sr-users November 2008

sr-users@lists.kamailio.org

97 participants
171 discussions

Re: [Kamailio-Users] [OpenSIPS-Users] [OT] about scalability, experiences, demands
by Alex G 16 Nov '08

16 Nov '08

well put dan! I really don't understand why you would bother bringing this to our attention at all Henning. On Fri, Nov 14, 2008 at 11:12 PM, Dan Pascu <dan(a)ag-projects.com> wrote: > > I fail to see what worries you if all is fine and dandy in paradise. Don't > mind us, we are just a bunch of loonies with weird ideas anyway. > > On Friday 14 November 2008, Henning Westerholt wrote: > > Hello all, > > > > recently some statement came to my attention that "there is a common > > consent that the current design/architecture of [..] OpenSER (inherited > > from SER) is no longer able to deliver and to meet the present > > requirements and demands". > > > > I don't want to argue that much about this opinion, in fact the demands > > to a Voice over IP solution depends very much on the certain setup. But > > i want to share some details from my experiences in developing and > > operating a big VoIP infrastructure here at 1&1. > > > > We've about 2 million customers on our platform, that uses over 5 > > million individual numbers and terminate about 1 billion minutes per > > month. We're able to provide a good service with the actual > > architecture of OpenSER without any real problems. Of course there is > > always some room for improvements, but so far the main challenges we > > faced were not in the scalability or performance areas. More important > > issues are e.g. the inherent complexity of the SIP protocol and the > > maintainance of a good quality assurance and integration process. > > > > We started some years ago with OpenSER 0.9.5, which we then extended a > > lot in house. For example we implemented more than 25 own modules, a > > own path implementation, did a lot of bug fixing and workarounds for > > certain problems we've found. We're able to reduce this amount of > > proprietary code a lot in the past, because of progress in the OpenSER > > development, intregration of our "key" modules and a lot of other > > contributions. We're using now something between OpenSER 1.3 and > > Kamailio 1.4 with only a few private extensions. > > > > So in my opinion the actual design of our server is not "[..] an > > inevitable dead-end that needs to be avoided.". I rather think that > > we'll be able with continuing improvements to tackle the upcoming > > challenges well, especially as we will work together in the future with > > the SER developers in improving important areas of this software. > > > > But this is just my personal opinion, everybody is of course free to > > have their own position. > > > > With best regards, > > > > Henning Westerholt > > > > -- > Dan > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] [presence] NOTIFY sent over UDP when UA is waiting on TCP - (New bug reported)
by Pascal Maugeri 15 Nov '08

15 Nov '08

I think so. I've filled in a new issue in the tracker system (#2291718). Regards, Pascal On Fri, Nov 14, 2008 at 11:37 PM, Klaus Darilion < klaus.mailinglists(a)pernau.at> wrote: > looks like a bug in the presence module .... > > Pascal Maugeri wrote: > >> Hi >> >> We've seen the following problem: >> >> 1) the UA subscribe to a presentity with a SUBSCRIBE that contains this >> Contact header: >> "p1i" <sip:p1i@88.30.104.13:1414;transport=tcp> >> >> 2) then the presence module outputs in log file: >> >> INFO:presence:send_notify_request: NOTIFY sip:p1i@genaker.net<sip%3Ap1i(a)genaker.net><mailto: >> sip%3Ap1i(a)genaker.net <sip%253Ap1i(a)genaker.net>> via >> sip:p1i@88.30.104.13:1414;transport=tcp on behalf of >> sip:pascal@genaker.net <sip%3Apascal(a)genaker.net> <mailto: >> sip%3Apascal(a)genaker.net <sip%253Apascal(a)genaker.net>> for event presence >> >> 3) but the user agent receives the NOTIFY over UDP when it should be sent >> over TCP. >> >> Any idea of what could be wrong ? >> Is there any workaround to force sending NOTIFY messages over TCP ? >> >> Regards, >> Pascal >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

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[Kamailio-Users] [presence] NOTIFY sent over UDP when UA is waiting on TCP
by Pascal Maugeri 14 Nov '08

14 Nov '08

Hi We've seen the following problem: 1) the UA subscribe to a presentity with a SUBSCRIBE that contains this Contact header: "p1i" <sip:p1i@88.30.104.13:1414;transport=tcp> 2) then the presence module outputs in log file: INFO:presence:send_notify_request: NOTIFY sip:p1i@genaker.net<sip%3Ap1i(a)genaker.net>via sip:p1i@88.30.104.13:1414;transport=tcp on behalf of sip:pascal@genaker.net <sip%3Apascal(a)genaker.net> for event presence 3) but the user agent receives the NOTIFY over UDP when it should be sent over TCP. Any idea of what could be wrong ? Is there any workaround to force sending NOTIFY messages over TCP ? Regards, Pascal

