Does freeswitch try transcode g729 annexb=no to g729 annex?
Hello, I have an issue with codec negotiation, seems to me that freeswitch is trying to transcode g729 to g729 annexb=no for some reason, even when I have different codecs available? I'm bridging 2 calls, with profile settings: <param name="inbound-late-negotiation" value="true"/> <param name="disable-transcoding" value="true"/> and dialplan: <action application="set" data="inherit_codec=true"/> with global codec: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/> here is invite and 183 progress message: Session Initiation Protocol (INVITE) Request-Line: INVITE sip: SIP/2.0 Message Body Session Description Protocol Media Description, name and address (m): audio 20268 RTP/AVP 18 8 0 101 13 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): rtpmap:13 CN/8000 Media Attribute (a): ptime:20 Session Initiation Protocol (183) Status-Line: SIP/2.0 183 Session Progress Message Body Session Description Protocol Media Description, name and address (m): audio 59852 RTP/AVP 18 0 8 101 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv Media Attribute (a): maxptime:20 After that I get error message: 2016-07-13 11:29:11.136011 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2016-07-13 11:29:11.136011 [ERR] switch_core_io.c:1434 Codec G.729 encoder error! and freeswitch sends CANCEL to B-side leg. Maybe I have missed my settings, but if codecs "mismatch" shouldn't freeswitch choose g711 a/u and not try to "transcode" the codec? Best Regards, Vladislav
Its probably not even what you think it is, Its probably trying to generate ringback and failing. On Wed, Jul 13, 2016 at 7:35 AM, Vladislav Ivanov <deforceczt@gmail.com> wrote:
Hello,
I have an issue with codec negotiation, seems to me that freeswitch is trying to transcode g729 to g729 annexb=no for some reason, even when I have different codecs available?
I'm bridging 2 calls, with profile settings: <param name="inbound-late-negotiation" value="true"/> <param name="disable-transcoding" value="true"/> and dialplan: <action application="set" data="inherit_codec=true"/>
with global codec: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/>
here is invite and 183 progress message:
Session Initiation Protocol (INVITE) Request-Line: INVITE sip: SIP/2.0 Message Body Session Description Protocol Media Description, name and address (m): audio 20268 RTP/AVP 18 8 0 101 13 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): rtpmap:13 CN/8000 Media Attribute (a): ptime:20
Session Initiation Protocol (183) Status-Line: SIP/2.0 183 Session Progress Message Body Session Description Protocol Media Description, name and address (m): audio 59852 RTP/AVP 18 0 8 101 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv Media Attribute (a): maxptime:20
After that I get error message:
2016-07-13 11:29:11.136011 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2016-07-13 11:29:11.136011 [ERR] switch_core_io.c:1434 Codec G.729 encoder error!
and freeswitch sends CANCEL to B-side leg.
Maybe I have missed my settings, but if codecs "mismatch" shouldn't freeswitch choose g711 a/u and not try to "transcode" the codec?
Best Regards, Vladislav
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Does freeswitch generate his own ringback if I dont execute ringback? I just straight bridge the call after it hits dialplan. What can to fix this behavior? Switch to g711, or passthrough g729? 2016-07-13 16:19 GMT+03:00 Brian West <brian@freeswitch.org>:
Its probably not even what you think it is, Its probably trying to generate ringback and failing.
On Wed, Jul 13, 2016 at 7:35 AM, Vladislav Ivanov <deforceczt@gmail.com> wrote:
Hello,
I have an issue with codec negotiation, seems to me that freeswitch is trying to transcode g729 to g729 annexb=no for some reason, even when I have different codecs available?
I'm bridging 2 calls, with profile settings: <param name="inbound-late-negotiation" value="true"/> <param name="disable-transcoding" value="true"/> and dialplan: <action application="set" data="inherit_codec=true"/>
with global codec: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/>
here is invite and 183 progress message:
Session Initiation Protocol (INVITE) Request-Line: INVITE sip: SIP/2.0 Message Body Session Description Protocol Media Description, name and address (m): audio 20268 RTP/AVP 18 8 0 101 13 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): rtpmap:13 CN/8000 Media Attribute (a): ptime:20
Session Initiation Protocol (183) Status-Line: SIP/2.0 183 Session Progress Message Body Session Description Protocol Media Description, name and address (m): audio 59852 RTP/AVP 18 0 8 101 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv Media Attribute (a): maxptime:20
After that I get error message:
2016-07-13 11:29:11.136011 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2016-07-13 11:29:11.136011 [ERR] switch_core_io.c:1434 Codec G.729 encoder error!
and freeswitch sends CANCEL to B-side leg.
Maybe I have missed my settings, but if codecs "mismatch" shouldn't freeswitch choose g711 a/u and not try to "transcode" the codec?
Best Regards, Vladislav
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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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Brian West
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Vladislav Ivanov