Re: [SR-Users] Over 100% of CPU usage on kamailio
Hello, On 6/24/12 10:50 PM, Konstantin M. wrote: Hello, Is there any reason to consume over 135% of CPU usage on kamailio ? (sreenshot: http://i48.tinypic.com/2u72hr8.png). Any ideas ? do you have heavy traffic on that instance? How many children have you configured? What you can do is to attach with gdb to a process using lot of CPU and do the backtrace: gdb /path/to/kamailio __pid__ Replace __pid__ with the PID of process eating the CPU. Then run bt It may be a deadlock/infinite loop somewhere. I saw three processes in top, others were down with no much cpu usage. Is the SIP routing going fine anyhow when CPU usage is high? How do you solve it, by restart or it just appears from time to time and solves itself periodically? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Reply 503 was replaced to 500
Hello, this 503 to 500 is a requirement from RFC, to prevent propagation of blacklisting/disabling destination hosts. I don't remember right now any configuration option for it, but you can try to enforce it from the failure route, like: t_reply(503, ...); Cheers, Daniel On 6/24/12 4:30 PM, Uri Shacked wrote: I just read the topic - _Copy reason field from 503 to 500. _ Is there a way to change it if i want to send back the original leg 2 503 reply? On Sun, Jun 24, 2012 at 4:17 PM, Uri Shacked ushac...@gmail.com mailto:ushac...@gmail.com wrote: Hi, Kamailio server is behind our company's softswitch and acts as a sip application server. I notice that there are calls that the softswitch replied with 503 service unavailable and kamailio sent to the originator leg 500 service unavaileable. When kamailio recieved 504 or 502 it sends them back as is. shouldn't it be the same with 503? It also does not have a to tag in the CDR. And the to tag in the 503 that was recieved is not equal to the 500 reply to tag kamailio sent back. any ideas? BR, Uri ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] ERROR: uri2dst: bad_uri
Hello, On 6/22/12 4:13 PM, Barthel Marco (CI/AFU1) wrote: Hi, we are using kamailio at Bosch with some several thousand users. From time to time we encounter the following in the logs of kamailio: /usr/sbin/kamailio[9247]: ERROR: tm [ut.h:255]: ERROR: uri2dst: bad_uri: sip: /usr/sbin/kamailio[9247]: ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches Am I right that this means a relay/forward happened and either $ru or $du was just set to sip: ? yes, it looks like. Check your rules for updating $ru/$du - it may be also the incoming request, probably unlikely. Cheers, Daniel Thanks in advance. Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com Tel. +49 711 811-3602341 marco.bart...@de.bosch.com Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Hermann Scholl; Geschäftsführung: Franz Fehrenbach, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Volkmar Denner, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Fwd: Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm
Within Kamailio there is nothing you can do to trigger a reINVITE. You need a B2BUA (e.g. Asterisk) to force a reINVITE, and even then it is not sure that the SIP client sends properly updated SDP and contact header (I would try this with a manually sent reINVITE). Further, even if there is no reINVITE, you should still have audio. How do you handle the media stream? Is it sent directly to Asterisk? Is there rtpproxy inbetween? (if yes, then you need the reINVITEs so that rtpproxy will accept the new source IP address of the RTP stream (lock-in)). regards Klaus On 22.06.2012 15:58, Shaun Clark wrote: Forgot to post the response to the list as well. Date: Fri, Jun 22, 2012 at 6:57 AM Subject: Re: [SR-Users] Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm To: Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at Thanks for the response! I see a series of what I believe are re-REGISTER statements: Message sent: (to dest=75.101.244.XXX:5060) REGISTER sip:75.101.244.XXX SIP/2.0 Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK1839704852 From: sip:99...@75.101.244.xxx;tag=1689684502 To: sip:99...@75.101.244.xxx Call-ID: 1867622191 CSeq: 1 REGISTER Contact: sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22 Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 Received message: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61 tel:32.158.143.61;rport=2407;branch=z9hG4bK1839704852 From: sip:99...@75.101.244.xxx;tag=1689684502 To: sip:99...@75.101.244.xxx;tag=c97b4d1cb1f3d0da549e06a8d482ef63.e10d Call-ID: 1867622191 CSeq: 1 REGISTER WWW-Authenticate: Digest realm=75.101.244.XXX, nonce=4fe476e50e00aa8700d49b8c668f4c6f1e6a367f2XXX Server: Kamailio Content-Length: 0 REGISTER sip:75.101.244.XXX SIP/2.0 Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK123406454 From: sip:99...@75.101.244.xxx;tag=1689684502 To: sip:99...@75.101.244.xxx Call-ID: 1867622191 CSeq: 2 REGISTER Contact: sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22 Authorization: Digest username=990XX, realm=75.101.244.XXX, nonce=4fe476e50e00aa8700d49b8c668f4c6f1e6a367f2XXX, uri=sip:75.101.244.XXX, response=1e1d558894f2c05c322c76efbb2f9XXX, algorithm=MD5 Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 Received message: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61 tel:32.158.143.61;rport=2407;branch=z9hG4bK123406454 From: sip:99...@75.101.244.xxx;tag=1689684502 To: sip:99...