sr-users April 2010

sr-users@lists.kamailio.org

65 participants
103 discussions

B2BUA removing a=nortpproxy:yes (rtpproxy-1 <---RTP---> rtpproxy-2)
by Iñaki Baz Castillo 26 Apr '10

26 Apr '10

Hi, I must deal with a B2BUA which keeps the original SDP unchanged except the fact that it removes the "a=nortpproxy:yes" line added by RtpProxy. The B2BUA intercommunicates two Kamailio, both forcing its own RtpProxy server. I've not tested it again but expect there will be no RTP as each RtpProxy will wait for RTP coming from the other. Am I right? Of course I could force RtpProxy just in NAT cases and so, but I prefer to force it always so the clients of each Kamailio can configure QoS for the media addresses (different in each server). Is there some workaround for it? In the other hand: is it possible two different Kamailio's to share the same RtpProxy server? I've never tryied it. If it's valid, would it solve the B2BUA issue? (I don't think so as there are still different media sessions created by different kamailio instances). Any thought? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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htable size
by Alex Balashov 26 Apr '10

26 Apr '10

The htable size is static; it doesn't dynamically resize itself up to the next power of 2 after a certain utilisation threshold, right? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Re-invite problems with t.38?
by Espen Berg 26 Apr '10

26 Apr '10

Hello all! I finally have something that I could call a working config (almost that is), the proxy works fine with voice and fax sent over g.711 (but I guess re-invites will cause problems in this scenario also). The problem occurs when I try to switch to T.38. Since it works with voice my guess is that the problem is caused by the re-invite to t.38. I guess it has to be solved by a if (loose_route()) section, but I am little bit clueless right now, I have tried multiple variants but with no success. RTP proxy is started with the -l flag to distinguish between local and public traffic. -l PUBLIC.IP.ETH0/LOCAL.IP.ETH1 The setup: Sonus UAC/Trunk <-private ip-> (eth1)Kamailio/RTPProxy(eth0) <-public ip-> Asterisk IP.TO.TRUNK = Sonus IP, public address range, but only reachable by private net (from the range of LOCAL.IP.ETH1). LOCAL.IP.ETH1 = Local net, static route to IP.TO.TRUNK. PUBLIC.IP.ETH0 = Public routable address. IP.TO.ASTERISK = Public routable address. The purpose of Kamailio/RTPProxy are to proxy everything between the routable and the private net. Does anyone here have any idea of how I'm going to solve my issue? It would also be nice if anyone here have better solutions to my approach, and there could also be some other errors in my config that causes this behaviour. ####### Routing ######## route { if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; } if (method != "INVITE" && method != "ACK" && method != "BYE" && method != "CANCEL") { sl_send_reply("503", "Service Unavailable"); exit; } if (!has_totag()) { record_route(); route("1"); } if (has_totag()) { if (!loose_route()) { if (t_check_trans()) { t_relay(); exit; } exit; } } if (has_body("application/sdp")) { if (method == "INVITE") { route("2"); t_on_reply("2"); } else if (method == "ACK") { rtpproxy_answer(); } } else if (method == "INVITE") { t_on_reply("2"); } if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } t_relay(); } #RELAY BY SOURCE ROUTE route[1] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") { force_send_socket("PUBLIC.IP.ETH0"); rewritehost("IP.TO.ASTERISK"); } else { force_send_socket("LOCAL.IP.ETH1"); rewritehost("IP.TO.TRUNK"); } } # RTP OFFER ROUTE route[2] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") force_rtp_proxy("eir"); else force_rtp_proxy("ier"); } # RTP OFFER ONREPLY onreply_route[1] { if (has_body("application/sdp")) route("2"); } # RTP ANSWER ONREPLY onreply_route[2] { if (has_body("application/sdp")) route("2"); } ## END OF CONFIG Sorry for the long post, but I hope someone outthere are able to help med with this issue. -- Espen C. Berg, Driftstekniker - Monsternett AS (www.monsternett.no) Violgata 3A, N-1776 HALDEN, NORWAY Sentralbord: 69701800 - Direkte: 69701810 - Faks: 69701801

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meeting in Berlin, June 7-8
by Daniel-Constantin Mierla 25 Apr '10

