[OpenSIPS-Users] Query about RTP flow

[Nandini]

Dear Ali Pey,

Sorry for creating kind of mess in my question.
No i am not talking about any IP routing, But  my question is totally
related only to RTP packet flow between client -to- client. I mean when SIP
establihes a call session between two SIP clients (Which are registered to
OpenSIPS server), in that point of time How RTP packets (which carries
voice/video) flows from one client to another client ?

Here below link is my test bed set-up :

My_test_bed_set-up.png
<http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7589774/My_test_bed_set-up.png>  

And now My question is:
What is RTP session client-to-client ?
Is this RTP packets in voice/video call from caller will pass through router
1 ---> switch ----> router 2---> destination SIP client ? or just Caller to
Destination client ? (without pass through any nodes in between).

(If suppose i run media proxy server with OpenSIPs server then RTP packets
has to pass through all the nodes (routers and switch) in between to get
relay to other end client right ?)

So in this context if i run only openSIPS server without any Media-proxy
servers, how the RTP packets are achieving Client-to-client transfer ?

Please help me in clarifying this questions.

Regards,
Nandini




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