@75.101.244.xxx;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8b5a Call-ID: 1867622191 CSeq: 2 REGISTER Contact: sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22;expires=120, sip:990XX@50.43.101.83:51879;line=59ecc207f06f4e9;expires=81 Server: Kamailio Content-Length: 0 But after this I would expect to see an INVITE but one is never sent, but if I switch back to the original IP on that device the call is reconnected, so it proves we're missing an INVITE I believe. What do I need to do on the server side to force a re-INVITE to be sent after this registration occurs? Thanks! On Fri, Jun 22, 2012 at 1:14 AM, Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at wrote: Hi Shaun! Your problem description is too short to give you any good help. Use tcpdump (or other tools) to capture the scenario with Asterisk and Kamailio. Then compare them to find out why it doesn't work. Is media sent directly to Asterisk then it ca not be the problem of Kamailio. I hope the mobile client is smart enough to also send a reINVITE when getting the new IP address (of the mobile connection) with proper Contact header - otherwise it can not receive SIP requests from Asterisk. regards Klaus On 20.06.2012 18:07, Shaun Clark wrote: The use case is that I have a SIP client registered to Kamailio talking to an Asterisk box connected to the PSTN. The client is a mobile phone and the user is connected to wifi. The user then steps out of wifi range and the phone drops the connection and picks up the 3g data connection. I want the media stream to reconnect to the client and the call to resume without having to redial. This works now if the client is directly connected to the Asterisk machine, but not when I am routing through my Kamailio server. How do I go about this, examples are always appreciated, thanks! _ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users --
Re: [SR-Users] Over 100% of CPU usage on kamailio
do you have heavy traffic on that instance? How many children have you configured? Yes, but only a voip. There are over 10k calls per day. fork=yes children=4 This configuration is working during 1.5 months without any issues till now... What you can do is to attach with gdb to a process using lot of CPU and do the backtrace: gdb /path/to/kamailio __pid__ Replace __pid__ with the PID of process eating the CPU. Then run bt It may be a deadlock/infinite loop somewhere. I saw three processes in top, others were down with no much cpu usage. Is the SIP routing going fine anyhow when CPU usage is high? It hard to say 'yes', when looking to call graphs (rrd) I saw that the calls throughput was slow down... Looking to logs/PCAP's/etc I'm seeing that some of calls were processed though (perhaps some of working forks). But there were 75-80% of failed calls with 478 Request Terminated. Also by analyzing a PCAP flows I saw that more than 70% of calls were de-jittered, de-sync'ed in RTP timestamps, etc... How do you solve it, by restart or it just appears from time to time and solves itself periodically? I did not restarted kamailio during 20 days, following a charts this problem was re-appeared during 2 days. After restart of kamailio -- I can't see any issues for now, there is 0.00 on CPU by all of kamailio instance/forks/etc... Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Over 100% of CPU usage on kamailio
Hello, when it occurs again, get the backtrace with gdb using the PIDs from all kamailio processes eating lot of CPU. Btw, what version are you using (kamailio -V)? Cheers, Daniel On 6/25/12 9:43 AM, Konstantin M. wrote: do you have heavy traffic on that instance? How many children have you configured? Yes, but only a voip. There are over 10k calls per day. fork=yes children=4 This configuration is working during 1.5 months without any issues till now... What you can do is to attach with gdb to a process using lot of CPU and do the backtrace: gdb /path/to/kamailio __pid__ Replace __pid__ with the PID of process eating the CPU. Then run bt It may be a deadlock/infinite loop somewhere. I saw three processes in top, others were down with no much cpu usage. Is the SIP routing going fine anyhow when CPU usage is high? It hard to say 'yes', when looking to call graphs (rrd) I saw that the calls throughput was slow down... Looking to logs/PCAP's/etc I'm seeing that some of calls were processed though (perhaps some of working forks). But there were 75-80% of failed calls with 478 Request Terminated. Also by analyzing a PCAP flows I saw that more than 70% of calls were de-jittered, de-sync'ed in RTP timestamps, etc... How do you solve it, by restart or it just appears from time to time and solves itself periodically? I did not restarted kamailio during 20 days, following a charts this problem was re-appeared during 2 days. After restart of kamailio -- I can't see any issues for now, there is 0.00 on CPU by all of kamailio instance/forks/etc... Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Over 100% of CPU usage on kamailio
Yes, I'll bt if it will appear again... Here is a version: # /opt/kamailio/sbin/kamailio -V version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 15:37:11 Jun 9 2012 with gcc 4.4.3 2012/6/25 Daniel-Constantin Mierla mico...