[Freeswitch-users] Possible BUG in G729 negotiation

Michael Collins msc at freeswitch.org
Tue Feb 26 22:14:10 MSK 2013


Flavio,

Let me know how this goes. If you need a hand with the late negotiation
stuff give us a shout. It's all documented here:
http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requires_param.29

We also cover it in the FS Cookbook; last recipe in chapter 2.

-MC

On Tue, Feb 26, 2013 at 10:43 AM, Brian West <brian at freeswitch.org> wrote:

> Inbound late negotiation and set the variable prior to answer.
>
> --
> 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 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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> http://www.freeswitchsolutions.com
>
> 
> 
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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