RTPProxy does work behind NAT. It's mediaporxy that must be on a public ip.
Thanks for your help. Nick. On 3/19/13, Muhammad Shahzad <shaherya...@gmail.com> wrote: > If you are unfamiliar with rtp proxy and how it works, then it would be > better for you to use engage_rtp_proxy rather then offer / answer model. > Also RTP Proxy requires public IP address, its likely not to work on > private subnets (unless you have all SIP entities on same LAN, in which > case theoretically it should work but i have never tested it myself). > > Thank you. > > > On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis <sym...@gmail.com> wrote: > >> I am not sure if this is the correct place to post OpenSIPS+RTPProxy >> questions however, I tried to subscribing to the RTP proxy mailing >> list and never heard from them since. If it is ok to post RTP proxy >> related questions here.... I am trying to test OpenSIPS with RTP proxy >> with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS, >> RTPPoxy) just for testing. >> >> The code I am using is: >> >> route { >> force_rport(); >> } >> route[1] { >> if (is_method("INVITE")) { >> t_on_branch("1"); >> t_on_reply("1"); >> t_on_failure("1"); >> >> if (has_body("application/sdp")) rtpproxy_offer(); >> } >> else if (is_method("BYE|CANCEL")) { >> unforce_rtp_proxy(); >> } >> >> if (!t_relay()) { >> sl_reply_error(); >> }; >> exit; >> } >> onreply_route[1] { >> if (has_body("application/sdp")) rtpproxy_answer(); >> } >> >> >> There is no way audio using RTP proxy, but audio is fine between the >> UA without including the RTP proxy related script. Looking at the log >> I found that RTP is prefilling the callers address twice, but not the >> callees address. >> >> >> INFO:main: rtpproxy started, pid 7287 >> INFO:handle_command: new session >> ae450168-538e-e211-8550-001b7700a65b@oakville, tag >> d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong >> INFO:handle_command: new session on a port 35010 created, tag >> d23f0168-538e-e211-8550-001b7700a65b;1 >> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5062 >> INFO:handle_command: new session >> ae450168-538e-e211-8550-001b7700a65b@oakville, tag >> d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong >> INFO:handle_command: new session on a port 22982 created, tag >> d23f0168-538e-e211-8550-001b7700a65b;2 >> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5064 >> INFO:handle_delete: forcefully deleting session 1 on ports 35010/0 >> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 >> 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 35010/0 is cleaned up >> INFO:handle_delete: forcefully deleting session 2 on ports 22982/0 >> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 >> 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 22982/0 is cleaned up >> >> Is it possible to test RTP relaying with everything on the same network? >> >> Thanks in Advance, >> >> Nick. >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > -- > Mit freundlichen Grüßen > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_78...@hotmail.com > Email: shaherya...@googlemail.com > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users