Take a look over this howto http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English
2013/9/28 Mike Tesliuk <[email protected]> > > ---- if i say something wrong please somebody correct my ---- > > > Hello Rajesh, > > You are using the nat_uac_test with parameter 23, this means parameters > 16, 4, 2, 1 , what means > > > - > > *1* - Contact header field is searched for occurrence of RFC1918 > addresses. > - > > *2* - the "received" test is used: address in Via is compared against > source IP address of signaling > - > > *4* - Top Most VIA is searched for occurrence of RFC1918 addresses > - > > *16* - test if the source port is different from the port in Via > > > > i dont know if you understand but this is a binary count, you can check in > this way > > 0010111 -> this is what you turn on > > in this case, if your package does not contains an Private ip address on > contact header, or does not contains a received on VIA different from the > ip address of the signalling, does not contais on VIA an private ip address > and the source port is not different from port on VIA , so your rule will > not match (just on match is enought) > > Look at this invite below (sended from a zoiper) > > 204.16.0.26:60340 -> 204.16.1.50:5060 > INVITE sip:[email protected];transport=UDP SIP/2.0. > Via: SIP/2.0/UDP 75.74.203.73:60340 > ;branch=z9hG4bK-d8754z-f6a3eadc786e7359-1---d8754z-;rport. > Max-Forwards: 70. > Contact: <sip:[email protected]:60340;transport=UDP>. > To: <sip:[email protected];transport=UDP>. > From: "102"<sip:[email protected];transport=UDP>;tag=489f8f45. > Call-ID: ZGNhYTQzNjIyOGFkYWNhOWQ3ZmQ2ZDVkYjhiNGI4MGE.. > CSeq: 1 INVITE. > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE. > Content-Type: application/sdp. > Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri. > User-Agent: Zoiper Communicator 2.04.10164 rev.10204. > Allow-Events: presence, kpml. > Content-Length: 352. > > > you can see the ip on signalling coming from 204.16.0.26 port 60340 > on via you have 75.74.203.73:60340, so you have a different ip address > from signalling or via , in this case you will set the NAT variable, but > check the invite below. > > # > U 204.16.0.26:5062 -> 204.16.1.50:5060 > INVITE sip:[email protected] SIP/2.0. > Via: SIP/2.0/UDP 204.16.0.26:5062;branch=z9hG4bK1527256431. > From: "Mike" <sip:[email protected]>;tag=1050377705. > To: <sip:[email protected]>. > Call-ID: [email protected]. > CSeq: 1 INVITE. > Contact: <sip:[email protected]:5062>. > Content-Type: application/sdp. > Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, > SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. > Max-Forwards: 70. > User-Agent: Yealink SIP-T20P 9.70.0.121. > Supported: replaces. > Allow-Events: talk,hold,conference,refer,check-sync. > Content-Length: 304 > > You have the same port on signalling and on VIA, in this case the rule > will no match and variable will not be set and this is a phone behind a nat > > > so, you should try to remove the if where you call the rtpproxy offer and > answer (just for test purpose) > > you should increment you debug info too > > /* uncomment the following lines to enable debugging */ > #debug=6 > #fork=no > #log_stderror=yes > > > > > > > > 2013/9/28 Rajesh Babu <[email protected]> > >> Hi,**** >> >> ** ** >> >> I have attached the logs and my routing file @ >> http://pastebin.com/hu0bQGVw**** >> >> ** ** >> >> Please help me out in nailing this.**** >> >> ** ** >> >> Thanks **** >> >> Rajesh**** >> >> ** ** >> >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *Mike Tesliuk >> *Sent:* Friday, 27 September, 2013 11:25 PM >> >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for >> different otuside netrwork**** >> >> ** ** >> >> If possible, paste your route file too**** >> >> ** ** >> >> 2013/9/27 Mike Tesliuk <[email protected]>**** >> >> start your opensips in debug mode, try to make the call, get all the >> message and paste in some pastebin website and show us the link**** >> >> ** ** >> >> 2013/9/27 Rajesh Babu <[email protected]>**** >> >> I am getting Error 483, too many Hops, There is no other error messages i >> am getting. Please some one help me out in this**** >> >> **** >> >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *Rajesh Babu >> *Sent:* Friday, 27 September, 2013 6:08 PM**** >> >> >> *To:* 'OpenSIPS users mailling list' >> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for >> different otuside netrwork**** >> >> **** >> >> HI Mike,**** >> >> **** >> >> Now the RTP is up and i am getting this message on my logs**** >> >> [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17800]: >> INFO:core:probe_max_sock_buff: using rcv buffer of 448 kb**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17810]: >> INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, >> support for it enabled**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17812]: >> INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, >> support for it enabled**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17815]: >> INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, >> support for it enabled**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17808]: >> INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, >> support for it enabled**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17811]: >> INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, >> support for it enabled**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]: >> INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, >> support for it enabled**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17814]: >> INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, >> support for it enabled**** >> >> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]: >> WARNING:drouting:dr_load_routing_info: table "dr_rules" is empty**** >> >> Sep 28 01:59:43 centos64 opensips: INFO:core:daemonize: pre-daemon >> process exiting with 0**** >> >> **** >> >> But my test tool is not connecting back my server. Is there any mistake i >> am doing.