sr-users

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37523 discussions

Re: [SR-Users] shm_malloc
by Daniel-Constantin Mierla 13 Jul '16

13 Jul '16

Hello, I don't expect to be very hard to extract the shm management code from kamailio. It is inside the subfolder mem/ . Then, there should be some dependecies that you need to get out from the core, coming in my mind is locking (mutexes). Maybe atomic operations (not remembering by heart if some counters are kept for memory management) and few other bits. It should work without crash, you just need to get it initialized as kamailio does it in it's main function. Cheers, Daniel On 12/07/16 13:57, Prakash wrote: > > Hello, > > > > Is it possible to take out shared memory code from kamailio? > > if so ,Will it work without any crash? > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Determining if request is from a local domain
by Shane Harrison 13 Jul '16

13 Jul '16

Hi all, >From previous posts my understanding is that RURI's are matched against reverse DNS lookup of listen addresses and aliases set via the "alias=" parameter in the config file (assuming the domain module isn't being used). However I am having difficulty in getting this to work. From looking at the code and the log file, it appears to me as if the alias isn't making it into the list of hosts that is checked against for some reason. Kamailio proxy is behind a loadbalancer (Kamailio dispatcher) and proxy is trying to relay the request because it doesn't recognise it as a local domain. Any thoughts on where my error is would be appreciated. Relevant CFG file section - note that RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT is 52.64.xxx.221 :5061 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ auto_aliases=yes # aliases and listening settings for sipproxy alias="aws-rcsstg.voip.systems" listen=tcp:10.4.1.62:5060 advertise RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT My request is a REGISTER as follows: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ received message REGISTER call-id b2ed2fdd(a)192.168.148.30 received from 10.4.1.62 raw msg REGISTER sip:aws-rcsstg.vvoip.systems;transport=tls SIP/2.0 Via: SIP/2.0/TCP 52.64.xxx.221:5060;branch=z9hG4bKa691.5fd57c2a783d3a5d7f6f80fcd0c815d6.0;i=d91 Via: SIP/2.0/TLS 192.168.148.30:5061;received=203.171.46.54;rport=64515;branch=z9hG4bK68b886fd00518;keep From: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls>;tag=4af7e7d8 To: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls> Call-ID: b2ed2fdd(a)192.168.148.30 CSeq: 1995005270 REGISTER Max-Forwards: 69 Contact: <sip:+6421608919@203.171.46.54:64515;transport=tls>;expires=240;+sip.instance="<urn:uuid:23624e3d-c98e-3b76-9acb-5499202e2f8d>";reg-id=1;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcs.sm";+g.oma.sip-im;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-is";+g.3gpp.cs-voice;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs";+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcse.sp";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";video User-Agent: FlowTalk-WindowsCommunicator_5.x P-Access-Network-Info: IEEE-802.3 Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO,MESSAGE,SUBSCRIBE,PUBLISH Allow-Events: refer, presence Supported: 100rel, replaces, timer, gruu, join, histinfo, path Date: Wed, 13 Jul 2016 00:27:39 GMT Content-Length: 0 Path: <sip:52.64.xxx.221:5060;transport=tcp;lr;received=sip:203.171.46.54:64515%3Btransport%3Dtls> Relevant section of the log where if( !uri == me ) is tested: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: ERROR: *** cfgtrace:request_route=[SIPOUT] c=[//etc/kamailio/kamailio.cfg] l=1052 a=16 n=if Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 24==9 && [aws-rcsstg.vvoip.systems] == [10.4.1.62] Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 5061) matches port 5061 Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:603]: grep_sock_info(): grep_sock_info - checking advertise if host==us: 24==13 && [aws-rcsstg.vvoip.systems] == [52.64.xxx.221] Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [forward.c:448]: check_self(): check_self: host != me Kind regards Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

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Detect 200 OK for ReINVITE
by Grant Bagdasarian 12 Jul '16

12 Jul '16

Hi, Is it possible in Kamailio, or SIP in general, to detect if a 200 OK is for a ReINVITE? Regards, Grant

