OpenSIPs RTPproxy integration

Question

I am trying to use RTP Proxy in OpenSIPs server to make a SIP call. I followed this guidelines to install RTP Proxy module in OpenSIPs server. http://voiprookie.blogspot.kr/2009/04/rtpproxy-12x-installation.html

I opened UDP port between 35000-65000, and checked if it is really working by using below command on SSH.

Tcpdump –n portrange 5060-65000

I used CSipSimple Android application to find out. I could see 5060 SIP connection but there was no connection between 35000-65000 port. I didn't even know if RTP proxy module is running or not.

How do I know that SIP call works under RTP proxy? Do I have to set something on CSipSimple Android application to make a call through RTP Proxy Server?

Thank you.

Answer 1

How do I know that SIP call works under RTP proxy?

In a typical setup where you have media bridged by a B2BUA (openSIPS + RTPproxy) IP:Port present in SDP Connection Data "C=" field of all requests sent by both clients will be rewritten using the the public IP where RTPproxy is sitting and Port provided by it. To make sure of that you can dump to a file outgoing SIP messages flowing through port 5060 of your SIP server and look for the public IP of RTP Proxy in their SDP part. Example:

INVITE sip:userB@ipB;user=phone SIP/2.0
... SIP HEADER FIELDS AND VALUES ...

v=0
o=J2S 3 1 IN IP4 99.99.99.99
s=session
c=IN IP4 <PUBLIC IP RTP PROXY HERE>
t=0 0
m=audio <PORT> RTP/AVP 112 ...
a=rtpmap:112 iLBC/8000/1
... MORE CODECS AND PARAMETERS

In case you have the original IP sent by the client there (in SDP of outgoing message sent from openSIPS to clients), then you have to check for some mistake in your setup. If RTP proxy public IP is there and you still have no audio, you should look further for connectivity issues (IPTables, etc).

Do I have to set something on CSipSimple Android application to make a call through RTP Proxy Server?

No. The proxy and ports negotiation are entirely up to B2BUA as described above. You can check details for RTP Proxy here.