[OpenSIPS-Users] Dispatcher Configuration

Gonzalez, Julio julio.gonzalez at cgi.com
Fri Oct 24 15:03:39 CEST 2008

Hi Bodgan,

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.



-----Original Message-----
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

Hi Julio,

Do you do dispatching for all requests (initial -like INVITE - and 
sequential - like ACK, BYE) or only for initial ones and you do 
record-route ?

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
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> Users at lists.opensips.org
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