sr-users February 2017

sr-users@lists.kamailio.org

74 participants
99 discussions

by Grant Bagdasarian

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 -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

4 years, 7 months

rtpproxy does not response, disable it

by Zodiac

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

4 years, 10 months

Kamailio Behind TCP Load Balancer

by Colin Morelli

Hey all, I'm looking to put Kamailio behind a TCP load balancer that is SIP-unaware. My application is deployed in AWS and I'm tying to place Kamailio behind an ELB. For the most part, everything is fine. For my specific implementation I'm disabling UDP as a signaling transport and using only TLS. This enables me to not have to worry about a SIP-aware LB at the edge because replies to an incoming request will be sent over the existing established TCP socket (avoiding any crazy routing requirements). However - this poses an issue with source addresses. Does Kamailio support anything like the proxy protocol ( http://www.haproxy.org/download/1.5/doc/proxy-protocol.txt) for getting TCP stream information from a load balancer? Do I need to go back to exposing it directly to the world so that I can get source addresses? Thanks, Colin

6 years, 5 months

Kamailio Websocket Questions

by Nagorny, Dimitry

Hi List, two questions regarding Kamailio Websockets: 1. How do I send INVITE request TO the listening Websocket? I usually used for UDP/TCP/TLS these format, but somehow this is not working for Websocket because it gets in an endless loop: if ($rU=~"^(\+|00|0)?[1-9][0-9]{4,20}$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ";user=phone;transport=udp"; force_send_socket(udp:MY_EXTERN_IP:MY_EXTERN_PORT); rtpengine_offer("internal trust-address RTP AVP"); route(RELAY); exit; } 2. Why do I get these errors every time my Websocket Client is registering, but it still works and DB is inserted with registered location: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TCP 10.250.5.17:51283#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: 6uLXlD/aBrzu/j3PYP3DWO7rvLE=#015#012Server: kamailio (4.3.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> I am using Kamailio 4.3.5 + CentOS 7.2. I appreciate every help I can get. Best Regards Dimitry Nagorny Trainee

6 years, 6 months

Re: [SR-Users] How to determine correct port to set in record-route header

by Pranathi Venkatayogi

Any help? From: Pranathi Venkatayogi Sent: Monday, January 23, 2017 2:35 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: How to determine correct port to set in record-route header Hi, I am using Kamailio behind NAT, unable to figure how to make it put “public ip” in Record-route header, I am manually inserting the hard-coded header myself as below. However now I am having trouble choosing the right port number in all scenarios. $var(dstPort) = 5061; if (dst_port==5060) { $var(dstPort) = 5060; } insert_hf("Record-Route: <sip:MY_PUBLICIP_ADDR:$var(dstPort);transport=tls;lr;nat=yes>\r\n", "Record-Route"); In one scenario I see the conflicting port numbers in “via” header vs “Record route”. The ACK is being set to port 5060 based on Record route header and is not being received by the callee. 2017-01-23 14:52:49.970233 [blink.exe 2780]: SENDING: Packet 11, +0:00:36.411867 10.0.27.108:58217 -(SIP over TLS)-> 172.31.211.31:5061 SIP/2.0 180 Ringing Via: SIP/2.0/TLS 63.149.103.72:5061;received=172.31.211.31;branch=z9hG4bK4799.7067bd2a48063748a4c353fa408eaefa.0 Via: SIP/2.0/UDP 10.0.16.52;rport=5060;branch=z9hG4bK4799.935a27c6ef5ad112225964cbe7c1be44.0;i=2 Via: SIP/2.0/TLS 10.11.200.12:51793;rport=51793;received=10.11.200.12;branch=z9hG4bKPj49f6aeca3f9b4155ab4c5304b544aa4d;alias Record-Route: <sip:63.149.103.72:5060;transport=tls;lr;nat=yes> Record-Route: <sip:10.0.16.52:5061;transport=tls;lr;nat=yes> Call-ID: 09ce10efa6a946bf9445ccc21857393e From: "cust1" <sip:cust1@devtranslation.sms-test.cyracom.com>;tag=95082caabecc42548c2fec5ccd29e5de To: <sip:spanish@translation.sms-test.cyracom.com>;tag=ec365cc8489b48a7bc5725d21b7d97a1 CSeq: 17422 INVITE Server: Blink 3.0.0 (Windows) Contact: <sip:20745891@10.0.27.108:58216;transport=tls> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Content-Length: 0 Questions: 1. How can I let in-built record-route function automatically choose “public ip” so I can get rid of my manual insertion altogether. 2. If no option, what is the right way to choose the port on which packet is being received, so it is same as what is on “VIA”. 3. Any other pointers to improve the logic here?

