I enabled DBUG logging when starting rtpproxy: The logs output:
INFO:handle_delete:369832-3648463033-238704: forcefully deleting session 1 on ports 57012/35102 INFO:remove_session:369832-3648463033-238704: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped, 464 ignored INFO:remove_session:369832-3648463033-238704: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped, 4 ignored INFO:remove_session:369832-3648463033-238704: session on ports 57012/35102 is cleaned up INFO:remove_session:369832-3648463033-238704: RTP stream from callee: SSRC=NONE, ssrc_changes=0, psent=0, precvd=0, plost=0, pdups=0 INFO:remove_session:369832-3648463033-238704: RTP stream from caller: SSRC=NONE, ssrc_changes=0, psent=0, precvd=0, plost=0, pdups=0 464 packets ignored. Are those actual RTP packets? It’s kinda odd when it says 0 in from caller and callee. Is there a way to dig a little deeper and find out why it ignored those packets? From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Grant Bagdasarian Sent: Thursday, August 13, 2015 3:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <sr-users@lists.sip-router.org> Subject: Re: [SR-Users] need help with RTPProxy in bridged mode Hello, Yeah, I also noticed I forgot the / . Now the SDP c parameter is set correctly, but the audio from private to public isn’t relayed by rtpproxy. I ran a tcp dump on both interfaces (private and public), and it showed me RTP is being received from Freeswitch and also from our carrier, but nothing is passed between the two interfaces by rtpproxy. Any ideas? Below a slim version of my config: Request_route { if (is_method("INVITE")) { record_route(); if (has_body("application/sdp")) { rtpproxy_offer("ei"); } } } onreply_route[MANAGE_REPLY] { if (has_body("application/sdp")) { rtpproxy_answer("ie"); } } route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if(is_method("BYE")) { rtpproxy_manage(); } route(RELAY); exit; } } From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of SamyGo Sent: Thursday, August 13, 2015 3:26 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Subject: Re: [SR-Users] need help with RTPProxy in bridged mode Hi, Try starting rtpprpxy with a / in between the two IP addresses. For example -l 1.1.1.1/2.2.2.2<http://1.1.1.1/2.2.2.2> Besides that it depends where you are placing your rtpproxy function. BR, Sammy On Aug 13, 2015 8:36 AM, "Grant Bagdasarian" <g...@cm.nl<mailto:g...@cm.nl>> wrote: Hello, I’m using RTPproxy for the first time in bridged mode and I can’t get kamailio/rtpproxy to rewrite the c parameter to the correct public ip address of kamailio. The setup is as following: Carrier ------[fiber]------ Kamailio ---------[lan]--------- Freeswitch Kamailio is listening on two interfaces: 1) Private: 172.0.0.1 2) Public: 192.168.0.1 (since we have a dedicated fiber with our carrier, this is its public address) Freeswitch is listening on: 1) 172.0.0.2 Carrier is on: 1) 10.0.0.1 I’ve started an rtpproxy instance on the Kamailio box using: rtpproxy -s udp:127.0.0.1:7721<http://127.0.0.1:7721> -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.1 172.0.0.1 I’ve played around with rtpproxy_manage() and the various flags (ie, ei), but I can’t get kamailio to set the correct public IP when the 200 OK has to be sent back to the carrier. It always sets it to its private address, instead of its public address. I’m using Kamailio 4.2 with sippy/rtpproxy 2.0. Could someone please point me into the right direction? Thanks! Grant _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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