4 5

Video conferencing with Kamailio
by Jay Li 12 Jul '16

12 Jul '16

Dear All, I'm curious if anybody has set up an infrastructure for video conferencing utilizing Kamailio as a proxy (like NAT support and so so). I found a kind of old tutorial "Run you own Skype-like service in less than one hour" kamailio:skype-like-service-in-less-than-one-hour [Asipto - SIP and VoIP Knowledge Base Site] written in 2013 on Kamailio 3.1. I wonder if this feature has been further developed in later releases. I know Kamailio is more often used as a SIP proxy, but somehow I got the impression that Kamailio supports WebRTC, so I wonder if it's possible to implement a many to many video conferencing infrastructure using Kamailio. If the answer is yes, how's the performance or the bottleneck I should pay more attention to (maybe bandwidth for video traffic, especially with NAT? ) Anyway I'm a newbie to both Kamailio and video conference, so any suggestion/discussion is appreciated. Thanks. Regards,Jay

2 3

destination ip address
by Péter Barabás 12 Jul '16

12 Jul '16

Hi All, i need to store some information in routing_logic script into database with useing SQLOPS module. I made a custom table and store such data like source and destination username, domain, ip address and port. It works fine for almost every parameter except destination IP address and port. I want to ask what is the correct way to determine the destination IP address and port of a user for a MESSAGE, INVITE or BYE message? The $Ri and $Rp is not the correct ones. Thanks. Dr. Péter Barabás

3 3

getting the local 408 response
by Jayesh Nambiar 12 Jul '16

12 Jul '16

Hi, I'm looking to catch the local 408 response generated by kamailio after the fr_timer strikes in. I was assuming to get it in the sl:local-request, but I dont see it in my log messages. I can catch responses like a 200 for register or a 487 for a message who's transaction is not yet created. Any proper way on how do I get to know when kamailio generates a local 408? I'm on the master branch (commit: 54562b873c2c3d0a04f0587e7706188a3eb19a74). Thanks, - Jayesh

4 6

Calculating time differences
by Sebastian Damm 11 Jul '16

11 Jul '16

Hi, we are suspending a transaction at one place in our route and resume it later. I want to log the time difference between suspending and resuming. I thought I could do it with saving $TV(Sn), but apparently I can't. Kamailio says, it can't convert the saved value into integer. This true, of course. I was searching for a transformation function converting this string into float, but there is none. Is there any way to calculate a time difference other than to live with seconds-only precision? Best Regards, Sebastian

2 1

Kamailio WEBRTC questions/ confusions
by Zaka 10 Jul '16

10 Jul '16

Dear List: I wonder if this has been asked before. Please ignore or point to the appropriate source, if needed. Simply, I have little experience with SIP servlets implementing WEBRTC/ Click2Call etc... Considering a Call Centre scenario offering a WEBRTC solution. Following are the confusions regarding requirements: 1: A Kamailio Box (running WEBSOCKET.config) 2: Customer will use web browser (websocket compatible of course) and enter address of the KAMAILIO Box or we shall need another KAMAILIO to statelessly forward the call to next available operator using Hunt Group or similar scheme? 3: If we don't plan to implement TLS/ MSRP (disabling these in the config file should be suffice?) 4: Last, but not the least, how will the users send request for login from browser? (assuming they have accounts created in the DB on Kamailio Box running WEBSOCKET supported KAMAILIO) With anticipatory thanks & regards, Zaka

3 6

Kamailio Enviroment
by Rodrigo Moreira 09 Jul '16

09 Jul '16

Hello, I would of your help installing Kamailio with the ability to authenticate to an HSS. Mainly want each IMS service to run on a different machine, ie, P / S / I-CSCF. I've tried several ways to set up, but without success. Does anyone have a step by step? I count on help. Hug. Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

1 0

re-INVITE on IP Change with Outbound
by Colin Morelli 08 Jul '16

08 Jul '16

Hey list, So I'm using the outbound module to ensure subsequent requests are delivered over an active TCP connection established by the mobile client. However, now I'm trying to add support for automatically responding to network changes (WiFi <-> LTE), and it's creating problems. Primarily, the issue is that when the device sends a re-INVITE for the call, its Route: header includes a flow-token corresponding to its *old* socket connection. Because the new request doesn't have the same source IP/port as the old connection did Kamailio thinks that the next route hop it needs to take is to pass the request over to the connection identified by the flow token. So it actually tries to re-open a TCP connection back out to the client even though I'm making the re-invite because that old transport is no longer valid. Does anyone have a recommended way of handling this case? I can add another header/param to the route header and manually remove it. Before loose_routing. Essentially, on in-dialog INVITE requests I want to ignore the outbound token on the originating side but I can't seem to find a way to do that without re-writing the header before loose_routing. Best, Colin

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