[Spce-user] mismatch codecs in certain calls

[Lorenzo Mangani]

Joan,

The P-RTP-Stat is generated by either UA involved in the call and injected
in the BYE message with a breakdown of the RTP/RTCP statistics as seen from
the UA side - yours looks a lot like a Linksys and it does indeed suggest
some codec negotiation trouble between the devices. Please privately share
the full SDP negotiation maybe the fault will be more evident there,

Best,

lorenzo

On Thu, Feb 14, 2013 at 1:47 PM, Andrew Pogrebennyk <
apogrebennyk at sipwise.com> wrote:

> Hi Joan,
> no, the line you have posted is proprietary to some vendor and serves
> only information purpose I think.. The codec preferences are sent in the
> message body (lines below Content-Length in INVITE and 200 OK of your
> 'ngrep-sip' trace). You have to compare the lists of both parties, the
> first common codec will be used. The negotiation is transparent to
> sip:provider.
>
> On 02/14/2013 01:39 PM, Joan Cifre (ibred) wrote:
> > Hi all,
> > we are experiencing problems with certain calls. The call is stablished
> > but we speakers can't hear each other. This is happening just with
> > certain calls and with certain peerings, so I'm almost sure this is not
> > a NGCP problem. However, I think this is due to a bad codec negotiation.
> > I can see this line i certaing calls
> >
> > P-RTP-Stat:
> > PS=828,OS=16560,PR=446,OR=71360,PL=0,JI=0,LA=0,DU=9,EN=G729a,DE=G711a'
> >
> > Does it means that one server is sending streams with G729a codec and
> > the other one with G711a codec? Which one decides which codec use if
> > both are in the list?
> > Thanks for the support.
> > Joan
>
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-- 


Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering
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