@gmail.com Hello, when it occurs again, get the backtrace with gdb using the PIDs from all kamailio processes eating lot of CPU. Btw, what version are you using (kamailio -V)? Cheers, Daniel On 6/25/12 9:43 AM, Konstantin M. wrote: do you have heavy traffic on that instance? How many children have you configured? Yes, but only a voip. There are over 10k calls per day. fork=yes children=4 This configuration is working during 1.5 months without any issues till now... What you can do is to attach with gdb to a process using lot of CPU and do the backtrace: gdb /path/to/kamailio __pid__ Replace __pid__ with the PID of process eating the CPU. Then run bt It may be a deadlock/infinite loop somewhere. I saw three processes in top, others were down with no much cpu usage. Is the SIP routing going fine anyhow when CPU usage is high? It hard to say 'yes', when looking to call graphs (rrd) I saw that the calls throughput was slow down... Looking to logs/PCAP's/etc I'm seeing that some of calls were processed though (perhaps some of working forks). But there were 75-80% of failed calls with 478 Request Terminated. Also by analyzing a PCAP flows I saw that more than 70% of calls were de-jittered, de-sync'ed in RTP timestamps, etc... How do you solve it, by restart or it just appears from time to time and solves itself periodically? I did not restarted kamailio during 20 days, following a charts this problem was re-appeared during 2 days. After restart of kamailio -- I can't see any issues for now, there is 0.00 on CPU by all of kamailio instance/forks/etc... Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Unexpectedly Terminating
Hello, thanks for troubleshooting further -- as I expected, it is a memalign problem, but some confusing reports by not using always the patches I made to registrar module to align the structure, made thinking is something else. Now is no longer in registrar, but in another module. There might be many of these, can you try to compile first without strict alignment to 8 bytes? Checking quickly to gcc, the option should be |-mno-faster-structs So try: make proper make FLAVOUR=kamailio cfg| make Q=0 CC_EXTRA_OPTS=|-mno-faster-structs all| ... By providing Q=0, you will see all compile flags, verify that |-mno-faster-structs is there. Cheers, Daniel | On 6/22/12 3:26 PM, Akan wrote: Hello, After doing some research, this is what I found out. On Solaris Sparc 64bit system, there is a mandatory alignment of memory accesses and also for data types. I went thru the core dump, disassembled the code and located the instruction that produced the error. The registers addresses in questioned are on a 4 byte alignment but not an 8 byte or 16 byte alignment. .The earlier patch must have forced the alignment which is why the error did not occur in the program common.c. It looks like there is a similar situation in t_funcs.c. Here is a link that I found that can better explain: http://blog.jgc.org/2007/04/debugging-solaris-bus-error-caused-by.html Core was generated by `/opt/kamailio-3.2/sbin/kamailio'. Program terminated with signal 10, Bus error. #0 0x7bd2b7bc in t_relay_to (p_msg=0x10047c698, proxy=0x0, proto=0, replicate=0) at t_funcs.c:352 352 if (!t_reply( t, p_msg , 100 , 0x7bd2b7b0 +976: ldx [ %l7 + %g1 ], %g1 0x7bd2b7b4 +980: ldx [ %g1 ], %g1 0x7bd2b7b8 +984: call 0x7be9bc80 t_reply@plt *= 0x7bd2b7bc +988: ldx [ %g1 + 0x38 ], %o3* 0x7bd2b7c0 +992: cmp %o0, 0 0x7bd2b7c4 +996: be,pn %icc, 0x7bd2bb14 t_relay_to+1844 0x7bd2b7c8 +1000: ldx [ %fp + 0x7f7 ], %o0 0x7bd2b7cc +1004: b %xcc, 0x7bd2b460 t_relay_to+128 End of assembler dump. (gdb) info registers g1 g1 0x7666c3e4 -2308520988 (gdb) info registers o3 o3 0x18f3d0 1635280 I hope this helps in trying to resolve this problem and to find a solution. Also, if you need a sparc system to test with, just let me know. We can help from a sparc perspective. Thanks Nathaniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio v3.3.0 packages
Hello, I'm summarizing the status of packages for latest release v3.3.0. Debian/Ubuntu packages are ready in the usual APT repository, including nightly builds of the GIT branch 3.3, details at: * http://www.kamailio.org/wiki/packages/debs RPMs for CentOS/RedHat, Fedora, openSuse (various versions for each OS) are now available on opensuse build factory repository, details at: * http://www.kamailio.org/wiki/packages/rpms There you find the note about Peter Dunkley's repository, which has also builds for Raspberry Pi. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] default kamailio configuration
Hi All, After reading default kamailio configuration i can't understand why does kamailio remove preloaded route headers from the incoming initial INVITE before calling record_route(). remove_hf(Route); if (is_method(INVITE|SUBSCRIBE)) record_route(); What will happen if an INVITE comes from another SIP proxy which wants to stay in the middle of the dialogue ? For example, during Re-INVITE UA from our local site will construct in-dialog INVITE with R-URI taken from the remote contact and send it to our local proxy according to RR. Local proxy will process it in route[WITHNDLG] where loose_route() function will remove top most RR header because it points to us and since there is no more RR records forward the request to a destination from the R-URI. As i understand in that case remote proxy will not receive any in-dialog requests from us. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
Klaus / Daniel Thanks again for assistance with this. I've tried the solution based on add_contact_alias() and handle_ruri_alias() and it works perfectly. Richard On 22 June 2012 13:47, Klaus Darilion klaus.mailingli...