[OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
Hello Everyone, I am having trouble getting the RTPProxy 1.2.0 to work with OpenISP 1.7. Starting the proxy using: ./rtpproxy -s udp:127.0.0.1:12221 -l public/private -p /var/run/rtpproxy.pid -u root root -F -d INFO LOG_LOCAL0 opensips.cf relevant pieces modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221") modparam("rtpproxy", "rtpproxy_autobridge", 1) modparam("rtpproxy", "rtpproxy_timeout", "0.5") modparam("rtpproxy", "rtpproxy_retr", 3) if (has_totag() && is_method("INVITE")) { engage_rtp_proxy("ie"); } if (is_method("ACK") && has_body("application/sdp")) { rtpproxy_answer(); } route[1] { xlog("Enter route 1"); if (has_body("application/sdp")) rtpproxy_answer(); if (is_method("INVITE")) { xlog("New Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]"); if (rtpproxy_offer()) { t_on_reply("1"); } else { t_on_reply("2"); } t_on_branch("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[1] { xlog("incoming reply\n"); if (has_body("application/sdp")) rtpproxy_answer(); exit; } onreply_route[2] { xlog("incoming reply\n"); if (has_body("application/sdp")) rtpproxy_offer(); exit; } When starting OpenSIPS everything looks fine: Dec 19 21:38:58 [3397] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3400] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3398] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3395] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3401] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3402] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Making a call howver, yields the following error (watch it work now ;): ERROR:rtpproxy:force_rtp_proxy_body: no available proxies /var/log/syslog Dec 19 21:28:18 opensips1 rtpproxy[3348]: INFO:main: rtpproxy started, pid 3348 Dec 19 21:28:31 opensips1 rtpproxy[3348]: INFO:handle_command: new session f679c215-bae17257-a0a8a5b8@192.168.2.11, tag C0847B09-9D90A2;1 requested, type strong Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:create_twinlistener: can't bind to the IPv4 port 50570: Cannot assign requested address Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:handle_command: can't create listener Dec 19 21:38:49 opensips1 rtpproxy[3379]: INFO:main: rtpproxy started, pid 3379 Dec 19 21:39:36 opensips1 rtpproxy[3379]: INFO:handle_command: new session 5537fa8e-69de2248-78eef465@192.168.2.11, tag A2311CB2-BB413627;1 requested, type strong Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:create_twinlistener: can't bind to the IPv4 port 57636: Cannot assign requested address Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:handle_command: can't create listener Your help is greatly appreciated, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
Hi Nick, are you sure the public and private IPs are local to the machine where you run rtpproxy ? as I see that rtpproxy complain on not being able to use the ips you configured via the "-l" param. Regards, Bogdan On 12/20/2011 04:45 AM, Nick Khamis wrote: Hello Everyone, I am having trouble getting the RTPProxy 1.2.0 to work with OpenISP 1.7. Starting the proxy using: ./rtpproxy -s udp:127.0.0.1:12221 -l public/private -p /var/run/rtpproxy.pid -u root root -F -d INFO LOG_LOCAL0 opensips.cf relevant pieces modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221") modparam("rtpproxy", "rtpproxy_autobridge", 1) modparam("rtpproxy", "rtpproxy_timeout", "0.5") modparam("rtpproxy", "rtpproxy_retr", 3) if (has_totag()&& is_method("INVITE")) { engage_rtp_proxy("ie"); } if (is_method("ACK")&& has_body("application/sdp")) { rtpproxy_answer(); } route[1] { xlog("Enter route 1"); if (has_body("application/sdp")) rtpproxy_answer(); if (is_method("INVITE")) { xlog("New Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]"); if (rtpproxy_offer()) { t_on_reply("1"); } else { t_on_reply("2"); } t_on_branch("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[1] { xlog("incoming reply\n"); if (has_body("application/sdp")) rtpproxy_answer(); exit; } onreply_route[2] { xlog("incoming reply\n"); if (has_body("application/sdp")) rtpproxy_offer(); exit; } When starting OpenSIPS everything looks fine: Dec 19 21:38:58 [3397] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3400] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3398] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3395] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3401] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Dec 19 21:38:58 [3402] INFO:rtpproxy:rtpp_test: rtp proxy found, support for it enabled Making a call howver, yields the following error (watch it work now ;): ERROR:rtpproxy:force_rtp_proxy_body: no available proxies /var/log/syslog Dec 19 21:28:18 opensips1 rtpproxy[3348]: INFO:main: rtpproxy started, pid 3348 Dec 19 21:28:31 opensips1 rtpproxy[3348]: INFO:handle_command: new session f679c215-bae17257-a0a8a5b8@192.168.2.11, tag C0847B09-9D90A2;1 requested, type strong Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:create_twinlistener: can't bind to the IPv4 port 50570: Cannot assign requested address Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:handle_command: can't create listener Dec 19 21:38:49 opensips1 rtpproxy[3379]: INFO:main: rtpproxy started, pid 3379 Dec 19 21:39:36 opensips1 rtpproxy[3379]: INFO:handle_command: new session 5537fa8e-69de2248-78eef465@192.168.2.11, tag A2311CB2-BB413627;1 requested, type strong Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:create_twinlistener: can't bind to the IPv4 port 57636: Cannot assign requested address Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:handle_command: can't create listener Your help is greatly appreciated, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
Hello Bogdan, Thank you so much for your response and time. I've been really struggling trying to get two way audio going. I think at some point I broke down in tears! This is a test virtual machine setup before migrating to the host servers. What I have is the following Router--->OpenSIPS/RTPProxy> Asterisk 1..n---> WAN ITSP Router 192.168.2.1 UC Polycom: 192.168.2.11 OpenSIPS: 192.168.2.102 Asterisk 1: 192.168.2.10 Asterisk 2: 192.168.2.11 My frist question is, do I need NAT and/or RTP Proxy since everything but the ITSP is on the same subnet? If not, I guess there is something wrong with my configuration. Eventually I will need NAT, and RTP, so I might as well try to get a grasp, and correct configuration using them. Can RTP proxy work when behind a router? Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
After resolving the incorrect rtp proxy -l flag, i'm still having a hard time getting two way audio behind a router. What I have in my config is the following: ./rtpproxy -s udp:127.0.0.1:7789 -l 192.168.2.102 -m 1 -M 2 -u root root -F -d INFO LOG_LOCAL0 modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7789") route{ if (is_method("INVITE") && has_totag()) engage_rtp_proxy("ie","192.168.2.102"); if (is_method("ACK") && has_body("application/sdp")) rtpproxy_answer(); } route[1] { if (is_method("INVITE")) { xlog("Start Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]"); if (has_body("application/sdp")) { if (rtpproxy_offer("ie","192.168.2.102")) t_on_reply("1"); else t_on_reply("2"); } t_on_branch("2"); t_on_failure("1"); } ... } onreply_route[1] { xlog("incoming reply\n"); if (has_body("application/sdp")) rtpproxy_answer(); exit; } onreply_route[2] { xlog("incoming reply\n"); if (has_body("application/sdp")) rtpproxy_offer(); exit; } I do have two way audio once every 30-50 calls, but cannot pin any combination that would keep things working. Outgoing audio always works. I really appreciate any help on this. Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users