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[Kamailio-Users] [presence] getting "record not found in hash_table" error
by Pascal Maugeri 14 Nov '08

14 Nov '08

Hi We get very ofter this error message when subscribing to presence: Nov 14 16:20:01 server2-operations /usr/local/sbin/kamailio[6224]: ERROR:presence:get_stored_info: record not found in hash_table Nov 14 16:20:01 server2-operations /usr/local/sbin/kamailio[6224]: ERROR:presence:handle_subscribe: getting stored info ... using kamailio 1.4 So far I haven't managed to find a way to reproduce it. Is there any solution or workaround ? Regards, Pascal

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[Kamailio-Users] How can i use the FIFO
by ALAEDDINE abbech 14 Nov '08

14 Nov '08

hi all I read the doc but i don't understand how can i use the FIFO to have many statistc. Can you give me a simple exemple. thanks.

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[Kamailio-Users] [OT] about scalability, experiences, demands
by Henning Westerholt 14 Nov '08

14 Nov '08

Hello all, recently some statement came to my attention that "there is a common consent that the current design/architecture of [..] OpenSER (inherited from SER) is no longer able to deliver and to meet the present requirements and demands". I don't want to argue that much about this opinion, in fact the demands to a Voice over IP solution depends very much on the certain setup. But i want to share some details from my experiences in developing and operating a big VoIP infrastructure here at 1&1. We've about 2 million customers on our platform, that uses over 5 million individual numbers and terminate about 1 billion minutes per month. We're able to provide a good service with the actual architecture of OpenSER without any real problems. Of course there is always some room for improvements, but so far the main challenges we faced were not in the scalability or performance areas. More important issues are e.g. the inherent complexity of the SIP protocol and the maintainance of a good quality assurance and integration process. We started some years ago with OpenSER 0.9.5, which we then extended a lot in house. For example we implemented more than 25 own modules, a own path implementation, did a lot of bug fixing and workarounds for certain problems we've found. We're able to reduce this amount of proprietary code a lot in the past, because of progress in the OpenSER development, intregration of our "key" modules and a lot of other contributions. We're using now something between OpenSER 1.3 and Kamailio 1.4 with only a few private extensions. So in my opinion the actual design of our server is not "[..] an inevitable dead-end that needs to be avoided.". I rather think that we'll be able with continuing improvements to tackle the upcoming challenges well, especially as we will work together in the future with the SER developers in improving important areas of this software. But this is just my personal opinion, everybody is of course free to have their own position. With best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany Vorstände: Henning Ahlert, Ralph Dommermuth, Matthias Ehrlich, Thomas Gottschlich, Robert Hoffmann, Markus Huhn, Hans-Henning Kettler, Dr. Oliver Mauss, Jan Oetjen - Aufsichtsratsvorsitzender: Michael Scheeren Amtsgericht Montabaur / HRB 6484

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Re: [Kamailio-Users] Problems with counter $var(i)
by ingdavidcespedes＠cable.net.co 14 Nov '08

14 Nov '08

Ok, now I understand. So the answer is, that you can't do that with script variables (http://www.kamailio.net/dokuwiki/doku.php/pseudovariables:1.4.x#script_vari…) there it says: Note: A script variable persists over the OpenSER process in which it was initialized, so be sure of giving it a new value before reading it or you'll get the value asigned in any other previous message processed by the same OpenSER process (pid). ----- Mensaje original ----- De: Ricardo Martinez <rmartinez(a)redvoiss.net> Fecha: Jueves, Noviembre 13, 2008 2:07 pm Asunto: Re: [Kamailio-Users] Problems with counter $var(i) > Hi David. > Yes, that exactly what i want, set the variable to 0 in the main > route,so every call that arrive it start the process again... > The problem is that in the failure_route the variable keeps the > value..and is not resetting to 0 > > Ricardo.- > > -----Mensaje original----- > De: ingdavidcespedes(a)cable.net.co [ingdavidcespedes(a)cable.net.co] > > Enviado el: jueves, 13 de noviembre de 2008 16:15 > Para: Ricardo Martinez > CC: users(a)lists.kamailio.org > Asunto: Re: [Kamailio-Users] Problems with counter $var(i) > > Hi > > In which place of your main route are you writing var(i)=0. Maybe > that'sthe problem, each time you enter in the main route you are > setting the i > variable to 0. > > David Cespedes > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] Problems with counter $var(i)
by ingdavidcespedes＠cable.net.co 13 Nov '08

