[Freeswitch-users] Does freeswitch try transcode g729 annexb=no to g729 annex?
Vladislav Ivanov
deforceczt at gmail.com
Wed Jul 13 17:25:34 MSD 2016
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 at 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 at 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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>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
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