@pernau.at wrote: On 22.06.2012 13:50, Richard Brady wrote: Thanks guys, fantastic answers. You mention that NAT detection happens before save() and the flag is set by lookup() which makes much more sense. However, if Kamailio is not the registrar, as is the case with my current project, those functions are not called, so an alternative is needed. There are clearly several options. The solution I have gone for is to replace fix_nated_register() with fix_nated_contact() so that the REGISTER request is relayed with a modified Contact header containing the external ip:port of the client. The cleanest solution would be to use add_contact_alias() and handle_ruri_alias(). The do not change the contact but put the public address into a uri parameter. Thus, the URI seen by the client is always the one it sends: http://www.kamailio.org/docs/modules/3.2.x/modules_k/nathelper.html#id2550431 That is then stored by the registrar (FreeSWITCH in my case) and used later to originate calls for that user. The FreeSWITCH know to send those calls to Kamailio through either use of the Path header and module in Kamailio, or through static configuration of fs_path or proxy parameters in FreeSWITCH. This is fine. The works for the first INVITE to the registered client behind NAT. But that client sends back a 200 OK with a Contact header containing its private IP address, and so fix_nated_contact() needs to be invoked on that response, and normally it would be due to FLB_NATB being set, but if Kamailio was not the registrar then that flag is not set. So I need to detect NAT on the client at the time of receiving the reply, or alternatively by having the registrar store a cookie and setting it based on that. You are correct. For in-dialog messages received from SIP clients I would always use add_contact_alias() and remove the NAT flags completely. I suppose then my next question then is can I call nat_uac_test() on a UAS? Today, almost any SIP clients are SIP symmetric. This means, that they receive SIP from the some port/connection where they send. Thus, skip the NAT tests completely and always use add_contact_alias() for messages receive from SIP clients and handle_ruri_alias() for messages sent to SIP clients. Depending on if you use Freeswitch also as media relay you can also remove the media proxy stuff from the Kamailio config. On important thing is NAT-keep-alive. This is usually done by nathelper module by querying the location table for contact with NAT-flag set. Thus, either you do NAT keep-alive by freeswitch (e.g sending OPTIONS requests) or do it in your Kamailio proxy (e.g. set the nat_bflag http://www.kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2541477 and call save(location,0x02) before relaying REGISTER to freeswitch. Then Kamailio will do NAT-pinging. Note, if you want to keep-alive only for successfully registered clients, you man want to call save() in the reply-route of the REGISTER request). regards Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
This is a great thread, really full of answers and concepts for me atleast. :) On Mon, Jun 25, 2012 at 5:57 PM, Richard Brady rnbr...@gmail.com wrote: Klaus / Daniel Thanks again for assistance with this. I've tried the solution based on add_contact_alias() and handle_ruri_alias() and it works perfectly. Richard On 22 June 2012 13:47, Klaus Darilion klaus.mailingli...@pernau.at wrote: On 22.06.2012 13:50, Richard Brady wrote: Thanks guys, fantastic answers. You mention that NAT detection happens before save() and the flag is set by lookup() which makes much more sense. However, if Kamailio is not the registrar, as is the case with my current project, those functions are not called, so an alternative is needed. There are clearly several options. The solution I have gone for is to replace fix_nated_register() with fix_nated_contact() so that the REGISTER request is relayed with a modified Contact header containing the external ip:port of the client. The cleanest solution would be to use add_contact_alias() and handle_ruri_alias(). The do not change the contact but put the public address into a uri parameter. Thus, the URI seen by the client is always the one it sends: http://www.kamailio.org/docs/modules/3.2.x/modules_k/nathelper.html#id2550431 That is then stored by the registrar (FreeSWITCH in my case) and used later to originate calls for that user. The FreeSWITCH know to send those calls to Kamailio through either use of the Path header and module in Kamailio, or through static configuration of fs_path or proxy parameters in FreeSWITCH. This is fine. The works for the first INVITE to the registered client behind NAT. But that client sends back a 200 OK with a Contact header containing its private IP address, and so fix_nated_contact() needs to be invoked on that response, and normally it would be due to FLB_NATB being set, but if Kamailio was not the registrar then that flag is not set. So I need to detect NAT on the client at the time of receiving the reply, or alternatively by having the registrar store a cookie and setting it based on that. You are correct. For in-dialog messages received from SIP clients I would always use add_contact_alias() and remove the NAT flags completely. I suppose then my next question then is can I call nat_uac_test() on a UAS? Today, almost any SIP clients are SIP symmetric. This means, that they receive SIP from the some port/connection where they send. Thus, skip the NAT tests completely and always use add_contact_alias() for messages receive from SIP clients and handle_ruri_alias() for messages sent to SIP clients. Depending on if you use Freeswitch also as media relay you can also remove the media proxy stuff from the Kamailio config. On important thing is NAT-keep-alive. This is usually done by nathelper module by querying the location table for contact with NAT-flag set. Thus, either you do NAT keep-alive by freeswitch (e.g sending OPTIONS requests) or do it in your Kamailio proxy (e.g. set the nat_bflag http://www.kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2541477 and call save(location,0x02) before relaying REGISTER to freeswitch. Then Kamailio will do NAT-pinging. Note, if you want to keep-alive only for successfully registered clients, you man want to call save() in the reply-route of the REGISTER request). regards Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Alias possible for presence...
