087dd0] 2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP: v=0 o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 20318 RTP/AVP 98 0 8 101 13 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1; usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms
Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is configured like this: outbound_codec_prefs=silk@24000,silk@16000,silk@8000,PCMU,PCMA,GSM
Thanks for your help, Oliver
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org (mailto:consulting@freeswitch.org) http://www.freeswitchsolutions.com
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Do you have some options/variables in mind I should try using auto_outcall_prefix? I tried setting codec_string=silk@24000 and absolute_codec_string=silk@24000, but still FS offers only SILK/8000 to the b-leg. Am 19.09.2012 01:35, schrieb Seven Du:
maybe try to set http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix ?
-- Seven Du Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
On Tuesday, September 18, 2012 at 9:22 PM, Oliver Saggau | Sysvision GmbH wrote:
Hi everyone, I got a strange problem with mod_conference and auto_outcall taking the wrong codec for b-leg. For what I can see from the log everything seems fine. The a-leg calls with the following SDP:
2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP: v=0 o=- 3556961565 3556961565 IN IP4 10.20.30.125 s=pjmedia t=0 0 m=audio 4000 RTP/AVP 99 101 c=IN IP4 10.20.30.125 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 10.20.30.125
And the SILK/24000 codec is accepted according to the log file: 2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/1000004@10.20.30.240 <mailto:1000004@10.20.30.240> SILK/24000 20 ms 480 samples 40000 bits 2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111 sofia/internal/1000004@10.20.30.240 <mailto:1000004@10.20.30.240> Original read codec set to SILK:120 2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP sofia/internal/1000004@10.20.30.240 <mailto:1000004@10.20.30.240>: v=0 o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 21622 RTP/AVP 99 101 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
The conference get's created and activates the read/write codecs for the first member: 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@24000hz 1 channel 20ms 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms
After that he's trying to do the auto_outcall and that's where things get weird: EXECUTE sofia/internal/1000004@freeswitch.aws.hosts.corpex.de <mailto:1000004@freeswitch.aws.hosts.corpex.de> conference_set_auto_outcall(sofia/internal/sip:1000003@10.20.30.108 <mailto:1000003@10.20.30.108>:63741;transport=UDP;ob) 2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel sofia/internal/sip:1000003@10.20.30.108 <mailto:1000003@10.20.30.108>:63741 [0236d005-1ff0-4f13-9370-bf7736087dd0] 2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP: v=0 o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 20318 RTP/AVP 98 0 8 101 13 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1; usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms
Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is configured like this: outbound_codec_prefs=silk@24000,silk@16000,silk@8000,PCMU,PCMA,GSM
Thanks for your help, Oliver
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Oliver Saggau | Sysvision GmbH
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Seven Du