13 Nov '08

Hi In which place of your main route are you writing var(i)=0. Maybe that's the problem, each time you enter in the main route you are setting the i variable to 0. David Cespedes Hello I realized that the counter $var(i) in the failure_route[1] seems not to be resetting to 0 when the 2nd , 3th ..etc call arrive. So for the second call I have the counter 0 in the main route, the route(3), route(1) but in the failure_route is set to 3 (the last values from the last call) I keep trying to figure it out why this is happening.. Anyone? Thanks, Ricardo.- De: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] En nombre de Ricardo Martinez Enviado el: jueves, 13 de noviembre de 2008 11:46 Para: users(a)lists.kamailio.org Asunto: [Kamailio-Users] Problems with counter $var(i) Sorry ¡ i forgot to put the subject. Hello list. I'm having some problems trying to build a counter for my kamailio.cfg file. This is what I have.. main() ... load avp from radius} var(i)=0; route(3); ... route[1] { t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; xlog("L_INFO","[$ci] $rm t_relay a $ru"); exit; } route(3) ... $var(nr)=$avp(s:caller_nr); If ( var(i) < $var(nr) ); $var(i) = $var(i) + 1; route(1); else exit; failure_route[1] { xlog("L_INFO", "[$ci] $rm:$ru failure_route\n"); if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled en failure_route\n"); exit; } if (!next_gw()) { xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n"); route(3); } else { xlog("L_INFO", "XXXXX"); } t_on_failure("1"); t_relay(); } After I restart the service the first time a call arrives the counter seems to go OK, the counter goes from 0 to 2 (if the caller_nr=3). But the second time the $var(i) at the main and route(3) it's = 0, but in the failure_route it keeps the value = 3. What I'm doing wrong? Can someone help me here? Thanks Ricardo Martinez.- _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] Using other media gateways
by Ruchir 13 Nov '08

13 Nov '08

I have a question. We can use media proxy to handle media through media proxies. I was wondering if it is possible to use any media servers for rtp by just specifying ip address of the media gateway.

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[Kamailio-Users] help understanding kamailio.cfg and processing call flow
by Fabian Borot 13 Nov '08

13 Nov '08

hello I have been struggling to get the grasp on kamailio for weeks now. My prob is not understanding the call flow but how to translate it using the functions provided by the modules and the core. I do not have clear the scope or reach of several functions,provided by either the core or the modules. Example: Register with Auth: 1- user sends REGISTER request w/o the encripted credentials 2- kamailio answers with 401 and sends the challenge 3- the uac sends the REGISTER again, this time with the right credentials and after verifying kamailio sends 200 OK 4- if the user is behind a NAT we can fix the contact info before saving it to the location db. With the help of the forum and the tutorials I have found in the internet I have been able to accomplish it with these commands: route{ if (uri==myself) { if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info.\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("1")){ xlog("L_INFO","mylog: REGISTER-Device behind NAT.\n"); fix_nated_register(); } force_rport(); save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } route(1); } route[1] { if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } but here come the first questions: 1- when the uac sends the first REGISTER, the function www_challenge() is the one that actually sends the "401" message? or the "exit" just exit out the "if" block and the script will continue with route(1) which in turns executes "t_relay()? ++from the doc, www_challenge() function++++ The function challenges a user agent. It will generate a WWW-Authorize header field containing a digest challenge, it will put the header field into a response generated from the request the server is processing and send the reply. +++++++++++++++++++++++++++++++++ 2- the "exit" after the save("location") function I assume would stop the execution of the script and at that point kamailio responds with "200 ok"? or just exit out the "if" block, the script then reaches the "route(1)" section and the "t_relay()" sends the "200 ok"? ++ from the doc, core-cookbook, exit() function+++++++ Stop the execution of the configuration script – it has the same behaviour as return(0). It does not affect the implicit action to be taken after script execution. ++++++++++++++++++++++++++++ 3- the "t_relay" function: does that function creates and sends the approriate response for each request? ex1: an INVITE is received, does t_relay() "know" that it can send "100 trying" automatically? ex2: a "200 OK" is received as a response to an INVITE, does t_relay knows that it has to send an "ACK"? ++++++from the doc, t_relay() funcion ++++++++++ >From user's perspective, the major function is t_relay(). It setup transaction state, absorb retransmissions from upstream, generate downstream retransmissions and correlate replies to requests. +++++++++++++++++++++++++++++ 4- why do I see ";" even after a "}", is it needed sometimes? 5- is "exit" the same as "exit()"? These may sound like dumb questions for some of the experienced users, but for a newbie they are not. Also, if this sounds like " hey, code my config file for me" I assure you that that is not the intended purpose. I would appreciate any help and thank everybody in advance. Fabian _________________________________________________________________ Color coding for safety: Windows Live Hotmail alerts you to suspicious email. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety…

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sr-users November 2008