On 2/7/13, Răzvan Crainea <[email protected]> wrote: > 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. >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
Hello Razan, I knew I should have posted more information, but did not want a really long email. Just to explain Nat box from hell <-------> OpenSIPS/RTPProxy <----> Asterisk (192.168.2.1) (192.168.2.105) (192.168.2.10) I did a trace on port 5060: The OpenSIPS Box: http://pastebin.com/p32rnBH6 The Asterisk Box: http://pastebin.com/HJ1SJjSU I really hope I was more proficient in reading these traces, but they all look fine to me! If you could point out something that does not look right, I would really appreciate it. Thanks in Advance, Nick. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
