Why not just use the front end FS box in bypass media mode, then send the calls to the regional RTP proxies... The sip will act as a controller for those remote locations... Or Use OpenSIPs or Kamilio to handle the front end sip, its designed to use things like RTP Proxy FreeSwitch is not On 5/22/13 3:59 PM, "adahary" <adahary@gmail.com> wrote:
My plan is to use the FS as a centralized SIP proxy/registrar and several RTPproxy media server in different locations so users could get better latency when bridged over.
Yes, the RTPproxy is controled by some API commands (google around). I'll use the xml_curl with PHP to request from the RTPproxy ip/udp ports and then update A&B SDPs.
I'm just missing the last part of 'update A&B SDPs' ;
adahary
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