sr-users December 2009

sr-users@lists.kamailio.org

67 participants
96 discussions

Re: [Kamailio-Users] Caller and Callee behind the same NAT

by juergen＠glowka.de

On Tuesday 29 December 2009 01:55:27 juergen(a)glowka.de wrote: > Hello, > > I'm new here. So I say "hello" to everybody. > > Maybe, somebody can help me or push me in the right direction. > > If there are Caller and Callee behind the same NAT, so it should > be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), > instead of using the rtp-proxy (this causes a jitter noise) . > > > I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. > > Has anybody a code snip or a sample kamailio.cfg for me? > I would be thankful for it. Better than giving you the solution, try to guest it ... Think a little ... what have in common a REQUEST from UAC A to UAC B if they are behing THE SAME NAT ROUTER ? -- Raúl Alexis Betancor Santana Dimensión Virtual ____________________________________________ Dear Mr. Santana Thanks for trying to help. oh, I guess, they've got the same IP-Address ;-) I found this code-snip in : dokuwiki/doku.php/examples:caller-callee-behind-same-nat ______________________________________ else if ( isflagset(2) and isflagset(3) ) { log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("i:450", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(2); resetflag(3); } } _______________________________________ But this Code-snip contains many mistakes so I changed it to : _______________________________________ if ( isflagset(5) and isbflagset(6) ) { lookup("location"); log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("$avp(i:450)", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(5); resetbflag(6); } } _______________________________________ This seems to work, but I still have got this looping, jittering feeping noise witch is getting louder and louder ... Two or three words I can talk, then it sounds, as if a group of Navajo-Indians gone attack me .... :-) Could it be the client? I use 2 PDAs, Windows Mobile, Portsip Ver. 2.6 Best Wishes, also for me Juergen Glowka

14 years, 10 months

[Serusers] using ser to involve nat

by WangZhenzhong

Hi every body I am using ser to solve the nat problem, In my environment: invite A-->SER-->B 1. SER receive the invite message REQA from A(invite B) 2. SER judge the use B, if the B is in public network, will fix nat in the REQA message if judge the B is in private network , will not fix anything and I want to know, how to judge the B's network, when the SER receive the INVITE from A? Is there any examle code used in ser.cfg thank you very much 2009-12-29 姓名：王振中公司：艾诺通信系统(苏州)有限公司地址：苏州工业园区苏虹西路81号苏虹工业坊A8单元二楼

14 years, 10 months

[Kamailio-Users] Caller and Callee behind the same NAT

by juergen＠glowka.de

Hello, I'm new here. So I say "hello" to everybody. Maybe, somebody can help me or push me in the right direction. If there are Caller and Callee behind the same NAT, so it should be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), instead of using the rtp-proxy (this causes a jitter noise) . I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. Has anybody a code snip or a sample kamailio.cfg for me? I would be thankful for it. Best wishes Juergen Glowka _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

14 years, 11 months

Re: [Kamailio-Users] adding inbound and outbound routes seperately

by Klaus Darilion

please always Cc: the list.... On 23.12.2009 16:24, Rodney wrote: > Well could you explain how to actually get peering functioning > between Kamailio on inbound and outbound? > There is no "peering functioning" in kamailio. You self have to define how you want to peer. For example if you want to peer with somebody sending you inbound requests, you have to define at least: - number format - authentication (IP-based, TLS, digest) One you have defined those parameters, you can implement it, e.g. digest authentication or IP based authentication. The same for outbound route - you have to configure Kamailio so that outbound requests are sent in the proper format. And Kamailio can not register to any other SIP service. If you need such a feature, you better use Asterisk. regards klaus > > -----Original Message----- From: Klaus Darilion > [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 23, > 2009 8:49 AM To: Rodney Cc: users(a)lists.kamailio.org Subject: Re: > [Kamailio-Users] adding inbound and outbound routes seperately > > > On 22.12.2009 22:35, Rodney wrote: >> Ok, I have kamailio and siremis, and I need to know how to set up >> a separate in bound and outbound route. >> >> Peering with inbound and registering with outbound >> > > Kamailio is a sip proxy, thus it can not register to other SIP > proxy. > > regards klaus > >> Any ideas guys? >> >> R >> >> >> >> _______________________________________________ Kamailio (OpenSER) >> - Users mailing list Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

