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
[email protected], 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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