Hello There,
I am new to RtpProxy and following is what I want to achieve.
1) SipUA(Xlite @ 10.30.10.107) registers to OpenSIPs registrar @
10.30.10.142 as s...@10.30.10.142
2) SipUA(Xlite @ 10.30.10.85) registers to OpenSIPs registrar @ 10.30.10.142
as s...@10.30.10.142
In a normal scenario - san1 callls san2 and the SIP call is established
between two with the RTP packets exchanged between 10.30.10.107 and
10.30.10.85 directly.
What I want to do - is divert the audio packets thru some RTP proxy running
on 10.30.10.142 along with OpenSips.
1) Is this possible?
2) I have attempted to configure this as follows
In the OpenSips.cfg - I have configured nathelper module as follows
modparam("nathelper", "rtpproxy_sock", "udp:10.30.10.142:8899")
modparam("nathelper", "force_socket", "10.30.10.142:8899")
I execute rtpproxy as follows
#rtpproxy -u root -l 10.30.10.142 udp:10.30.10.142:8899
I execute openSips and I can see that OpenSIPs is able to detect the
proxy and making the connections to it.
What I don't achieve is that in this setup - when I establish the SIP
Call, the audio is NOT routed through my rtp proxy but it's still
exchanged directly between the two
user agents.
Please let me know, if my attempt is valid in first place and yes,
some steps in the direction that I can proceed. My requirement is --
to route the SIP audio thru the RTP proxy,
be it a NON-NAT or NATT'd environment.
Any help in this regards is appreciated.
Thanks and Regards,
Santosh
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