sr-users January 2012

sr-users@lists.kamailio.org

72 participants
114 discussions

re-route SIP response message
by Robert R 18 Jan '12

18 Jan '12

Hi, Can I re-route a SIP response message? In other words, after the SIP proxy receives a 503 from node A, I need to change the TO field and re-route it to node B. I tried to modify $rd or $rU and didn't work. Thanks, R

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New feature: embedded MSRP relay
by Daniel-Constantin Mierla 18 Jan '12

18 Jan '12

Hello, a new module is now on development branch, named msrp, offering an embedded MSRP relay. Message Session Relay Protocol is specified by IETF ( core and relay part in RFC4975 and RFC4976), one of typical use cases is Instant Messaging sessions negotiated via INVITE-200ok-ACK, a different approach than SIP MESSAGE request. There is no external dependency, the transport layer from core is reused to manage TCP/TLS connections used for MSRP. A MSRP relay is pretty much an independent node, thus you can run Kamailio just to deal with MSRP traffic. But there is no problem to run Kamailio to handle SIP and MSRP traffic in the same instance, same or different sockets (it is working even when SIP/MSRP are sent over the same connection or different connections to same port). I couldn't find a reliable and trustable open source SIP phone to test with it so far (hints are welcome), the module was tested with network tools and chaining Kamailio instances. Help with testing and feedback is very appreciated -- I can provide guidelines to adjust the config file to fit tester's needs, just contact me off list or via IRC channel. For the moment, the relation (user,session) management is done in the config, using htable module for example (see the README), it may be added inside the module for the future to make it easier overall, although is less than 10 lines of config -- this is mainly for extra security reasons, to check if the session id matches the connection that was opened when the session was created, otherwise the user authentication functionality is done reusing existing functions from auth module. Another benefit is performances, storing local socket and connection IP/port saves some time to lookup the connection. Building MSRP relay on top of SIP server was done first for the benefit of reusing the transport layer from the core for IPv4/IPv6 and TCP/TLS, which is mature, scalable and offers asynchronous communication. Besides that, you have most of the config tools to route SIP requests available for routing MSRP (authentication, authorization, IP checking, accounting, a.s.o.). I made a news post with more details: * http://www.kamailio.org/w/2012/01/new-module-embedded-msrp-relay/ The readme of the new module is available at: * http://kamailio.org/docs/modules/devel/modules/msrp.html Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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sip-router and related projects at FOSDEM 2012 - social event
by Stefan Sayer 17 Jan '12

17 Jan '12

Hello everyone, as has become already a tradition, we would like to invite all VoIP people and everyone interested in sip-router and related projects (SER, Kamailio/OpenSER, OpenSIPS, SEMS etc) who is attending FOSDEM, Europe's biggest Free and Open Source conference, to a dinner on the evening of Saturday February 4th. This is the perfect opportunity to socialize among the developers, get to know the people behind email addresses and commits, do networking, find the people for the next project, and/or just hang out together with fine food and drinks. Participation at the event is free of charge, everybody pays for their own expenses. In order to do reservation in a restaurant please send me a short mail if you would like to attend. Please also indicate if you don't want to have your contact details communicated to the other social event attendees. If you are still wondering whether it's worth to go to FOSDEM - among all the other interesting talks this year in the telephony-devroom on Sunday, 5th, there will be a few directly related talks (the exact time may be subject to change): 11:30 Andreas Granig: From zero to VoIP provider in 15 minutes 12:55 Stefan Sayer: Session Border Control with SEMS 15:25 Daniel-Constantin Mierla: Secure SIP Communications with Kamailio 16:15 Saúl Ibarra Corretgé: SIP beyond VoIP http://fosdem.org/2012/schedule/track/telephony_and_communications_devroom Hope to see you all in Brussels! With Best Regards Stefan Sayer -- tel:+491621366449 sip:sayer@iptel.org mailto/xmpp:stefan.sayer@gmail.com

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new developer: Anca Vamanu
by Henning Westerholt 17 Jan '12

17 Jan '12

Hello all, I'm happy to announce a new developer for the project: Anca Vamanu. Welcome to the development team! Anca joined 1&1 last year and worked since then on several internal modules and also many important bug fixes for our kamailio server setup. She is a experienced C and C++ developer and contributed in the past a lot to OpenSER and other VoIP projects. One interesting work she just finished and that we also like to contribute to the community is a db_cassandra module, providing access to this NoSQL database [1] from kamailio. She will take over the maintenance of this module and also work on improving other parts of the server. Best regards, Henning [1] http://cassandra.apache.org/ -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

