[Rtcomm] [Rtcomm] [Bug 3824] SIP Calls keep ringing despite being picked up by other side

From: bugzilla-daemon at maemo.org bugzilla-daemon at maemo.org
Date: Tue Nov 25 17:36:11 EET 2008
https://bugs.maemo.org/show_bug.cgi?id=3824





------- Comment #31 from mikhail.zabaluev at nokia.com  2008-11-25 17:36 GMT+3 -------
(In reply to comment #23)
> Mikhail: Can you take a look at the Packet dump in comment 19 / comment 20? Is
> it useful?

Yes, it gives some clue. In the "non-working" call, the proxy is not responding
to the INVITE beyond the initial 100 Trying, so the request is cancelled by
timeout. Seeing as calls to other numbers work, I think it's a server-side
problem.
There is one odd thing about the client though: it does not update the Contact
header to reflect the NAT binding, despite proper signalling from the proxy.
Do you have discover-binding ("Discover public address" in the UI) enabled?
I'd like to see the registration sequence as well.


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