[OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE

Iñaki Baz Castillo ibc at aliax.net
Fri Oct 23 12:25:57 CEST 2009


El Viernes, 23 de Octubre de 2009, Uwe Kastens escribió:
> Hi,
> 
> > IMHO a proxy shouldn't behave as a UAC. Perhaps it can monitor dialogs
> > and so because this features just requires requests inspection, there is
> > no intrusion (adding a Record-Route parameter is not intrusion XD).
> > But behaving as an UAC is 100% intrusion.
> >
> > Yes, OpenSIPS is very flexible and can be used to solve some UA problems,
> > but the proxy shouldn't be the key for this purpose (IMHO).
> 
> Ok. I am with you.
> 
> But for example looking at the problem with mediaproxy (see email from
> this morning), opensips is doing to much or to less ATM. So
> mediaproxy/opensips will talk to the wrong SDP Ports, since its using
> the 2nd 200 OK with SDP from the UAC answer.

Yes, I've replied to that mail right now. It seems to be a bug in mediaproxy.

PS: IMHO you should try to avoid those two 200 OK (INVITE) at the same time 
(even if it's correct).
Perhaps you could add a "Wait(1)" in top of the dialplan of the second 
Asterisk server so if there won't be a race between CANCEL and 200 (INVITE).

Or even better: why you send the INVITE to both Asterisk at the same time 
(parallel forking)? Is not enough for you to do load balancing and serial 
forking in case of failure)? (of course it could be non suitable in your 
case).


-- 
Iñaki Baz Castillo <ibc at aliax.net>



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