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Homer SIP Capture Setup Question
by JR Richardson 17 Jan '12

17 Jan '12

Hi All, I spent a few hours this afternoon setting up a Homer SIP Capture server today. The build went well, good example setup howto's. My setup is a single kamailio/sipcature/homer node sitting on a LAN with 2 NIC's, eth0 for network access ssh/web, eth1 is the capture port connected to switch mirrored port. On the switch I have a kamailio proxy server being monitored. All SIP traffic through the proxy server is logged to the homer server, but the homer server is receiving SIP packets on the capture interface and sending out SIP messages through the network access eth0 port. The messages are being processed from the on_reply route. The default config for the capture node is to capture all SIP packets, but this seems to create the problem. Here is the route and onreply_route default config: route { sip_capture(); exit; } onreply_route { sip_capture(); exit; } If I replace the onreply_route exit; with drop; then the capture server no longer sends SIP messages out the eth0 interface and everything seems to work as expected. So my question; what is the difference between a drop; and exit; within the on_reply route and how the hell is the capture server sending messages out the eth0 network interface when the server is 100% bound to eth1 capture interface? Any insight or guidance is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

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NoSQL Backends for Subscriber Database.
by David 16 Jan '12

16 Jan '12

I was wondering if any NoSQL adapters are available for Kamailio to do "subscriber authentication" backend? I was looking at Mongo or CouchDB as two possible backends Seems like these would be good candidates for Backends for Kamailio. Any ideas on this? Thanks David.

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Selecting a database for Kamailio, PostgreSQL or MySQL
by Spencer Thomason 16 Jan '12

16 Jan '12

Hello all, We are currently in the process of reworking our billing platform which uses Kamailio for routing in conjunction with a Django based web interface. I am curious if anyone has any feedback on which database engine they chose for their deployment and why. Obviously performance is a concern but data integrity is essential. Currently we have two Kamailio based clusters in two datacenters using MySQL in a dual master setup for each of the primary database nodes. I've been looking into PostgreSQL as a possible alternative and would like to hear any community thoughts. Thanks, Spencer

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Re: [SR-Users] sr-users Digest, Vol 80, Issue 49
by Javier Gallart 16 Jan '12

16 Jan '12

Hello Yaron can you try "mi mt_reload"? Regards Javi > > Message: 2 > Date: Mon, 16 Jan 2012 11:54:04 +0200 > From: Yaron Nachum <nachum.yaron(a)gmail.com> > Subject: [SR-Users] using XMLRPC module for fifo commands > To: sr-users(a)lists.sip-router.org > Message-ID: > <CADKX-JDqsxBb3T6XOsiUUs=Xj+idNAXEECBx1HeuAu8dSEdbsA(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi everyone, > Is it possible to use the xmlrpc module in order to send fifo commands to > the kamailio. for example I would like to use it in order to trigger mtree > table reload. > > I have already enabled the xmlrpc module and it responds. however I didn't > succeeded in sending the appropriate syntax in lrder to make the mtree > reload work. > > I tried using mtree.mt_reload method or fifo.mt_reload . > > Does anyone have a clue? > > Yaron. >

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using XMLRPC module for fifo commands
by Yaron Nachum 16 Jan '12

16 Jan '12

Hi everyone, Is it possible to use the xmlrpc module in order to send fifo commands to the kamailio. for example I would like to use it in order to trigger mtree table reload. I have already enabled the xmlrpc module and it responds. however I didn't succeeded in sending the appropriate syntax in lrder to make the mtree reload work. I tried using mtree.mt_reload method or fifo.mt_reload . Does anyone have a clue? Yaron.

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drouting - do_routing with $avp does not work
by Uri Shacked 16 Jan '12

16 Jan '12

Hi, this is my cfg: if(do_routing("0")){ $avp(tdr)=$avp(dr_attrs); } else { $avp(tdr)="1"; } subst_user('/(.*)/$avp(Ssrvindx)/'); xlog("L_CRIT","$C(rg) group is $avp(tdr)$C(xx)\n"); if(!do_routing("$avp(tdr)")){ xlog("L_CRIT","$C(rg)No TDR is found for $avp(DID) with Index $avp(Ssrvindx)$C(xx)\n"); t_reply("404", "Not found"); exit; } in the log, i get the following: 1(25799) CRITICAL: <script>: group is 8 1(25799) ERROR: drouting [drouting.c:706]: failed to get group id when i change the do_routing("$avp(tdr)") to do_routing("8"), everything works..... it looks like with the avp (although it is 8) it does not work. any ideas?

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sr-users January 2012