[SR-Users] Routing calls

[mark at brightvoip.co.uk]

Hi,

I'm looking for pointers on the following scenario.

I have two sites (A and B) and sip traffic is sent from A to B.  At each site there are a number of SIP (Asterisk) servers and a Kamailio server.

The two sites have various network routes (X, Y & Z) between them.  I have been asked to load balance the inter-site SIP traffic across the available links with a 20/20/60 traffic split (dispatcher, carrierroute).

Each Kamailio server has multiple IP addresses and these are used to direct traffic to specific routes (Kamailio at site A IP 111.111.111.1 to Kamailio site B 111.111.111.2 defines route X for example).

For any given call, the complete call (SIP and both RTP streams) should travel wholly over one route.   

I would welcome any suggestions about how best to route the whole call (sip & rtp) over a specific route,

I was working on a solution with multiple RTPProxy instances, but was wondering if this is the best solution.

Thanks in advance.
Mark



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