$rtp and &net.sip.addrecordroute set based on availabili

[vinylmike]

1. Brekeke Product Name and Version:
3.1.9.0 Advanced

2. Java version:
1.6.0_31

3. OS type and the version:
Red Hat Enterprise Linux Server release 5.3 (Tikanga) x86_64

4. UA (phone), gateway or other hardware/software involved:
AudioCodes Mediant 2000

5. Your problem:

We have two Brekeke SIP Servers (server1 & server2). The UAC sends traffic to server1 and server1 then forwards the traffic to server2 if a match occurs on the Request URI (least cost routing). Upon sending the INVITE to server2 we remove server1 from the signaling and media path by setting both $rtp and &net.sip.addrecordroute to "false". If successful, UAC is now communicating directly with server2. This part of the equation all works very well.

However, what if server2 is unavailable for some reason? What we're trying to do is use the $session failover plug-in to have server1 route the call to another, local provider, *but* we now need $rtp and &net.sip.addrecordroute set to "true" since the UAC does not have direct communications with the local provider. The UAC only has direct access to server1 & server2.

server1: 172.16.1.116
server2: 172.16.1.120
failover local provider: 10.3.1.33

Matching Patterns
$request = ^INVITE
To = sip:(\+120395|\+120391)(.+)@


Deploy Patterns
$rtp = false
&net.sip.addrecordroute = false
To = sip:%1%2@172.16.1.120:5060
$session = failover sip:%1%2@10.3.1.33:5060
&failover.timer.inviting = 5


What would be the best way to handle this?

[james]

It is an interesting challenge.

Let you try "$rtp=auto" instead of "$rtp=false" in the Deploy Patterns.
The SIP Server will relay RTP packets automatically if it detects NAT.


OR, put 3rd server between the server1 and the local provider.
This 3rd server needs to relay RTP packets to the local provider.