10 Jan
2007
10 Jan
'07
1:55 p.m.
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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