sr-users November 2014

sr-users@lists.kamailio.org

91 participants
129 discussions

Training/Tutorials?
by Jordan Rinke 19 Nov '14

19 Nov '14

For the life of me I can't find any in depth online resources on setting up and configuring Kamailio. Do anyone know of some good resources? Specifically I want to put this in front of a couple Asterisk servers so I need to learn how to connect a trunk to it, and then based on the call destination route it to the right Asterisk servers. I feel like this shouldn't be complicated I just don't understand where/how Kamailio is configured coming from only using Asterisk previously.

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rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE
by Juha Heinanen 19 Nov '14

19 Nov '14

when i make call from UDP/TLS/RTP/SAVP baresip to RTP/AVP sems, rtpengine gets called on initial invite/200 ok like this and audio works fine: Nov 18 02:46:06 rautu /usr/bin/sip-proxy[926]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 RTP/AVP trust-address) Nov 18 02:46:06 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 trust-address) however, when baresip makes re-invite, rtpengine gets called using the same offer/answer flags, but sems clears the call. on syslog, i see this: Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog INVITE <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com> Nov 18 02:46:59 rautu /usr/bin/sip-proxy[879]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 replace-session-connection replace-origin) Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50444] SRTP output wanted, but no crypto suite was negotiated Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog ACK <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com> Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50462] Discarded invalid SRTP packet: authentication failed Nov 18 02:46:59 rautu sems[20439]: [#b7193b70] [receive, AmRtpAudio.cpp:212] ERROR: decode() returned -4 Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_delete() Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: Routing in-dialog BYE <sip:jh-0x9a95b00@192.98.102.30:5066;transport=tcp> from <sip:jh@as.test.tutpro.com> to <sip:192.98.102.30:46718;transport=TCP> based on gruu Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50463] Error parsing RTCP header: invalid packet type and on baresip console this: dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80 dtls_srtp: incoming DTLS connect from 192.98.102.30:50524 dtls_srtp: verified sha-1 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) sip:jh@as.test.tutpro.com: session closed: Connection reset by peer could it be that at some point during the re-invite, sems gets srtp audio and therefore clears the call? if so, is it a bug in rtpengine? -- juha

2 2

reREGISTER problems: contact is inserted instead updated
by Klaus Darilion 18 Nov '14

18 Nov '14

Hi! Kamailio 4.1.6 with default settings for usrloc and registrar modul and db_mode=1 (write through). Fritzbox sends a reREGISTER and Kamailio tries to insert the contact into the database, but this fails as this contact for this AoR already exists. So I think there are 2 scenarios: Kamailio false INSERTs instead of UPDATEs the contact, or the contact was deleted from usrloc, but not from the database. Frankly I see the in the mysql logfile that immediately after the failed INSERT the record is deleted from DB. Does anybody have a pointer what might be the problem here? AFAIS the client always uses the same Contact and the same call-id. only Cseq increases. Thanks Klaus

1 1

rtpengine flags UDP/TLS/RTP/SAVP and UDP/TLS/RTP/SAVPF
by Juha Heinanen 18 Nov '14

18 Nov '14

rtpengine README lists these rtp protocol and profile flags: + RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF rtpengine source code, however, seems to know also these DTLS ones: daemon/call.c: .name = "UDP/TLS/RTP/SAVP", daemon/call.c: .name = "UDP/TLS/RTP/SAVPF", can they be used in rtpengine_offer/answer calls even when they are missing from README? -- juha

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Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
by Dan Christian Bogos 17 Nov '14

17 Nov '14

Hey Paul, Thanks for your feedback! In this case the call will come in via UDP (so it has 2 record route headers). I can still play tricks like reading the record-route headers but, I was in some way hoping that is the responsibility of loose_route() function to properly use the right protocol out of route headers since we are a proxy. Cheers, DanB On 14.11.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Message: 3 > Date: Thu, 13 Nov 2014 16:55:18 +0530 > From: Varghese Paul<varghesepaul87(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE > Message-ID: > <CAM33Ch4furD1FM-ZNKfbQ0nXNQd88BmzW73gjjb=cVyuC0K=8w(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Dan, > > You can check the transport value in the RURI and relay the SIP message as > tcp. > > if ($*rP* == "tcp"){ > t_relay_to_tcp() > } > > Regards > > Varghese Paul

