[Asterisk-Dev] RE: Re-invite Issue Revisited

[Gene Willingham]

Olle:

Sorry about not getting back to you sooner.  We have been chasing issues
with re-invite for a week.  After posting, I realized that the specific
error was related to 1.0.9, and did not want to waster your time.  Since
then we have been trying to determine why some re-invites fail while others
succeed.  Looking in the logs as you suggested we have discovered that the
cases where the re-invites fail appear to be an inconsistent NAT issue.
Basically with each re-invite the NAT device allocates a new RTP port.  We
see a message in the log:

Jan 12 10:09:09 DEBUG[27675] chan_sip.c: SIP response 200 to RE-invite on
outgoing call c42d78c2-cddcae9 at 192.168.0.5
Jan 12 10:09:09 DEBUG[27675] chan_sip.c: build_route: Contact hop:
<sip:7328766707-300 at 72.xx.xx.93:32768>
Jan 12 10:09:09 DEBUG[2630] rtp.c: Oooh, 'SIP/732xxxxxxx-300-4955' changed
end address to 72.xx.xxx.93:16422 (format 4)
Jan 12 10:09:09 DEBUG[2630] rtp.c: Oooh, 'SIP/732xxxxxxx-300-4955' was
72.xx.xxx.93:32937/(format 1309)
Jan 12 10:09:09 DEBUG[2630] chan_sip.c: Deferring reinvite on SIP
'2ae0431c082da8543987ce85180a7ddb at 4.xx.xx.60' - It's audio will be
redirected to IP 72.xx.xxx.93
Jan 12 10:09:09 DEBUG[27675] chan_sip.c: = Found Their Call ID:
2ae0431c082da8543987ce85180a7ddb at 4.xx.xx.60 Their Tag
SDpsfna99-e8f90004-0-634312553 Our tag: as278460ad
Jan 12 10:09:09 DEBUG[27675] chan_sip.c: Acked pending invite 103
Jan 12 10:09:09 DEBUG[27675] chan_sip.c: Stopping retransmission on
'2ae0431c082da8543987ce85180a7ddb at 4.xx.xx.60' of Request 103: Match Found
Jan 12 10:09:09 DEBUG[27675] chan_sip.c: SIP response 200 to RE-invite on
outgoing call 2ae0431c082da8543987ce85180a7ddb at 4.xx.xx.60

Apparently the far end nat is changing the port during the re-invite.  I am
not sure how well this is handle, I do know that I am experiencing a strange
one-way audio issue, where the outbound audio is not getting re-invited.
   

> 
> ------------------------------
> 
> Message: 6
> Date: Tue, 10 Jan 2006 16:38:08 +0100
> From: Olle E Johansson <oej at edvina.net>
> Subject: Re: [Asterisk-Dev] Re-invite Issue
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <43C3D4E0.903 at edvina.net>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> Gene Willingham wrote:
> >
> > We are experiencing some issues with re-invites.  The two issues I can
> > reproduce are:
> >
> >  one-way outbound audio after re-invite completes.
> >
> >  Not sending the BYE message to end-point that does not support re-
> invite.
> >
> > I checked the bug list and did not find anything, but I may have missed
> it.
> > I am not a developer so it is unlikely I can submit a patch, but I am
> > willing to work with anyone interested in taking a look at it.
> >
> Please open an issue report and make sure you attach a full SIP debug
> with SIP history enabled, verbosity and debug set to 4.
> 
> I'd like to see the details of this transaction with an endpoint that
> does not support the re-invite.
> /O
>