[Freeswitch-users] Possible BUG in G729 negotiation

Brian West brian at freeswitch.org
Tue Feb 26 21:43:45 MSK 2013


Inbound late negotiation and set the variable prior to answer.

--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
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Twitter: @FreeSWITCH_Wire
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On Feb 26, 2013, at 10:52 AM, Flavio Goncalves <flavio at sippulse.com> wrote:

> Hi Brian, 
> 
> This setting was already set. For outbound calls it works just fine. For inbound calls, I can't disable the annexb. I need to reply (200 Ok) with annexb=no.
> 
> Best regards,   
> 
> Flavio E. Goncalves
> CTO - SipPulse Routing and Billing Solutions for SIP
> 
> 2013/2/25 Brian West <brian at freeswitch.org>
> Try setting this variable:
> 
> sip_append_audio_sdp=a=fmtp:18 annexb=no
> 
> Can you verify that your provider is actually sending CN on 19 and NOT 13.  There are some Sonus switches that say 19 but mean 13 and send on 13.  19=13 in hex.
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> ISN: 410*543
> 
> 
> 
> 
> 
> 
> On Feb 25, 2013, at 8:08 AM, Flavio Goncalves <flavio at sippulse.com> wrote:
> 
> > Hi,
> >
> > I have searched the list, but I couldn't find a final answer for this case. I'm receiving calls from a provider using G729 annex B.
> >
> > INVITE
> > a=rtpmap:18 G729/8000.
> > a=fmtp:18 annexb=yes.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=rtpmap:19 CN/8000.
> >
> > and FS is answering
> >
> > 200OK
> > m=audio 31388 RTP/AVP 18 101 19.
> > a=rtpmap:18 G729/8000.
> > a=fmtp:18 annexb=yes.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=rtpmap:19 CN/8000.
> > a=ptime:20.
> >
> > If freeswitch  supports  only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes?
> >
> > The audio quality is really bad when I receive calls in this way. After removing the annexb on the gateway the audio is fine, but unfortunately I can't remove from it from all VoIP providers terminating calls. For outbound calls I simply removed the annexb, but for inbound there is no way to do it.
> >
> > Flavio
> >
> >
> > _________________________________________________________________________
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> >
> > 
> > 
> >
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> 
> _________________________________________________________________________
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> 
> 
> 
> 
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> 
> 
> 
> 
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