sr-users January 2015

sr-users@lists.kamailio.org

94 participants
136 discussions

Possible memory leak dealing with presence in kamailio
by Nuno Reis 12 Feb '15

12 Feb '15

Hello all. I'm consistently watching a memory increase in kamailio when dealing with PRESENCE events, namely SIP PUBLISH events. The system eventually hangs running out of memory. This behavior is seen at least in kamailio 4.1 and 4.2. I'm currently using the latest stable 4.2.2. If I disable the SIP PUBLISH handling in kamailio i don't observe the issue anymore but as a side effect I don't have presence (name BLFs) also. What do you think can be the right approach here? Should I open an issue in github for this? Should I run kamailio under valgrind for some time? Are there any other possible debug hints here? Please find attached a code snippet with the presence related parts I'm using right now. Looking forward to hear from you. Best Regards, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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Re: [SR-Users] Wrong location entries when using usrloc with Mongo
by Mickael Marrache 10 Feb '15

10 Feb '15

I forgot to precise that I allow only one contact per AOR. modparam("registrar", "max_contacts", 1) save("location", "0x04") From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Friday, January 30, 2015 10:51 AM To: sr-users(a)lists.sip-router.org Subject: Wrong location entries when using usrloc with Mongo Hi, I start with no location nor in Mongo, nor in memory. My UA registers successfully and I can see the location entry in Mongo. Then, I close my UA which unregisters (by setting Expires header to 0). Then, I open the app again and a new registration is made. The entry after first registration. Looks okay. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "username" : "A", "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" : ISODate("2015-01-30T08:55:34Z"), "q" : -1, "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8809, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip:XXXXXXXXXXX:54217;transport=tcp", "path" : "<sip:XXXXXXXXXX;lr;received=sip:XXXXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp:XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:55:34Z"), "ruid" : "uloc-54cb38df-3378-2", "instance" : null, "reg_id" : 0 } The same entry after un register (Expires 0). Note that the username field is missing. In any case, I expected the entry to be deleted. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" : ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0, "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" } The entries after second registration. The new entry looks okay. But, the old entry is still here. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" : ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0, "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" } { "_id" : ObjectId("54cb3a7884e581337a25b895"), "username" : "A", "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" : ISODate("2015-01-30T09:02:00Z"), "q" : -1, "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8813, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T08:02:00Z"), "ruid" : "uloc-54cb38df-337a-1", "instance" : null, "reg_id" : 0 } The issue is the entry is not deleted after un registering. I precise that registrations are dispatched over multiple registrars with all accesses the same Mongo cluster. Thanks, Mickael

2 7

Setting dialog variable when BYE is received before accounting
by Mickael Marrache 10 Feb '15

10 Feb '15

Hi, Is there a way to set a dialog variable when a BYE is received so that it can be written in a CDR? I tried as follows but it didn't work: if (loose_route()) { .. if(is_method("BYE")) { if(is_direction("downstream")) { #Call disconnected by caller $dlg_var(dr) = 700; } else { #Call disconnected by callee $dlg_var(dr) = 701; } xlog("Setting disconnect reason to $dlg_var(dr)\n"); } .. exit; } The xlog statement prints: ERROR: <script>: Setting disconnect reason to <null> The doc states: This pseudo-variable will be available only for subsequential requests after doing loose_route(). Thanks, Mickael

2 1

Multiple OR into a if
by Igor Potjevlesch 06 Feb '15

06 Feb '15

Hello, I'm very disappointed because of the following behaviour: if ($tu=~"^sip:0[1-9]{9}") { [.] In that case, Kamailio returns TRUE because the instructions in the block are executed. } if ($rU=~"^33" || $rU=~"^0033" || $rU=~"^0[1-9]{9}" || $rU=~"^\+33" || $tu=~"^sip:33" || $tu=~"^sip:0033" || $tu=~"^sip:0[1-9]{9}" || $tu=~"^sip:\+33" ) { [.] In that case, Kamailio should return FALSE because the instructions in the block are not executed. } Am I missed something regarding 'OR' ? Regards, Igor.

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Kamailio load balansing 2 servers issues
by Yuriy Gorlichenko 06 Feb '15

06 Feb '15

Hello. We use 2 kamailio server 4.3 master brancher for load balansing cluster. We have some problems with this deplooiment fist of all we have REgstering issue: Witth one server all works fine (we have some endpoints with same creditians) all endpoints reinging well. But at 2 servers we hawe problem with this- call ringing only at one endpoint. We use Database for USRLOC module modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) As i think it mean al connections to servers ives at database. We have one database for 2 servers and this is means it only one table LOCATION for its. The second problen id that when loadbalanser sends bye to kamailio from the outside- bye is not recieves to endpoint. It say that no transaction with this CallID. I think it is because transaction goues through another kamailio and kamiaio that recieves BYE does not know about it transaction. So my first qestion- Does anybody have some information about loafd balansiong between kamailios? And second question - how to resolve issue with LOCATION table? Thanks

