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.