[Freeswitch-users] Different ptime causes call to fail

[Ben N]

Hi All

Wondering if someone could help me out with a ptime issue I'm having?

Some background info, we have a SIP client that uses G729 at 20ms for Wi-Fi,
and G729 at 50ms for 3G.  The 3G call is the one I'm having trouble with.
 When a call is made from the SIP client to the PSTN via our carrier when a
ptime of 50ms is used, the call fails.  The carrier does support G729 with
variable ptimes.  I also think that the audio stream sent back from the
carrier is G729 50ms, I have confirmed this by analyzing the RTP stream in
Wireshark.

In SIP terms, Freeswitch receives a 183 with SDP from the carrier which
gets passed to the SIP client.  Freeswitch then gets a 200 OK from the
carrier, but Freeswitch does not send an ACK to the carrier for this.  When
20ms is used, the ACK is sent.  This could just be a by-product of a
different problem, but I've yet to pin-point what that is.

In my vars.xml, I have both global and outbound codecs set to "=G729 at 50i,G729",
I've also tried them the other way around but I don't think this makes a
difference.

I have late negotiation set to false on my internal profile, and disable
transcoding set to true so that whatever is dynamically used on  the A-leg,
gets used on the B-leg.

I'm not using absolute codec strings for G729 at 50i, as I would think this
make things fairly static, and then affects Wi-Fi calls that are trying to
use 20ms.  Out of curiosity, i set the absolute codec string to 50ms (
{absolute_codec_string=G729 at 50i} within bridge application) and the same
thing happened.

Oh and also I have my codec negotiation to "generous".

In terms of the call failing, what happens is that the PSTN device rings,
the SIP client goes to a ringing state, but when the PSTN device is picked
up the SIP client does not progress to the next step.  I have noticed that
the signalling by this point is still stuck on sending an ACK to the
carrier, so signalling probably isn't working anyway.

Finally, I have mod_com_g729 loaded with a couple of licenses on the
server.  The idea is to not use a license for this kind of call, as both
legs are supposed to be using G729 at 50i.  I believe this is happening too
when making a call, because I can't see and encoder/decoder being created
in the FS console.

Has anyone had G729 with ptime of 50ms working in passthru mode when using
mod_com_g729?  I

My Freeswitch version is about a year old now, but I would like to avoid
updating if possible.  I can provide console/SIP logs if required!

I really am stuck on this one, so thanks in advance for any help!  If this
turns out to be beyond the scope of this mailing list, then Freeswitch
consultation would not be out of the question.
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