Daniel, Thanks for the info, will dive into it together with an programmer that does the client.. Thanks. From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: woensdag 20 juni 2012 11:42 To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gertjan Wolzak Subject: Re: [SR-Users] Alias possible for presence... Hello, On 6/20/12 9:19 AM, Gertjan Wolzak wrote: Well, the account name is not public, but I would like to have the possibility to have a private account for registering and a publicaccount to publish. The goal is to have others to be able to send messages to your public id, while you log in with your private id. most of user agent have two fields, one to be used as SIP id and another one for authentication id. I know, think the next step is to get an communicator that has that possibility... So I want to be able to publish the public account and not the private one. For example: I register to the presence server with uid gertjan and my password, but I want to be known to others as newbie. So my friend can send an invitation/message to newbie as well as to gertjan . Just like the db_alias which is used in the kamalio for linking sip accounts to aliases. For presence, people have to subscribe to a SIP id, in this case it your public ID known by everybody. The easiest will be that the user agent will publish in behalf of the public id. But ultimately, you can replace the values in headers and xml body (see textops/xmlops functions), then do presence handling, after applying the changes to the message (see textopsx module). Cheers, Daniel Rgds, Gertjan From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: dinsdag 19 juni 2012 10:27 To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gertjan Wolzak Subject: Re: [SR-Users] Alias possible for presence... Hello, On 6/14/12 10:28 AM, Gertjan Wolzak wrote: Hello All, I have the following challenge. We are using kamailio 3.2 and have enabled presence. Which works fine. But we would like to be able to work with aliases, so that the account name and presentation name can be different. Any Ideas. Or did I miss the documentation on that? the account name is public or not? If it is some internal id, eventually is used for authentication, but for presence the contacts should know the public id. Or maybe you can give some example to understand how you use aliases in the context of presence... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- This message was scanned by Foize and is believed to be clean. Click here to report this message as spam. http://filter.foize.com/cgi-bin/learn-msg.cgi?id= ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- This message was scanned by Foize and is believed to be clean. Click here to report this message as spam. http://filter.foize.com/cgi-bin/learn-msg.cgi?id= ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Subscription-State: terminated;reason=terminated?
HI when I removed 102 from 101's contact list (using jitsi nightly 1.1 build), kamailio 3.3 send out NOTIFY to 102 like this: NOTIFY sip:102@192.168.122.147:5060;transport=udp;registering_acc=192_168_122_32 SIP/2.0. Via: SIP/2.0/UDP 192.168.122.32;branch=z9hG4bK1bfb.afbf0a85.0. To: sip:102@192.168.122.32;tag=f6a40771. From: sip:101@192.168.122.32;tag=a6a1c5f60faecf035a1ae5b6e96e979a-5724. CSeq: 4 NOTIFY. Call-ID: c7c52dd058268596ec84dd3c645a2f17@0.0.0.0. Content-Length: 0. User-Agent: kamailio (3.3.0-rc0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: sip:192.168.122.32:5060;transport=udp. Subscription-State: terminated;reason=terminated. - Note the reason code is:terminated. From rfc3265, The defined reason codes are: deactivated/ probation/rejected/ timeout/giveup/noresource What is the reason to send: reason=terminated instead one of the well-defined reason codes? There was a discussion regarding at: http://sip-router.org/tracker/index.php?do=detailstask_id=133 http://sip-router.org/tracker/index.php?do=detailstask_id=133 but I did not see the explaination of reason=terminated. Thanks min ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] EOF Error in pass_fd
Hello, there was a wrong unlocking of the profile while printing its content via MI, that could caused a race and eventually this kind of crash (in the case a dlg was added/removed in/from profile while the mi command was run). I committed a patch, you have to use latest branch 3.2. Let us know if all goes fine now. Cheers, Daniel On 6/25/12 4:45 PM, Ricardo Martinez wrote: Hello List. The last weekend our kamailio process crashed with this error : Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10661]: : core [pass_fd.c:293]: ERROR: receive_fd: EOF on 29 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: core [main.c:751]: child process 10601 exited by a signal 11 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: core [main.c:754]: core was generated Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: INFO: core [main.c:766]: INFO: terminating due to SIGCHLD Can someone tell me what seems to be the problem here? We're using : version: kamailio 3.2.2 (x86_64/linux) 98ba92-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 98ba92 -dirty compiled on 12:45:36 Feb 2 2012 with gcc 4.4.6 And this is the bt full from gdb Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 mi_profile_list (cmd_tree=value optimized out, param=value optimized out) at dlg_profile.c:789 789 if ( mi_print_dlg( rpl, ph-dlg, 0)!