25 Apr '10

Hello, several developers will be around Berlin in June 7-8 time frame and we think to organize a face to face meeting. It is a good time frame if you want to attend Amoocon in Rostock (June 4-6, aprox 230km north of Berlin) or LinuxTag (June 9-12, in Berlin). http://www.linuxtag.org/2010/ http://www.amoocon.de/2010 I will have two presentations at Amoocon, about Asynchronous processing in SIP Router / Kamailio and extending your routing logic with Lua (via the new app_lua module). Hennig has one talk at LinuxTag about Geographical redundant VoIP systems with Kamailio. So far, interest in event and participation was shown by: - Andrei Pelinescu-Onciul - Jiri Kuthan - Inaki Baz Castillo - Jesus Rodriguez - Elena-Ramona Modroiu - Daniel-Constantin Mierla - Henning Westerholt - Marcus Hunger A good number to move it into public space and see if others from community are interested in such event. Based on that we will decide the type of event and look for a proper place to host it. Among topics could be: hands on hacking or docs update sessions, quick start guide to start writing your own extensions as module, open debates about current devel state and future directions... Please drop an email if you want to participate to the event - there is no participation fee - however, late registrations may be rejected if the capacity of location is filled - priority is first come first served. You can email me at miconda(a)gmail.com or to registration(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

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[OT] Hitler should use Kamailio
by Iñaki Baz Castillo 24 Apr '10

24 Apr '10

http://www.youtube.com/watch?v=4nKh8bg3eHo -- Iñaki Baz Castillo <ibc(a)aliax.net>

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LCR in openser 1.2 - is it possible to disable a gateway?
by Iñaki Baz Castillo 24 Apr '10

24 Apr '10

Hi, I need to add LCR support to an existing OpenSer 1.2 server (not possible to upgrade it for now). I've realized that 1.2 doesn't include gateways monitorize feature. Ok, I can do the check with an external tool, but I would like some way to dissable such gw from the LCR module, something like upgrading the gw table and performing the MI reload function. But I don't see how to set in the gw table that such gateway is disabled. Is it possible? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Solutions to missing BYEs, accounting for them
by Alex Balashov 23 Apr '10

23 Apr '10

Hi all, Please forgive the slightly long post, but if you have anything to contribute on this topic, please consider giving it a read as I could really use your input. :-) As I'm sure many others of you running proxy-based service delivery platforms of some description also, I am faced with the problem of trying to account for calls with missing BYEs in a realistic way. There is no shortage of mailing list posts over the years on this topic. Inevitably, in a platform with sufficient call volume, with some NAT'd and/or endpoint diversity and other technical causes, there will be some calls that are never officially terminated from the point of view of a proxy. The ability of the 'dialog' module to spoof bidirectional BYEs on timeout[1] goes a long way toward addressing this problem theoretically. However, there are practical obstacles to relying on it solely as a solution, mainly because there is not an acceptable timeout value to use as a trade-off. If the timeout period is set to a very low value, users will obviously complain, and in any case, depending on the destination, the worst-case scenario for maximum call billing may still be far too high. If the timeout period is set high--perhaps something like 5-8 hours--then all calls that fail to end in the normal way will be billed some excessively large amount that certainly will not sit well with users either. If either the core delivery element of the platform or the user agent is tightly controlled by the operator of the proxy from an administrative point of view, it is indeed probably possible to rely on RTP timeouts or SIP Session Timers (SSTs) on one of the endpoints. That doesn't create a satisfying resolution for those of us dealing with indeterminate call completion scenarios with a great deal of user and vendor diversity, though. For instance, I route to about 15 ITSPs and carriers; I think maybe one of them does 15-minute SSTs, and the rest are certainly not going to turn them on just for me, even if their SBCs/switches/things have the capability. The user endpoints are mostly Asterisk and do RTP timeout, of course, and in most cases I do get the resulting BYE. However, this discussion is about the minute but nontrivial percentage of cases in which I do not get the BYE, whether because of NAT statekeeping problems or network reachability or whatever underlying causes--in truth, I cannot accurately characterise these. So, it seems to me that from a theoretical point of view, there are basically two directions someone in this position can go from here: 1) Inline B2BUA in the signaling path of all calls; 1a) Make it do SSTs; or 1b) Make it relay media, too, and hang up the call (bidirectional BYE) on RTP receive timeout; 2) Couple the proxy to an RTP relay and provide some mechanism by which the proxy can be made aware, in an asynchronous fashion, that an RTP timeout was detected by the relay. It seems to me from a brief and informal survey of prior mailing list literature that #1 is the usually recommended option here. If #1 is pursued, what is the best tool to use in the Kamailio/SIP-Router-oriented ecosystem? My default instinct would say SEMS; I really like SEMS, and use it a lot for various related chores. The problem is that the pre-built modules and examples for SEMS mostly center on application-level functionality, while low-level documentation of its powerful C++ API is a bit impoverished, so this would take a lot of work. Needless to say, I am interested in the option that requires the least work but still solves the problem in an elegant way from a technical and--dare I say--aesthetic perspective. For instance, it seems clear from looking at the SEMS-1.1.1 sources that SSTs are supported in principle in core/plug-in/session_timer. But unless I am missing something, I cannot find anywhere in the sources or examples where it is actually used. So, I suppose one option is to figure out how to make this stuff work in SEMS, and make it work. But for some reason who is not attune to the universe of its C++ API, it is a rather formidable chore. I think the same would hold true of making it observe bidirectional RTP timeout. Turning attention to option #2, I have looked at rtpproxy (my preferred default), iptrtpproxy, and mediaproxy modules but have not found any evidence that the control protocols Kamailio/SR uses to engage them support any notion of backward asynchronous feedback in case of RTP timeout. It would be really nice if one of these stream control protocols was augmented to kick back a packet to Kamailio that can be caught in a special event_route, like event_route[nathelper:rtp-stream-timeout], but that is clearly not the case today. To be honest, I would not use MediaProxy even if it had this feature, because, well, let's be bluntly honest and acknowledge what the more politically aware presumably already conjecture: in light of AG Projects' zealous OpenSIPS partnership, it's difficult to muster confidence in future compatibility of MediaProxy with Kamailio. The module is there, it works, and I'm sure its maintainers are dedicated to doing whatever it takes to reverse engineer and keep it working, lift patches from OpenSIPS as necessary, etc., but who wants to be on the wrong side of the project ecosystem fence? Not I. That leaves iptrtpproxy, whose 'switchboard' concept I do not fully comprehend due to lack of experience with it, but which holds a potentially viable, if slightly kludgy/Rube Goldbergian answer. Of the three RTP proxies, it is the only one that provides a ready means of exporting a list of media streams it is currently tracking, together with statistics on how many packets have been received, etc. It is not inconceivable to cook up an external process that will frequently check this 'switchboard', as it were, and incite Kamailio/SR to do dlg_bye() via MI if it appears that the media stream has disappeared from either side; the dialog module helpfully exports the MI command dlg_end_dlg. Still, this does not seem nearly as parsimonious and reliable a solution as simply building some kind of RTP stream leg timeout notification into the control socket. After all, the control socket is open persistently, right, not on-demand? The various RTP proxies all seem to have some kind of dead peer detection internally in order to have some means of gracefully expiring resources allocated to media streams that have gone away, so it would just be a matter of passing a control frame up the socket to Kamailio/SR and wiring that to a custom event_route or a more static callback in the code. By the way, I should mention that I am aware of and historically very sympathetic to the perspective that this kind of call control is alien to the nature of a proxy, and an appropriate job for UAs and not proxies at all. However, we all have to make pragmatic concessions to the realities of real-world operation, which I assume is the motivation for dialog timeouts, dlg_bye(), and other perversions from the point of view of a purist. :-) I welcome your thoughts and suggestions about the easiest and most technically meritorious approach. Thanks, -- Alex [1] Enabled via $dlg_ctx(timeout_bye) = 1 -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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MI functions
by Hector.Ortiz＠swisscom.com 23 Apr '10