**** >> >> **** >> >> Thanks **** >> >> Rajesh**** >> >> **** >> >> *From:* [email protected] [ >> mailto:[email protected]<[email protected]>] >> *On Behalf Of *Rajesh Babu >> *Sent:* Friday, 27 September, 2013 2:34 PM**** >> >> >> *To:* 'OpenSIPS users mailling list' >> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for >> different otuside netrwork**** >> >> **** >> >> Hi Mike,**** >> >> **** >> >> This is log i am geting wheni try to start the service**** >> >> **** >> >> [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12839]: >> WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has >> been disabled temporarily**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: >> ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy >> Connection refused**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: >> ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not >> respond, disable it**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: >> WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: >> WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has >> been disabled temporarily**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: >> ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy >> Connection refused**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: >> ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not >> respond, disable it**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: >> WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy**** >> >> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: >> WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has >> been disabled temporarily**** >> >> Sep 27 22:29:03 centos64 opensips: INFO:core:daemonize: pre-daemon >> process exiting with 0**** >> >> **** >> >> *From:* [email protected] [ >> mailto:[email protected]<[email protected]>] >> *On Behalf Of *Mike Tesliuk >> *Sent:* Thursday, 26 September, 2013 10:25 PM**** >> >> >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for >> different otuside netrwork**** >> >> **** >> >> When you use the residential script almost all configuration come alredy >> working for this**** >> >> i have a tutorial (in portuguese ( i think that i should translate to >> english :) )) , where you can see a routing script working with nat >> >> http://opensips.com.br/wiki/index.php?title=Opensips_1.9**** >> >> You can take a look at modules documentation too >> >> http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html >> http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html**** >> >> There is on this maillist too a lot of discussions about this, below you >> can see one case >> >> http://opensips.org/pipermail/users/2011-January/016130.html**** >> >> If you get some information from an old version of opensips probably will >> be necessary to take a look on the module documentation to check about >> little diferences , but i think that this is the start point :)**** >> >> and if you is new to opensips i recommend to you the book about opensips >> ( >> http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book) >> **** >> >> **** >> >> **** >> >> 2013/9/26 Rajesh Babu <[email protected]>**** >> >> Hi Mike,**** >> >> **** >> >> Thanks for the response, I am totally new to this world, can you please >> help me by directing to on how to configure links. It will be great. **** >> >> Thanks in advance**** >> >> Regards**** >> >> Rajesh**** >> >> **** >> >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *Mike Tesliuk >> *Sent:* Thursday, 26 September, 2013 12:25 PM >> *To:* OpenSIPS users mailling list >> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for >> different otuside netrwork**** >> >> **** >> >> you should configure the nathelper and rtpproxy, this should help in you >> issue.**** >> >> **** >> >> 2013/9/26 Rajesh Babu <[email protected]>**** >> >> Hi,**** >> >> **** >> >> I am new to the OpenSIP world. I have installed a OpenSIP on my >> network. If i make a Call inside the network between two users i don’t have >> any issue, where as from outside the network, even though i can see the >> user registered in my server i am not able to call registered user (I see >> the user in my UL show listing). The call is established but i am not able >> to talk (Mean the audio and video are not getting transffered).**** >> >> **** >> >> Where as messages are going fine without any issue. I guess it is because >> message transmit over XMPP where calls on SIP right.**** >> >> **** >> >> **** >> >> I am really struck and i don’t know how to proceed, please help me out*** >> * >> >> **** >> >> **** >> >> **** >> >> Thanks**** >> >> Rajesh**** >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users**** >> >> **** >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users**** >> >> **** >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users**** >> >> ** ** >> >> ** ** >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >
_______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