=0 ) Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt full #0 mi_profile_list (cmd_tree=value optimized out, param=value optimized out) at dlg_profile.c:789 node = value optimized out rpl_tree = 0x20a6790 rpl = 0x20a67b0 profile = 0x7fe99a2dda00 ph = 0x0 profile_name = value optimized out value = value optimized out i = value optimized out #1 0x7fe9dce287b1 in run_mi_cmd (fifo_stream=value optimized out) at ../../lib/kmi/mi.h:77 No locals. #2 mi_fifo_server (fifo_stream=value optimized out) at fifo_fnc.c:509 mi_cmd = 0x208f8f0 mi_rpl = value optimized out hdl = 0x0 line_len = 41 file_sep = 0x2039017 command = 0x7fe9dfdbdef9 profile_list_dlgs file = 0x7fe9dfd98d78 /tmp/openser_receiver_27809 f = 0x2039018 reply_stream = 0x20b0980 #3 0x7fe9dce2a640 in fifo_process (rank=value optimized out) at mi_fifo.c:247 fifo_stream = 0x209a0b0 #4 0x7fe9dce2a9de in mi_child_init (rank=0) at mi_fifo.c:211 pid = value optimized out #5 0x004dba41 in init_mod_child (m=0x7fe9dfc6ba10, rank=0) at sr_module.c:886 No locals. #6 0x004db9c4 in init_mod_child (m=0x7fe9dfc6bd08, rank=0) at sr_module.c:883 No locals. #7 0x004db9c4 in init_mod_child (m=0x7fe9dfc6c248, rank=0) at sr_module.c:883 No locals. #8 0x004db9c4 in init_mod_child (m=0x7fe9dfc6c748, rank=0) at sr_module.c:883 No locals. #9 0x004db9c4 in init_mod_child (m=0x7fe9dfc6d288, rank=0) at sr_module.c:883 No locals. #10 0x004db9c4 in init_mod_child (m=0x7fe9dfc6d928, rank=0) at sr_module.c:883 No locals. #11 0x004db9c4 in init_mod_child (m=0x7fe9dfc6dee0, rank=0) at sr_module.c:883 No locals. #12 0x004db9c4 in init_mod_child (m=0x7fe9dfc6e1f0, rank=0) at sr_module.c:883 No locals. #13 0x004db9c4 in init_mod_child (m=0x7fe9dfc6e5d0, rank=0) at sr_module.c:883 No locals. #14 0x004db9c4 in init_mod_child (m=0x7fe9dfc6e830, rank=0) at sr_module.c:883 No locals. #15 0x004db9c4 in init_mod_child (m=0x7fe9dfc6ebd0, rank=0) at sr_module.c:883 No locals. #16 0x004db9c4 in init_mod_child (m=0x7fe9dfc6f220, rank=0) at sr_module.c:883 No locals. #17 0x004db9c4 in init_mod_child (m=0x7fe9dfc6f648, rank=0) at sr_module.c:883 No locals. #18 0x004db9c4 in init_mod_child (m=0x7fe9dfc6fa60, rank=0) at sr_module.c:883 No locals. #19 0x004db9c4 in init_mod_child (m=0x7fe9dfc6fe40, rank=0) at sr_module.c:883 No locals. #20 0x004db9c4 in init_mod_child (m=0x7fe9dfc702c0, rank=0) at sr_module.c:883 No
Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
On 6/25/12 3:34 PM, SamyGo wrote: This is a great thread, really full of answers and concepts for me atleast. :) looks like we will have a new wiki page with the digested content of this thread :-) Cheers, Daniel On Mon, Jun 25, 2012 at 5:57 PM, Richard Brady rnbr...@gmail.com mailto:rnbr...@gmail.com wrote: Klaus / Daniel Thanks again for assistance with this. I've tried the solution based on add_contact_alias() and handle_ruri_alias() and it works perfectly. Richard On 22 June 2012 13:47, Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at wrote: On 22.06.2012 13:50, Richard Brady wrote: Thanks guys, fantastic answers. You mention that NAT detection happens before save() and the flag is set by lookup() which makes much more sense. However, if Kamailio is not the registrar, as is the case with my current project, those functions are not called, so an alternative is needed. There are clearly several options. The solution I have gone for is to replace fix_nated_register() with fix_nated_contact() so that the REGISTER request is relayed with a modified Contact header containing the external ip:port of the client. The cleanest solution would be to use add_contact_alias() and handle_ruri_alias(). The do not change the contact but put the public address into a uri parameter. Thus, the URI seen by the client is always the one it sends: http://www.kamailio.org/docs/modules/3.2.x/modules_k/nathelper.html#id2550431 That is then stored by the registrar (FreeSWITCH in my case) and used later to originate calls for that user. The FreeSWITCH know to send those calls to Kamailio through either use of the Path header and module in Kamailio, or through static configuration of fs_path or proxy parameters in FreeSWITCH. This is fine. The works for the first INVITE to the registered client behind NAT. But that client sends back a 200 OK with a Contact header containing its private IP address, and so fix_nated_contact() needs to be invoked on that response, and normally it would be due to FLB_NATB being set, but if Kamailio was not the registrar then that flag is not set. So I need to detect NAT on the client at the time of receiving the reply, or alternatively by having the registrar store a cookie and setting it based on that. You are correct. For in-dialog messages received from SIP clients I would always use add_contact_alias() and remove the NAT flags completely. I suppose then my next question then is can I call nat_uac_test() on a UAS? Today, almost any SIP clients are SIP symmetric. This means, that