sr-users March 2013

sr-users@lists.kamailio.org

92 participants
141 discussions

by Victor V. Kustov

Hi. Can someone tips for Kamailio + ABillS works? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

11 years, 7 months

Yet another IPv6 question related with rtpproxy ipv4/ipv6 bridging

by Miguel Baptista

Hello, I have been doing some tests with kamailio and IPv6. My initial setup was IPv6 only and now am I extending it to a dual-stack environment. Well, and now I am starting to face some (interesting) challenges. So, the first step in the dual-stack environment was to install RTP Proxy and configured kamailio to use it. With this setup, the UAs locally registered were able to communicate with each-other no matter with address family (IPv4/IPv6) they were using. So far so good. But now I want to extend my tests a bit more ... I want to communicate with the "outside world" (using ENUM and domain based SIP URIs). Do I have a way to know if the "destination" is IPv4 or IPv6? Because I need that information in order to properly bridge the calls on rtpproxy. I tried to use the onsend_route but it didn't work. I mean, I am able to know if the "next-hop" is IPv6 or IPv4 but it seems that it is "too late" to use rtpproxy. BTW ... I am assuming that if the "next-hop" is IPv6 then the final user agent will also be IPv6 (the same for IPv4) Should I use some other approach? For example, failure route instead? Does anyone have a similiar setup? How are you solving this issue? - Miguel Baptista

11 years, 7 months

SIP contact matching - location db ops

by Daniel-Constantin Mierla

Hello, apparently all the db operations to update/delete contacts are done by matching on username, contact, call-id and domain (when use_domain=1) -- it is like this for way many years, therefore needs some debate if/how to change. While the module has a matching_mode parameter for various contact matching algorithms, it is used for in-memory matching only. - http://kamailio.org/docs/modules/stable/modules/usrloc.html#matching-mode IIRC, for initial SIP specs (with no GRUU extensions, etc), aor contact address should be used for matching. Call-Id has a relevance only when CSeq is lower than previous received REGISTER with same call-id. Do I miss any specs requirements? Following the report on tracker FS#278, matching on path value for db ops will be probably very inefficient. Matching on call id for this case should be sufficient if the REGISTER comes from different UA. Can it happen that same UA sends same REGISTER via two paths? With this question I mean if anyone is aware of an UA able to do this. Relying on previous hop address to use in the db key, would not be enough, as the different path can be in the hops before. I plan to keep for now call-id for db operations when matching mode involves path, one way to fix it properly could be usage of ruid as db key, but needs further analysis. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

11 years, 7 months

Help with SIP over Websocket audio call

by Brad Johns

Hi, New to Kamailio. I have my Kamailio 4.0 server with websocket support, and the users can register using the JsSIP Tryit sample WebRTC application. However, after registration, the users can't place an audio call. I see no ringing on the remote browser. I don't know how to debug this further to find out what the problem is. Can anyone help with clues or debug? In Debug log I can see the websocket ws_frame.c decode the websocket message into SIP, and I see normal SIP call flow for an INVITE. However, nothing indicating a call. With this JsSIP, I can do chat through Kamailio SIP over WebSockets. With this Kamailio server, SIP User Agent Clients work just fine to register and place SIP call with audio. It's just that WebRTC audio calls don't work with JsSIP sample application with Kamailio 4.0 websocket module. Kamailio websocket configuration borrowed from: https://gist.github.com/jesusprubio/4066845 Any help debugging this appreciated. Brad

11 years, 7 months

Help with SIP over Websocket audio call: 405 Method Not Allowed

by Brad Johns

Hi, New to Kamailio. I have my Kamailio 4.0 server with websocket support, and the users can register using the JsSIP Tryit sample WebRTC application. They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over websockets module. However, after registration, the users can't place an audio call. I see no ringing on the remote browser. Can anyone help with clues or debug? In Debug log I can see the websocket ws_frame.c decode the websocket message into SIP, and I see normal SIP call flow for an INVITE. However, nothing indicating a call. I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm getting an error response to browser UA of "405: Method Not Allowed". I've isolated it down to the this snippet in the kamailio.cfg for route[LOCATION]: $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "TEST: Method Not Allowed"); exit; } } The switch case is returning -2, for some reason. Any help in debugging this appreciated.

11 years, 7 months

sems.conf

by Maaz Bin Mahmood

Hi there, I am trying to try SEMS but after make install, file sems.conf is not available in /usr/local/etc/sems/ Can you please suggest a solution? as tutorial here ( http://ftp.iptel.org/pub/sems/doc/current/howtostart_noproxy.html ) asks to change sems.conf as a first step. Regards, Max

11 years, 7 months

carrierroute module

by Angel Angelov

Hello, When I tried to start kamailio-4.0 with carrirroute module include, the follow error appeared 0(8732) ERROR: [sr_module.c:572]: ERROR: load_module: could not open module : /usr/local/kamailio-4.0/lib/kamailio/modules/carrierroute.so: undefined symbol: cfg_getnint . Do you need more information? Thanks in advanced. Regards. Angelov?

11 years, 7 months

IPv4 / v6 request to global parameter "TOS"

by Klaus Feichtinger

Hello list, in a bridging scenario with kamailio 3.3.4 and rtpproxy 1.2.1 for bridging signalling and media from an IPv4 to an IPv6 network and vice versa I found that the TOS value, which is set in kamailio.cfg, is used for IPv4 packets only. IPv6 packets have the traffic class value set to the default value 0x0. In other words: kamailio doesn´t use this variable for IPv6 packets. In the cookbook (http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tos) I haven´t found any hint that it _is_ limited to IPv4 only ("...for the sent IP packages"). I know that the name "tos" may be misleading, as the original definition was outdated by dscp+ecn, but it was/is working fine now. However, as IPv6 is using dscp+ecn, too, I wonder if the tos variable should support IPv6 packets, too. Could anybody give me a hint? Is there maybe an alternative way to prioritise SIP in IPv6 with kamailio? Thanks in advance, Klaus Feichtinger

11 years, 7 months

wiki accounts cleanup

by Daniel-Constantin Mierla

Hello, as there were many spamming accounts created in the wiki, used to add inappropriate content (http://www.kamailio.org/wiki/start?do=recent), I went through them and deleted those having more like random name and random id for email addresses. Hopefully I haven't deleted any valid account, but if you try yours and does not work, try to create it again. If you still have troubles, write me mentioning the account id. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

11 years, 7 months

Re: [SR-Users] Iimitations of xcap server in kamailio

by David Thomson

Thanks Peter! I'll try that out!! Is there a doc that goes through how to tweak Kamailio into production mode? What about having two Kamailio servers sharing one db and allowing calls between the two servers to be completed? That's another hurdle I've yet to get over. Ttyl, Dave ------Original Message------ From: Peter Dunkley To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Iimitations of xcap server in kamailio Sent: Mar 28, 2013 10:20 PM There are a number of core parameter and modparam values you need to increase from their defaults in order to be able to handle large HTTP requests (and therefore large XCAP documents). For example, * tcp_rd_buf_size (core parameter) * tcp_wq_max (core parameter) * sql_buffer_size (core parameter) * buf_size (xcap_server modparam) Regards, Peter > Has anyone else noticed issues when adding a large number of contacts to > your buddy list? I was using Jitsi and at about 40 contacts I was > encountering issues in that the subsequent contacts would not get added to > my buddy list. > > Ttyl, > Dave > > > > _______________________________________________ > 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 > -- Peter Dunkley Technical Director Crocodile RCS Ltd _______________________________________________ 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

11 years, 7 months

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sr-users March 2013