Inline.
Gerardo Amaya wrote:
Hello All. After investigating quite a while about this issue I'm kind of desperate now, I will appreciate your help on this.
I have a SER Server with multiple interfaces, I'm able to handle all the SIP packets with the multi homed enabled. Now the part that I'm having real trouble with is the RTP audio streams. After struggling with MediaProxy I finally started using RTPProxy for this, since it has bridge mode available. Now the problem is that I have more than one network to connect audio from, so the internal and external configuration is not enough for me. So my questions are:
1.-Can I have RTPProxy Bridge mode working for all the interfaces in my server?
No, but you can "layer" ser where each instance is bridging it's own network.
2.-Can I have more than one RTPProxy socket available for each of the multiple interfaces I'm connecting in order to have the audio working, so each route will use the proxy of their own network?
The issue is to make sure that SDP is correctly replaced for INVITE and 200 OK dependent on the network origin and destination of the message. rtpproxy will only handle an internal and an external network. You also have the problem of nat_uac_test() testing for private addresses to detect NAT, so you need to replace those tests with IP-based tests to make sure you detect which network is src and dst. g-)
3.-If #2 is true, is this where the force_socket and force_rtp_proxy functions play a role in my configuration?
Please help!!
Thanks you so much in advance
Gerardo Amaya
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