they receive SIP from the some port/connection where they send. Thus, skip the NAT tests completely and always use add_contact_alias() for messages receive from SIP clients and handle_ruri_alias() for messages sent to SIP clients. Depending on if you use Freeswitch also as media relay you can also remove the media proxy stuff from the Kamailio config. On important thing is NAT-keep-alive. This is usually done by nathelper module by querying the location table for contact with NAT-flag set. Thus, either you do NAT keep-alive by freeswitch (e.g sending OPTIONS requests) or do it in your Kamailio proxy (e.g. set the nat_bflag http://www.kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2541477 and call save(location,0x02) before relaying REGISTER to freeswitch. Then Kamailio will do NAT-pinging. Note, if you want to keep-alive only for successfully registered clients, you man want to call save() in the reply-route of the REGISTER request). regards Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org
Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
On Jun 25, 2012, at 8:57 AM, Richard Brady wrote: Klaus / Daniel Thanks again for assistance with this. I've tried the solution based on add_contact_alias() and handle_ruri_alias() and it works perfectly. Richard Do you have an example of the cfg you can share? Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] EOF Error in pass_fd
Thanks Daniel. We are going to do some test with the new patch, but before can you guide me on how to download only the modification via git ¿ Thanks!!. Best Regards, Ricardo Martinez.- *De:* Daniel-Constantin Mierla [mailto:mico...@gmail.com] *Enviado el:* lunes, 25 de junio de 2012 12:43 *Para:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List *CC:* Ricardo Martinez *Asunto:* Re: [SR-Users] EOF Error in pass_fd Hello, there was a wrong unlocking of the profile while printing its content via MI, that could caused a race and eventually this kind of crash (in the case a dlg was added/removed in/from profile while the mi command was run). I committed a patch, you have to use latest branch 3.2. Let us know if all goes fine now. Cheers, Daniel On 6/25/12 4:45 PM, Ricardo Martinez wrote: Hello List. The last weekend our kamailio process crashed with this error : Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10661]: : core [pass_fd.c:293]: ERROR: receive_fd: EOF on 29 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: core [main.c:751]: child process 10601 exited by a signal 11 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: core [main.c:754]: core was generated Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: INFO: core [main.c:766]: INFO: terminating due to SIGCHLD Can someone tell me what seems to be the problem here? We’re using : version: kamailio 3.2.2 (x86_64/linux) 98ba92-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 98ba92 -dirty compiled on 12:45:36 Feb 2 2012 with gcc 4.4.6 And this is the bt full from gdb Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 mi_profile_list (cmd_tree=value optimized out, param=value optimized out) at dlg_profile.c:789 789 if ( mi_print_dlg( rpl, ph-dlg, 0)!=0 ) Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt full #0 mi_profile_list (cmd_tree=value optimized out, param=value optimized out) at dlg_profile.c:789 node = value optimized out rpl_tree = 0x20a6790 rpl = 0x20a67b0 profile = 0x7fe99a2dda00 ph = 0x0 profile_name = value optimized out value = value optimized out i = value optimized out #1 0x7fe9dce287b1 in run_mi_cmd (fifo_stream=value optimized out) at ../../lib/kmi/mi.h:77 No locals. #2 mi_fifo_server (fifo_stream=value optimized out) at fifo_fnc.c:509 mi_cmd = 0x208f8f0 mi_rpl = value optimized out hdl = 0x0 line_len = 41 file_sep = 0x2039017 command = 0x7fe9dfdbdef9 profile_list_dlgs file = 0x7fe9dfd98d78 /tmp/openser_receiver_27809 f = 0x2039018 reply_stream = 0x20b0980 #3 0x7fe9dce2a640 in fifo_process (rank=value optimized out) at mi_fifo.c:247 fifo_stream = 0x209a0b0 #4 0x7fe9dce2a9de in mi_child_init (rank=0) at mi_fifo.c:211 pid = value optimized out #5 0x004dba41 in init_mod_child (m=0x7fe9dfc6ba10, rank=0) at sr_module.c:886 No locals. #6 0x004db9c4 in init_mod_child (m=0x7fe9dfc6bd08, rank=0) at sr_module.c:883 No locals. #7 0x004db9c4 in init_mod_child (m=0x7fe9dfc6c248, rank=0) at sr_module.c:883 No locals. #8 0x004db9c4 in init_mod_child (m=0x7fe9dfc6c748, rank=0) at sr_module.c:883 No locals. #9 0x004db9c4 in init_mod_child (m=0x7fe9dfc6d288, rank=0) at sr_module.c:883 No locals. #10 0x004db9c4 in init_mod_child (m=0x7fe9dfc6d928, rank=0) at sr_module.c:883 No locals. #11 0x004db9c4 in init_mod_child (m=0x7fe9dfc6dee0, rank=0) at sr_module.c:883 No locals. #12 0x004db9c4 in init_mod_child (m=0x7fe9dfc6e1f0, rank=0) at sr_module.c:883 No locals. #13 0x004db9c4 in init_mod_child (m=0x7fe9dfc6e5d0, rank=0) at sr_module.c:883 No locals. #14 0x004db9c4 in init_mod_child (m=0x7fe9dfc6e830, rank=0) at sr_module.c:883 No locals. #15 0x004db9c4 in init_mod_child (m=0x7fe9dfc6ebd0, rank=0) at sr_module.c:883 No locals. #16 0x004db9c4 in init_mod_child
Re: [SR-Users] Subscription-State: terminated;reason=terminated?