3 3

Issue with dlg_setflag vs. setflag
by Anthony Messina 06 Feb '15

06 Feb '15

I'm currently using Kamailio built from master@e59db79 and have been following the recent threads about the dialog module as I'm moving toward adding dialog support for presence and CDRs in my setup. However, I seem to have come across an issue where dlg_setflag() does nothing, while setflag() works properly with the following (trimmed) script. So for now, I issue setflag(FLT_DIALOG) for INVITE rather than dlg_setflag(FLT_DIALOG). Is there something I'm missing about dlg_setflag()? #!define FLT_DIALOG 4 request_route { route(REQINIT); route(NATDETECT); if(is_method("CANCEL")) { if(t_check_trans()) { route(RELAY); } exit; } route(WITHINDLG); if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); route(AUTH); remove_hf("Route"); if(is_method("INVITE|SUBSCRIBE")) record_route(); if(is_method("INVITE")) { setflag(FLT_ACC); # Enable dialog support (dlg_setflag not working) #dlg_setflag(FLT_DIALOG); setflag(FLT_DIALOG); } route(SIPOUT); route(PRESENCE); route(REGISTRAR); if($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(PSTN); route(LOCATION); route(RELAY); } route[WITHINDLG] { if(!has_totag()) return; if(loose_route()) { route(DLGURI); if(is_method("BYE")) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); # Testing dialog flag xlog("L_INFO", "Completed $dlg(from_uri) to $dlg(to_uri) - $DLG_lifetime duration\n"); } else if(is_method("ACK")) { route(NATMANAGE); } else if(is_method("NOTIFY")) { record_route(); } route(RELAY); exit; } if(is_method("SUBSCRIBE") && uri==myself) { route(PRESENCE); exit; } if(is_method("ACK")) { if(t_check_trans()) { route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not Found"); exit; } -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 2

kamailio 4.2.1 core in timer.c:877
by 张顺通 03 Feb '15

03 Feb '15

I user rtimer module : loadmodule "rtimer.so" modparam("rtimer", "timer", "name=p_route;interval=1;mode=1;") modparam("rtimer", "exec", "timer=p_route;route=P_ROUTE") application will coredump in sometimes, and core file in below, what's the reason? Program terminated with signal 11, Segmentation fault. #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 877 timer.c: No such file or directory. in timer.c (gdb) bt full #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 tl = 0x7f9ed47f7e98 ret = 0 #1 0x00000000005fc686 in timer_handler () at timer.c:953 saved_ticks = 1043586690 run_slow_timer = 0 i = 268 __FUNCTION__ = "timer_handler" #2 0x00000000005fdd4d in timer_main () at timer.c:992 No locals. #3 0x00000000004aa571 in main_loop () at main.c:1700 i = 12 pid = 0 si = 0x0 si_desc = "udp receiver child=11 sock=10.1.*.*:5160\000\000\b\000\000\000\000\000\000\000\000\060S\324\236\177\000\000\060=#z\001\000\000\000(\205V\324\236\177\000\000\060=#z\377\177\000\000\214\216N\000\000\000\000\000\037t\372D\000\000\000\000(\205V\324\236\177\000\000\220@ #z\377\177\000\000\000\000\000\000\001\000\000" nrprocs = 12 __FUNCTION__ = "main_loop" #4 0x00000000004af166 in main (argc=11, argv=0x7fff7a234098) at main.c:2561 cfg_stream = 0x1da75e0 c = -1 r = 0 tmp = 0x7fff7a234681 "" tmp_len = 32767 port = 2049130512 proto = 32673 options = 0x706ec0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 220012340 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xb p = 0x0 __FUNCTION__ = "main"

2 5

Video Key-Frame Request using RTCP FIR or SIP INFO message
by Muhammad Shahzad 02 Feb '15

02 Feb '15

Hi, This may be a bit out of focus topic for this forum but i am posting it here anyway with hope that some guru would shed some light on it and point me to right direction. The problem is that i want to establish video call between a webrtc and a sip client using kamailio (for signalling) and RTPEngine (for media relay). Both signalling and the audio stream seems to work perfectly fine The remote video on webrtc client side (i.e. video stream from sip client) takes about 20-30 seconds to establish but once it starts it works fine. However, the remote video on sip client side (i.e. video stream from webrtc client) starts almost immediately (within 3-5 seconds) but it gets stuck after 1 or 2 seconds, then it goes blank after about 30 seconds. After a long discussion with sip client developer, we now understand the fact that sip client sends a request for so called key-frame, which is ignored by webrtc client. This request is sent through both RTCP stream and SIP INFO message. The SIP INFO message seems to be pointless as media is internally managed by chrome/firefox and these browsers don't give us such sophisticated access and control over media streams. Please let me know if this assumption is wrong. For the RTCP stream based request (RTCP-FIR), i only see "Invalid RTCP packet type" error message in RTPEngine logs (not sure if it drops this packet or relay it anyway). Does anyone has any idea on how can we either, 1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO message and issue a key-frame in RTP video stream in response to this SIP request? OR 2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video stream on webrtc client's behalf? If there is any other solution to this, please feel free to share. Thank you.

