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

[From: bugzilla-daemon at maemo.org]

https://bugs.maemo.org/show_bug.cgi?id=3824


mikhail.zabaluev at nokia.com changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|NEW                         |RESOLVED
         Resolution|                            |WORKSFORME




------- Comment #7 from mikhail.zabaluev at nokia.com  2008-10-28 17:31 GMT+3 -------
Now I can see the problem. It goes away if you set the Outbound Proxy field in
the advanced SIP account options to "callcentric.com". The reason is, the
domain has several SRV entries for SIP, and our stack does SRV lookups if the
proxy hostname is not specified. It then uses the resolved transport addresses
interchangeably even within one dialog when sending ACKs and such, which does
not go well with the proxies: they apparently know nothing of each other's
dialogs.

NB: This is one more case where the "loose-routing" parameter matters. It's not
exposed in the Diablo UI, but it's possible to set it in GConf (the key name is
/apps/telepathy/mc/accounts/<account id>/param-loose-routing). Fremantle should
have it exposed. Still, there are issues with resending authentication requests
to alternating proxy addresses, which we have not fixed completely in
Fremantle. I'll file a Sofia-SIP request about this.


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