[OpenSIPS-Users] OpenSIPS - RTP Proxy Integration (One Way Audio Debugging)

[Răzvan Crainea]

Hi, Nick!

 From what I see in your trace, the callee (Asterisk) is not sending 
anything to RTPProxy. Have you tried taking a trace on the asterisk 
ports to see if it is indeed sending anything?

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com


On 02/07/2013 02:20 AM, Nick Khamis wrote:
> Hello Everyone,
>
> It's been an on-again/off-again experience with OpenSIPS + RTP Proxy
> Integration. Throwing together bits and pieces of code found from
> various sources. The only absolute is one way (outgoing) audio. This
> is two weeks of testing, and I am asking for the advice of the
> experts. The basic flow of packets is intended to be:
>
> Router (192.168.2.1) ->  OpenSIPS/RTPProxy (192.168.2.105) ->  Asterisk
> (192.168.2.10) ->  Back to the Router (192.168.2.1)
>
> A little about the network:
> Port Forwarding ports (5060, and 8000-60000) to OpenSIPS (192.168.2.1)
> The OpenSIPS server is also in the DMZ for testing, hopefully I don't have to
> keep it as such when things are working.
> Not sure if it's related, I am using the Dlink DIR615 router, and ALG
> is checked. Unchecked, nothing works....
> The firewall on the router is turned off.
>
> For one call I have the following trace from RTP Proxy:
>
> INFO:main: rtpproxy started, pid 3565
> INFO:handle_command: new session
> a7c30ffb-8fb3bd0d-5f25720c at 192.168.2.11, tag 46A441DF-6FB2C1FE;1
> requested, type strong
> INFO:handle_command: new session on a port 8030 created, tag 46A441DF-6FB2C1FE;1
> INFO:handle_command: pre-filling caller's address with 192.168.2.11:10004
> INFO:handle_command: adding strong flag to existing session, new=1/0/0
> INFO:handle_command: lookup on ports 8030/18930, session timer restarted
> INFO:handle_command: pre-filling callee's address with 192.168.2.10:47686
> INFO:process_rtp: session timeout
> INFO:remove_session: RTP stats: 0 in from callee, 35 in from caller,
> 35 relayed, 0 dropped
> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: session on ports 8030/18930 is cleaned up
>
> tshark slice from OpenSIPS/RTPProxy:
>
> 103.009046 192.168.2.11 ->  192.168.2.105 UDP 214 Source port: 10004
> Destination port: 18930
> 103.009266 192.168.2.105 ->  192.168.2.10 UDP 214 Source port: 8030
> Destination port: 47686
>
> tshark slice from Asterisk:
>
> 102.939445 192.168.2.105 ->  192.168.2.10 UDP 214 Source port: 8030
> Destination port: 47686
> 102.939696 192.168.2.10 ->  199.47.127.10 UDP 214 Source port: 51758
> Destination port: 20680
>
> Taking OpenSIPS/RTPProxy out of the picture (i.e., only asterisk), I
> have two way audio. I hope this is enough info, and I can add related
> ngrep traces if needed.
>
>
> Your Help is Greatly Appreciated!!!!
>
> Nick.
>
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