Hello Sammy
No, I don't forget. My rtpproxy's startup script is "/usr/local/bin/rtpproxy -l 111.22.33.44/10.1.2.103 -s udp:10.1.2.103 19999 -F -m 36000 -M 36010 -d DBUG" I have forwarded the 36000~36010 port to rtpproxy server. Firewall ----------------------- Opensips/rtpproxy Server 111.22.33.44:5060 10.1.2.103:5050 111.22.33.44:36000/36010 10.1.2.103:36000/36010 But, Now sometime no audio. Do you have any suggest Thanks Nick From: [email protected] [mailto:[email protected]] On Behalf Of SamyGo Sent: Tuesday, September 11, 2012 10:28 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT Hi Nick, So you've forwarded the 5060 port to opensips server, cool. Did you forget to mention about the RTP ports forwarded to server ? Start RTPproxy with param -m and -M to assign range of RTP ports to use. Then forward that UDP port range to opensips server too. Then see what happens with audio. Also what is your total voip diagram with their network addresses. Thanks, Sammy On Sep 11, 2012 6:46 AM, "Nick Chang" <[email protected]> wrote: Hello Sammy My NAT type is Full cone NAT. In Firewall, I setting a IP & port 5060 to LAN IP. Example: Firewall ----------------------- Opensips Server 111.22.33.44:5060 10.1.2.103:5050 Thanks Nick From: [email protected] [mailto:[email protected]] On Behalf Of SamyGo Sent: Monday, September 10, 2012 7:32 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT Hi, Nick what is your NAT type? how is your network topology working !! BR Sammy P.S: The blog only takes care of configurations of RTPproxy with servers on Public IPs and clients behind NAT using Kamailio - This won't make much sense here. Also alot of things have evolved in kamailio since I wrote that post. On Mon, Sep 10, 2012 at 4:13 PM, Binan AL Halabi <[email protected]> wrote: Hi Nick, see this link posted by Qasim, http://saevolgo.blogspot.com/2012/03/making-rtpproxy-work.html I like it really, you can read it and adjust it as u need, //Binan --- On Mon, 9/10/12, Nick Chang <[email protected]> wrote: From: Nick Chang <[email protected]> Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT To: "'OpenSIPS users mailling list'" <[email protected]> Date: Monday, September 10, 2012, 2:16 AM Dear Binan Yes, It's my setting. route[nat_fixups] { #---- RTP Proxy handling ---# if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ #---- Activates the RTP Proxy for the CALLEE ---# rtpproxy_offer("cowf","202.55.233.194"); }; # catch and fix replies t_on_reply("2"); } onreply_route[2] { #xlog("incoming reply\n"); #---- Handling of the SDP for the 200 or 183 reply ----# #---- If behind nat (flags 5 or 6) start RTP Proxy ----# #---- Activates the RTP Proxy for the CALLER ----# if ( is_method("INVITE") && (isflagset(5) || isbflagset(6)) && has_body("application/sdp") ){ rtpproxy_answer("cowf","202.55.233.194"); } #---- If the CALLEE is behind NAT, fix the CONTACT HF ----# if (isbflagset(6)) { fix_nated_contact(); search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } exit; } Thanks Nick From: [email protected] [mailto:[email protected]] On Behalf Of Binan AL Halabi Sent: Monday, September 10, 2012 5:12 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT Hi Nick, and you are setting rtpproxy_offer() for Invite right ! //Binan --- On Mon, 9/10/12, Nick Chang <[email protected] <http://mc/[email protected]> > wrote: From: Nick Chang <[email protected] <http://mc/[email protected]> > Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT To: "'OpenSIPS users mailling list'" <[email protected] <http://mc/[email protected]> > Date: Monday, September 10, 2012, 2:02 AM Hi Binan But, My problem still it. It's not work for this setting. Or Do you have any suggest? Thanks Nick From: [email protected] <http://mc/[email protected]> [mailto:[email protected] <http://mc/[email protected]> ] On Behalf Of Binan AL Halabi Sent: Monday, September 10, 2012 4:57 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT Hi Nick, thank you too, //Binan --- On Mon, 9/10/12, Nick Chang <[email protected]> wrote: From: Nick Chang <[email protected]> Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT To: "'OpenSIPS users mailling list'" <[email protected]> Date: Monday, September 10, 2012, 1:46 AM Hello Binan force_rtp_proxy do not support opensips 1.8.1. It is old function. You can saw this page http://www.opensips.org/Resources/DocsMigration163to164 Now, I setting "rtpproxy_answer("cowf","202.55.233.194")" Thanks Nick From: [email protected] [mailto:[email protected]] On Behalf Of Binan AL Halabi Sent: Monday, September 10, 2012 4:37 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RTPPRoxy behind NAT Hi Nick, use force_rtp_proxy here is the link: http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id271191 //Binan --- On Mon, 9/10/12, Nick Chang <[email protected] <http://mc/[email protected]> > wrote: From: Nick Chang <[email protected] <http://mc/[email protected]> > Subject: [OpenSIPS-Users] RTPPRoxy behind NAT To: "'OpenSIPS users mailling list'" <[email protected] <http://mc/[email protected]> > Date: Monday, September 10, 2012, 1:01 AM Hello My RTPPRoxy and Opensips installed on the same server. Then It behind a NAT. It's my log. How to change "c=IN IP4 10.1.1.51" to "c=IN IP4 public IP" U 2012/09/10 15:57:22.704873 10.1.2.103:5060 -> 111.81.182.252:2151 INVITE sip:[email protected]:2151;line=d210bc79dfd2c52 SIP/2.0. Record-Route: <sip:202.55.233.194;lr;ftag=3554;did=72b.08b17766>. Via: SIP/2.0/UDP 202.55.233.194;branch=z9hG4bK6aff.e20d892.0. Via: SIP/2.0/UDP 10.1.1.51:5060;received=118.163.61.141;rport=40696;branch=z9hG4bK3339. From: <sip:[email protected]>;tag=3554. To: <sip:[email protected]>. Call-ID: 32313. CSeq: 20 INVITE. Contact: <sip:[email protected]:40696;nat=yes>. Content-Type: application/sdp. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Max-Forwards: 69. User-Agent: Linphone/3.5.2 (eXosip2/3.6.0). Subject: Phone call. Content-Length: 337. . v=0. o=886988000076 1752 1752 IN IP4 10.1.1.51. s=Talk. c=IN IP4 10.1.1.51. t=0 0. m=audio 7078 RTP/AVP 112 111 110 3 0 8 9 101. a=rtpmap:112 speex/32000. a=fmtp:112 vbr=on. a=rtpmap:111 speex/16000. a=fmtp:111 vbr=on. a=rtpmap:110 speex/8000. a=fmtp:110 vbr=on. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. 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