23 Apr '10

Hi everyone, In the page where a module is described, there is a list of dependencies that have to be met in order to use the module, the list of exported parameters that might need to be set in the configuration file and a list of exported functions which can be used in the configuration script. I've noticed as well, that there are some "Exported MI functions" but I was wondering if any of you know what the purpose of those functions is, where and how such functions could be used? Thanks Héctor

2 1

Error with Presence: no shmem
by Tristan Mahé 23 Apr '10

23 Apr '10

Hi List, I'm writing a little message because I encountered today a strange error on my Kamailio 3.0 proxy/presence server ( and can't ask on IRC there's nobody atm ;p ) I get these errors in the logs since a few hours: Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: tm [t_msgbuilder.c:1371]: build_uac_req(): no shmem Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: tm [uac.c:327]: t_uac: Error while building message Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1577]: in function tmb.t_request_within Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1673]: sending Notify not successful Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1348]: Could not send notify for dialog That prevents NOTIFYs from being sent, but I can also see sometimes NOTIFYs get sent without theses errors. The strange thing is that the error indicate no shared memory, but on the server, I'm far from being memless (16G total RAM, 4G free not even used for cache ). I have presence, presence_xml, presence_dialoginfo, pua, pua_dialoginfo and pua_bla loaded. Anyone got an idea on where to look ? Regards, Gled.

2 1

[Kamailio-Users] How can i install and configuration CDRTool + Freeradius + mediaproxy + kamailio
by RAJNIKANT VANZA 23 Apr '10

23 Apr '10

Hi All, I would like to use CDRTool for accounting and integrate with kamailio + mediaproxy + freeradius. so, please let me know about configuration or installation steps for CDRToolon centos-5.X os. I have followed this link for CDRTool configuration http://cdrtool .ag-projects.com/wiki/Install but not success and confused. I have seen above url for CDRTool that provides information for deban os. but i want to install on centos. I am installed and run kamailio and mediaproxy successful. I am confused about compile or install about CDRTool and i want to help about Freeradius integration with kamailo and CDRTool. Thanks in advance. -- Best Regards, Rajnikant

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sr-users April 2010