[SR-Users] Routing calls

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