[OpenSIPS-Users] Dispatcher Configuration
julio.gonzalez at cgi.com
Fri Oct 24 15:03:39 CEST 2008
Thanks for answering..
I do dispatching on the first message (it could be INVITE, MESSAGE,
etc.) and then record-route.
RTP does not flow over my AS.
The issues is when I sent the bye and no answer from original AS, then I
expect that the BYE goes to the back-up AS. At that time, RTP (the
entire session) is over.
From: Bogdan-Andrei Iancu [mailto:bogdan at voice-system.ro]
Sent: Freitag, 24. Oktober 2008 14:42
To: Gonzalez, Julio
Cc: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Dispatcher Configuration
Do you do dispatching for all requests (initial -like INVITE - and
sequential - like ACK, BYE) or only for initial ones and you do
Also, in step 2 , how is the call (RTP) moved on the backup AS? does it
take over the IP of the primary AS?
Gonzalez, Julio wrote:
> Hi All,
> I am using the dispatcher (in OpenSip 1.4.2) module to balance the
> load but I am struggling with the fail-over configuration. As far as I
> know, OpenSip is a transaction stateful server (not dialog statefull)
> so that It will not correlate the different transactions of one
> My problem is in case of failover when a dialog was already
> established. The scenario is as follows:
> 1. During the course of the call, the application server goes down
> 2. Application Server back-up takes over.
> 3. Call is not affected. RTP continues flowing.
> 4. User disconnects the call.
> 5. When the user disconnect the call, obviously the BYE will try the
> get the old route to the destination, but as this Application Server
> was "moved" to another IP Address
> 6. OpenSip receives a timeout.
> 7. OpenSip notes the problem, marks destination unavailable and
> selects the new destination from the list of proxies.
> 8. Message goes to destination.
> I looked at the list and there are a couple of examples but all of
> these are assuming the initial INVITE. Here, I already sent the INVITE
> and the call was successful established...
> How can I set up (or where in) the configuration file to capture this
> reply message?
> I tried to used the reply_route section, but I it is not allowed to
> put calls like dst_mark() ..
> Any help will be much appreciated.
> Julio Gonzalez-Saenz
> Users mailing list
> Users at lists.opensips.org
More information about the Users