Summary:ASTERISK-11033: chan_sip updates route set on re-invites, which is not allowed
Reporter:Matt King, M.A. Oxon. (kebl0155)Labels:
Date Opened:2007-12-13 18:17:05.000-0600Date Closed:2008-01-15 21:06:52.000-0600
Versions:Frequency of
Environment:Attachments:( 0) broken-reinvite.txt
( 1) reinvite-patch.txt
Description:We sometimes get calls disconnected when reinvite is enabled.

Our provider suggested a patch (attached).


We have calls coming in from and going out to the same provider.

We identified reinvite as a Good Thing in that it lets us drop out of the RTP stream.

All worked well until our provider got extra machines so more IP addresses were involved in the conversation - we started to get some calls drop when the reinvite kicked in.

Attached is a sample disconnected call (pre-patch), and the patch that fixed the problem.

I'm not a SIP expert, so I apologise if this is a dupe bug, or if the patch is not SIP compliant.  It does seem to work though, and the nice people on #asterisk suggested I post it.

Our provider does think the current implementation is broken - "it certainly appears that Asterisk has broken re-invites and ignores the record route when sending the ACK."

Hope this helps.
Comments:By: Ronald Chan (loloski) 2007-12-14 05:45:34.000-0600

reinvite-patch.txt patch appears to be missing?

By: Jared Smith (jsmith) 2007-12-14 09:07:11.000-0600


The reinvite-patch.txt file has been uploaded, but his license is pending, so it won't show up until his license has been approved.

By: Olle Johansson (oej) 2007-12-15 03:49:50.000-0600

There are related bug reports about not updating the route table after the initial call setup. This is a duplicate.

A very good catch though. Propably needs fixing in 1.4 too.

By: Matt King, M.A. Oxon. (kebl0155) 2007-12-15 08:22:31.000-0600

Yes I can confirm it's broken in 1.4 (which is what we run in production).

By: Joshua C. Colp (jcolp) 2008-01-15 21:06:51.000-0600

Fixed in 1.4 as of revision 98955 and trunk as of revision 98956.