J Thousand thanks to you mike, with this steps mentioned in the wiki was able to connect the cal. I guess i have to tune the same for the performance. The calls are having lot of delay, please advise me on the same. One more time thanks a lot mike (m...@ultra.net.br) you made my day.
From: mike.tesl...@ultra.net.br [mailto:mike.tesl...@ultra.net.br] On Behalf Of Mike Tesliuk Sent: Sunday, 29 September, 2013 2:21 AM To: rajesh.b...@goodcoresoft.com Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork the same for this Take a look over this howto http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPRO XY_-_English 2013/9/28 Mike Tesliuk <m...@ultra.net.br> Take a look over this howto http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPRO XY_-_English 2013/9/28 Mike Tesliuk <m...@ultra.net.br> ---- 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:101@204.16.1.50 <mailto:sip%3A101@204.16.1.50> ;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:102@75.74.203.73:60340;transport=UDP>. To: <sip:101@204.16.1.50 <mailto:sip%3A101@204.16.1.50> ;transport=UDP>. From: "102"<sip:102@204.16.1.50 <mailto:sip%3A102@204.16.1.50> ;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:102@204.16.1.50 <mailto:sip%3A102@204.16.1.50> SIP/2.0. Via: SIP/2.0/UDP 204.16.0.26:5062;branch=z9hG4bK1527256431. From: "Mike" <sip:101@204.16.1.50 <mailto:sip%3A101@204.16.1.50> >;tag=1050377705. To: <sip:102@204.16.1.50 <mailto:sip%3A102@204.16.1.50> >. Call-ID: 83821284@10.254.254.6. CSeq: 1 INVITE. Contact: <sip:101@204.16.0.26:5062>. 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 <rajesh.b...@goodcoresoft.com> Hi, I have attached the logs and my routing file @ http://pastebin.com/hu0bQGVw Please help me out in nailing this. Thanks Rajesh From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 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 <m...@ultra.net.br> 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 <rajesh.b...@goodcoresoft.com> 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: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 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: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 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: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 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 <rajesh.b...@goodcoresoft.com> 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: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 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 <rajesh.b...@goodcoresoft.com> 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 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 _______________________________________________ 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
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