4 5

What could cause an intermittent fault (28) with http_query()?
by Tim Chubb 01 Feb '15

01 Feb '15

Hi All I seem to be experiencing an intermittent fault with the utils http_query() method. We are implementing a routing and white list component that is accessed via a REST api, however i have observed several occasions where this is logged: Jan 30 16:04:17 vs-kam-prod02 /usr/local/sbin/kamailio[13184]: ERROR: utils [functions.c:149]: http_query(): failed to perform curl (28) The indicated error number (28) seems to suggest a timeout is occurring with curl, however examining a capture of network traffic when this happens shows that a http request is not sent from the server to the destination at all, usually attempting a second call results in everything working correctly. As stated its intermittent in its nature and so I cannot reliably trigger this issue, other than through sheer repetition, so any ideas as to what could be causing this issue would be gratefully received,

2 2

Async module taking down our server
by Will Ferrer 31 Jan '15

31 Jan '15

Hi All We are trying to use the async module to to delay sending a bye on from one end of the call to the other. We are using async_route(routename, seconds) to delay the WITHINDLG route. The idea is that in the future we want to be able to have our billing min duration enforced (though currently we are having issues with the dialog module that we are discussing in another thread). After running this on our deploy servers, the delays before sending on the byes get longer and longer, and then kamailio goes down. Then the receive udp buffer fills up. We tried it with both 4 and 400 async workers, and it made no difference. I am including a screen capture of the servers stats when this happens taken from voip monitor. Here are the relevant parts of the config: ... loadmodule "async.so" ... modparam("async", "workers", ASYNC_THREADS) ... request_route { ... route(DELAYED_BYE_STATIC); ... route[DELAYED_BYE_STATIC] { xlog("L_DEBUG","route DELAYED_BYE_STATIC"); #!ifdef WITH_DELAYED_BYE_STATIC if (is_method("BYE")) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, from self \n"); #if (from_uri == myself) { if ((allow_trusted() || allow_source_address()) && from_uri == myself) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, Bye detected, from self \n"); send_reply("200", "OK"); xlog("L_DEBUG","route DELAYED_BYE_STATIC, sent 200 about to sleep \n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails if (has_totag()) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG_DELAYED \n"); async_route("WITHINDLG_DELAYED", MIN_DURATION); } else { xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG \n"); async_route("WITHINDLG", MIN_DURATION); } xlog("L_DEBUG","route DELAYED_BYE_STATIC, slept\n"); exit; } } #!endif return; } ... route[WITHINDLG_DELAYED] { xlog("L_DEBUG", "route WITHINDLG_DELAYED: triggered \n"); $avp(was_delayed) = 1; route(WITHINDLG); } ... route[WITHINDLG] { xlog("L_DEBUG", "route WITHINDLG: will -- DLG triggered, request method: $rm \n"); #!ifdef WITH_DISPATCHER if(is_method("BYE|CANCEL")) { xlog("L_DEBUG","route WITHINDLG: cancel or bye detected, request method: $rm \n"); #!ifdef WITH_DISPATCHER_LOAD_AWARE xlog("L_DEBUG","route WITHINDLG: running ds_load_update, request method: $rm \n"); ds_load_update(); #dlg_get ("$ci","$ft","$tt"); #dlg_bye ("all"); #!endif } #!endif if (has_totag() || $avp(was_delayed) == 1) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG has totag or was_delayed: $avp(was_delayed) \n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG has loose route \n"); route(DLGURI); if (is_method("BYE")) { xlog("L_DEBUG","route WITHINDLG: BYE detected"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails xlog("L_DEBUG","route WITHINDLG: ACC flag set"); } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } xlog("L_DEBUG", "route WITHINDLG: will -- DLG RELAY 1\n"); route(RELAY); } else { xlog("L_DEBUG", "route WITHINDLG: will -- DLG else \n"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG is ack \n"); if ( t_check_trans() ) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG t_check_trans \n"); # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_DEBUG", "route WITHINDLG: will -- DLG RELAY 2\n"); route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } ... Does any one know if this is a bug or a leak with in the async module, or perhaps something I am doing in my config? Thanks in advance for an assistance you can offer me. All the best. Will Ferrer Switchsoft

3 5

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sr-users January 2015