3 4

Siremis Installation Issues
by Daniel Shoemaker 17 Nov '14

17 Nov '14

This is my first time posting to the mailing list. I'm currently running Kamailio 4.1, per the instructions on the website I got the tar bal for Siremis 4.1 s well. I have followed the install instructions at http://kb.asipto.com/siremis:install41x:main and I get all the way down to the bottom where I need to login to the web interface and when I try to use the default I get a very length openbiz error: [{"target":"ERROR","content":" \n[2014-11-17 11:51:15 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::dbConnection() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php:423<\/font> \n *function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/sysheader.inc 117\nfunction:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 423\nfunction:<\/b> getDBConnection ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 406\nfunction:<\/b> _run_search ( Array(2) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 345\nfunction:<\/b> fetchRecords ( \"[username]='admin' and status='1'\", Array(0), 1 ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 98\nfunction:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 66\nfunction:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/user\/form\/LoginForm.php 92\nfunction:<\/b> Login ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 310\nfunction:<\/b> invoke ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 110\nfunction:<\/b> dispatchRequest ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 32\nfunction:<\/b> include_once ( \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/bin\/controller.php 6<\/div>\n------------------------------Please ask system administrator for help...<\/div>\n"}]* My web server is configured to use apache and I've checked the ownership on the files and folder. I did a chmod 775 starting at the siremis-4.1.0/ directory. I did a search and found that the openbiz folder needs to be right under the siremis-4.1.0/ folder which it is. I have tried reinstalling several times only to get the same result. I'm sure I'm missing something simple I just need to know what it is. I'm currently running CentOS 7, Kernel 3.10.0-123.9.3, php 5.6.2-3, mariadb 10.0.14-1, & httpd 2.4.6-18. Dan --

1 0

Kamailio TCP Problems
by Andrew Mason 17 Nov '14

17 Nov '14

Hi All, Looking for some help with a TCP issue in Kamailio. I am running load testing through the proxy with a topology of UAC > Proxy > UAS. With UDP it works fine (using SIPp scripts) - when I turn to TCP as the transport i get the following error in the kamailio logs: ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect 192.168.0.19:5060 failed The IP address stated is the UAC which is sending an INVITE to the Proxy, which seems to send back a 100 TRYING to the UAC which is followed by repeated INVITE messages. It looks like the leg from the Proxy to the UAS is never established using TCP - however if I send the SIP messages to the Proxy via TCP and set the UAS to receieve in UDP Kamailio completes with TCP > Kamailio > UDP. When the UAC is set to send on TCP and UAS set to receive on TCP netstat shows the ports as listening. Can anyone please help?

1 0

Kamailio 4.2 TLS and csipsimple
by nadie 16 Nov '14

16 Nov '14

Hello. I installed K 4.2 with TLS. I can login users with csipsimple in TLS. but users can't call to other. in 5060 all correct. Thanks.

1 0

kamailio with pipelimit crashes when CPU load average >2
by Julia Boudniatsky 16 Nov '14

16 Nov '14

Hello, Sometimes kamailio with pipelimit crashes with segmentation fault, when CPU load average >2. The problem is not reproduced in the LAB (we used stress –c ). Please find attached core file. Thank you, Julia

1 0

401 vs 407 UAC auth issue
by Satish Patel 15 Nov '14

15 Nov '14

We have opensip configured with UAC auth to register SIP provide. I have configured two provider and both got registered but very interesting thing happened. Provider "A" sending me 407 challenge for authentication - Working Provider "B" sending me 401 for authentication - its failed to auth what is the solution here? does UAC module work with 401?

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sr-users November 2014