[SR-Users] Call routing

[mark at brightvoip.co.uk]

Hi all,

Posted a similar query a few weeks ago, without much interest - any advice appreciated.

I have two sites and will send calls between them.  I have Kamailio at each site which will route the calls out/in.

There are multiple distinct network routes between the sites, accessible via different IP addresses.  Each Kamailio has multiple IP's, one for each route.

The purpose of the multiple routes is mainly fault tolerance. Some of the network links are unreliable, so routing must adapt when route(s) are unavailable.  When all routes are available, all should handle some traffic, at differing ratios to match the bandwidth available to each route (e.g route A - 50%, route B - 30%, route C - 20%).

I know that the Dispatcher can manage the routing for the SIP traffic, with the %ge distribution, and with SIP OPTIONS 'pings' to detect route availability.

My main headache is that RTP must follow the same route as SIP for each call. 

After a bit of web digging, I was thinking of a solution where each of the Kamailio servers will run multiple instances of rtpproxy (one for each ip/route).  Then once the dispatcher has chosen a route for the call, to use the matching rtpproxy instance to direct the audio.

Any comments or alternate solutions/suggestions would be of interest.

Many thanks,
Mark



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