[Freeswitch-users] Possible BUG in G729 negotiation

[Michael Collins]

Hey Flavio! Hope all is well with you (aside from this little SDP issue).
Looking forward to seeing you guys at ClueCon 2013. :)

Are you able to reproduce this issue on latest git HEAD? If so, could you
open a ticket at jira.freeswitch.org please? That will be the quickest way
to get it resolved.

Thanks!

-Michael

On Mon, Feb 25, 2013 at 6:08 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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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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