Razvan, I got rtpproxy from <http://opensips.org/pub/rtpproxy/> http://opensips.org/pub/rtpproxy/ as you wrote.
I started it using such cli command “/usr/local/rtpproxy1/bin/rtpproxy -u opensips -l 1.1.1.1 -s /var/run/rtpproxy.sock -T 80 -i -n /var/run/timer.sock -d INFO” and made test call. Callee has been ringing during about 2 minutes and nothing happens at all. What I did wrong? P.S. I use such function in my script for rtp proxy “rtpproxy_offer("con");” From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Wednesday, January 12, 2011 3:14 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Pacth rtpproxy Hello Denis, In the official release of RTPProxy, the timeout parameter (-T) controls both session establishment and rtp timeout. This is a problem since we would like to have a long period for call establishment, but a fast media timeout detection. In the patched version of RTPProxy, the -W parameter allows you to specify a longer period for call establishment. If not set, it has the default value of 60 seconds. If you decide not to use patched version of RTPProxy, the timeout notification will work, but you will have the same timeout in both situations. Regards, Razvan On 01/12/2011 07:38 AM, Denis Putyato wrote: Hello Razvan, “OpenSIPS shouldn't even try to terminate the call because it isn't established yet” As I understand I just do not need to use –W key when starting rtpproxy, it does not work at all? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Tuesday, January 11, 2011 6:49 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Pacth rtpproxy Hello Denis, You are right, OpenSIPS shouldn't even try to terminate the call because it isn't established yet. I just added a small fix to solve this problem. Please update your code from svn to use this fix. The RTPProxy patch was done against commit "600c80493793bafd2d69427bc22fcb43faad98c5". You can either get the RTPProxy from git, change it's branch and then apply the patch, or you can download an already patched version from http://opensips.org/pub/rtpproxy/. Regards, Razvan On 1/11/2011 2:19 PM, Denis Putyato wrote: Hello! I try patch rtpproxy gotten from git. And there is such error during patching patch < rtpproxy_timeout_notification.patch patching file main.c Hunk #1 succeeded at 70 (offset 2 lines). Hunk #2 FAILED at 120. Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines). Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines). Hunk #5 succeeded at 276 (offset 4 lines). Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines). Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines). 1 out of 7 hunks FAILED -- saving rejects to file main.c.rej patching file rtpp_command.c Hunk #1 FAILED at 795. Hunk #2 FAILED at 888. 2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej patching file rtpp_defines.h Hunk #1 FAILED at 95. 1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej patching file rtpp_notify.c rtpproxy_timeout_notification.patch is a patch for timeout notification which divide rtp timeout and session initiation timeout notification as said in http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142 This patch I got from SVN version of latest Opensips. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Razvan Crainea www.voice-system.ro
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