6 years, 7 months

rtjson + uac + event - OK, ACK and BYE display name not updated

by Diego Nadares

Hi Guys, We are modifying "from" and "to" headers with rtjson module via a json struct. It works fine in INVITE, TRYING, RINGING and OK but not in the following messages. Kamailio It's changing ONLY "from" and "to" URI BUT NOT the DISPLAY NAME. I think it's the tm module but I can't figure out how to confirm this. Maybe I need to configure something else in my cfg or could it be a bug? My json struct that is handled by rtjson module response.routing = "serial"; //response.routing = "parallel"; response.routes = []; response.routes[0] = {}; response.routes[0].uri = "sip:1111111111@172.16.213.31:5060"; response.routes[0].headers = { // Headers to be modified "from": { "display": "1111111111", "uri": "sip:111111111@172.16.213.38" }, "to": { "display": "2222222222", "uri": "sip:2222222222@172.16.213.38" }, }; response.routes[0].headers.extra = {};//"X-Hdr-A: abc\r\nX-Hdr-B: bcd\r\n"; This is part of my cfg .... modparam("dialog","dlg_flag", FLD_START) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) .... # account only INVITEs if (is_method("INVITE") && !has_totag()) { setflag(FLT_ACC); # do accounting * setflag(FLD_START); # do accounting* setflag(FLT_ACCMISSED); setflag(FLT_ACCFAILED); } ..... >From source/Aleg: *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* To dest/Bleg: *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Incorrect display name in BYE and ACK: *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* interface: any filter: ( port 5060 ) and (ip or ip6) # U 2016/12/06 15:52:13.638731 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:5555555555@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: "SIPP" <sip:sipp@172.16.213.21:5060>. Max-Forwards: 70. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Remote-Party-ID: <sip:sipp@172.16.213.21 >:5060;party=calling;id-type=subscriber;privacy=off;screen=no>. Content-Length: 549. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.734159 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. . ................ # U 2016/12/06 15:52:13.878600 172.16.213.38:5060 -> 172.16.213.31:5060 INVITE sip:1111111111@172.16.213.31:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Max-Forwards: 69. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Content-Length: 788. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40236 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=sendrecv. a=rtcp:40237. a=ice-ufrag:QNCzXhUr. a=ice-pwd:Ci90BRpdZH51Hx5BaPgGMrGCeh. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40236 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40237 typ host. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.879441 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. ................ # U 2016/12/06 15:52:13.936565 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 180 Ringing. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 316. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:20.882165 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 213. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ................ # U 2016/12/06 15:52:21.173798 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 380. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:21.174390 172.16.213.21:5060 -> 172.16.213.38:5060 ACK sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-4. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:21.248172 172.16.213.38:5060 -> 172.16.213.31:5060 ACK sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.98116b5795d4c30f52156187a72059c8.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.178826 172.16.213.21:5060 -> 172.16.213.38:5060 BYE sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.468639 172.16.213.38:5060 -> 172.16.213.31:5060 BYE sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.469153 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.519423 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Content-Length: 0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . ................ exit 13 received, 0 dropped Sorry for my english. Thanks in advance! Diego.

6 years, 9 months

pua.publish using RPC

by Alex Balashov

Hello, Is pua.publish strictly an MI function, or is it possible nowadays to call it via the RPC channel, and specifically, using jsonrpc_exec()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

6 years, 11 months

fix_nated_contact and IPv6

by Sebastian Damm

Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: <sip:bob@192.168.8.132> outbound Contact: <sip:bob@1234:1234:0:1234:0:0:0:2> The correct way of transforming the Contact URI would be: Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]> This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian

7 years

NatHelper ignoring force_socket module parameter

by Asgaroth

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

7 years

Kamailio IMS deployment

by Ramya Y

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

7 years

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sr-users February 2017