[Freeswitch-users] Possible BUG in G729 negotiation

Flavio Goncalves flavio at sippulse.com
Tue Feb 26 19:52:48 MSK 2013


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
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>
>
>
>
>
>
> 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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