14 years, 11 months

[Kamailio-Users] Newbie questions

by Antonio Goméz Soto

Hi, I have used asterisk before, but now am looking at kamailio for scaling up. I've read 'Building Telephony Systems with OpenSER', but I still have lots of questions.... so, here it goes: - I could not find a way to reboot a phone remotely, can it send SIP NOTIFY to all phones? - does it have something like the Asterisk Manager Interface, so I can connect/create an operator panel for it? Does that already exist somewhere? Something like FOP where operators can transfer calls, and check availability? - according to the docs it is not a BTBUA. What does this mean? I know that asterisk is a BTBUA, but what can a BTBUA do that a SIP proxy cannot? Is it only the moh and conferences? - lastly, is there a good tutorial on load-balancing somewhere? Thanks, Antonio.

14 years, 11 months

[Serusers] About iptel 400 missing cookie

by Dennis.Cheng

Hi folks, I used a Draytek Vigor 2820PBX to register iptel, and I found 400 missing cookie issue. When my pbx send the reinvite to hold another side, iptel always responded pbx 400 missing cookie. But I checked invite Via field, its branch included magic cookie "z9hG4bK". I attached wireshark packets. BR Dennis

14 years, 11 months

[Kamailio-Users] Merry Christmas and a Happy New Year!

by Elena-Ramona Modroiu

Hi everybody, before you leave the computers and go on holidays, I want to thank you all for the good work and support of the Kamailio and SIP Router projects. I wish you a Merry Christmas and a Happy New Year! As the project name is a Hawaiian word - the "Kamailian" greeting is: "Mele Kalikimaka me ka Hau'oli Makahiki Hou!" How would it be in your language? May the new year bring you joy, peace, good health and prosperity! Enjoy the winter holidays! Ramona

14 years, 11 months

[Kamailio-Users] Transparent bridge mode

by Vikram Ragukumar

Hello All, I am trying to setup a test scenario, where i have Kamailio and rtpproxy running on one CentOS box (Server1) and i have Asterisk running on another CentOS box (Server2). Server1 has 2 NIC's eth0 and eth1 that are both assigned Public IP's. There is a transparent bridge br0 connecting eth0 and eth1 which also has its own Public IP. Finally eth0 on Server2 also has a Public IP. Server2 must be assigned a Public IP. My goal is to modify rtpproxy so that i can intercept packets traveling to Server2, process them and let them resume along their original path. I would like to know if there is another way of setting this up so that i dont use as many Public IP's ? Do any of you see a problem with this setup, things that may not work eventually, or any other concerns ? Thanks, Vikram.

14 years, 11 months

[Kamailio-Users] Server Internal Error with sample config file : Kamailio 3.0 rc3

by Ricardo Martinez

Hello list. I'm using the sample configuration that comes with Kamailio RC3. (version: kamailio 3.0.0-rc3 (i386/linux)). I have two IP phones registered in the proxy. If I call from one to another and the call is cancelled a "500 Server Internal Error" message is send back to the caller (please see the debug attached) by the proxy. This is not happening with Kamailio 1.4 or Opensips 1.6 (using the sample configurations). Could be some problem? 10.0.0.220 - Caller 10.0.0.211 - Proxy 10.0.0.152 - Callee U 10.0.0.220:5060 -> 10.0.0.211:5060 INVITE sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==1". . U 10.0.0.211:5060 -> 10.0.0.152:5062 INVITE sip:5501122334@10.0.0.152:5062 SIP/2.0. Record-Route: <sip:10.0.0.211;lr=on>. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.220:5060 -> 10.0.0.211:5060 CANCEL sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Supported: gruu, path. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 CANCEL sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9831 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.211:5060 via_cnt==1". . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 ACK sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM). Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==2". . U 10.0.0.220:5060 -> 10.0.0.211:5060 ACK sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . Ricardo.-

14 years, 11 months

[Kamailio-Users] kamailio/ sip-router at 26c3

by Henning Westerholt

Hi, I'll (like in the last years) participate in the annual congress of the CCC in Berlin next week - the 26c3. More informations can be found on their site: http://events.ccc.de/congress/2009/wiki/Welcome So if you are in Berlin as well in this week (not necessarly on this event) and are interested in meeting for some discussions, please drop me a mail. Best regards, Henning

14 years, 11 months

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sr-users December 2009