HI I did more analysis: as before, the configure is: 101 --- kamailio proxy/xcap server -- 102 101/102 : jitsi night build, xcap/SIMPLE mode kamailio is 3.3 101 has 102 on its contacts list, and 102 has 101 on its contacts list as well now 101 remove 102 from its contact, proxy will send out NOTIFY to 102 (1) if reason=terminated returned in NOTIFY ( this is the current kamaili behavior) According to RFC 3265: If no reason code or an unknown reason code is present, the client MAY attempt to re- subscribe at any time (unless a retry-after parameter is present, in which case the client SHOULD NOT attempt re-subscription until after the number of seconds specified by the retry-after parameter). Jitsi 1.1 nightly will keep on re-subscribe ( at some random time ). And kamailio/proxy will keep on reject the subscribe with: NOTIFY, with reason=terminated. It seems to waste some resources (bandwidth/db/cpu etc). Image if there are a lot of deleted contacts :(. (2) if reason=rejected returned in NOTIFY according to the same RFC: rejected: The subscription has been terminated due to change in authorization policy. Clients SHOULD NOT attempt to re-subscribe. The retry-after parameter has no semantics for rejected. So the client will not send any re-subscribe, which is good, will save some resources. But there is an issue: when 101 add 102 again, after 101 puting pres-rules (allow 102) to the xcap server , there will be two cases: (2.a). If that subscription expired, or deleted by the kamailio timer ( I hope I understand the code correctly) of course kamailio will not send any NOTIFY (to 102). (2.b). if that subscription do still exist in active_watcher, that subscriptions will be marked as active kamailio will send the NOTIFY to 102 indicating 101's status But from 102 point of view: since the subscription has been terminated , this notify will be rejected as 481 non-exist. In neither case, can 102 see 101's status ( since 102 to 101's subscription has been rejected/terminated from 102 point of view) So from pure end-user point of view, that is not the expected behavior. User expect the 102 can see 101's status since 101 now allow 102 again and 102 did not remove 101 from his contact list The question is: how can we do it right? Thanks. min On 06/25/2012 05:43 PM, Min Wang wrote: HI when I removed 102 from 101's contact list (using jitsi nightly 1.1 build), kamailio 3.3 send out NOTIFY to 102 like this: NOTIFY sip:102@192.168.122.147:5060;transport=udp;registering_acc=192_168_122_32 SIP/2.0. Via: SIP/2.0/UDP 192.168.122.32;branch=z9hG4bK1bfb.afbf0a85.0. To: sip:102@192.168.122.32;tag=f6a40771. From: sip:101@192.168.122.32;tag=a6a1c5f60faecf035a1ae5b6e96e979a-5724. CSeq: 4 NOTIFY. Call-ID: c7c52dd058268596ec84dd3c645a2f17@0.0.0.0. Content-Length: 0. User-Agent: kamailio (3.3.0-rc0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: sip:192.168.122.32:5060;transport=udp. Subscription-State: terminated;reason=terminated. - Note the reason code is:terminated. From rfc3265, The defined reason codes are: deactivated/ probation/rejected/ timeout/giveup/noresource What is the reason to send: reason=terminated instead one of the well-defined reason codes? There was a discussion regarding at: http://sip-router.org/tracker/index.php?do=detailstask_id=133 http://sip-router.org/tracker/index.php?do=detailstask_id=133 but I did not see the explaination of reason=terminated. Thanks min ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
I had this thread already starred here in inbox, a wiki page is definitely going to be bookmarked. On Mon, Jun 25, 2012 at 9:44 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: On 6/25/12 3:34 PM, SamyGo wrote: This is a great thread, really full of answers and concepts for me atleast. :) looks like we will have a new wiki page with the digested content of this thread :-) Cheers, Daniel On Mon, Jun 25, 2012 at 5:57 PM, Richard Brady rnbr...@gmail.com wrote: Klaus / Daniel Thanks again for assistance with this. I've tried the solution based on add_contact_alias() and handle_ruri_alias() and it works perfectly. Richard On 22 June 2012 13:47, Klaus Darilion klaus.mailingli...@pernau.at wrote: On 22.06.2012 13:50, Richard Brady wrote: Thanks guys, fantastic answers. You mention that NAT detection happens before save() and the flag is set by lookup() which makes much more sense. However, if Kamailio is not the registrar, as is the case with my current project, those functions are not called, so an alternative is needed. There are clearly several options. The solution I have gone for is to replace fix_nated_register() with fix_nated_contact() so that the REGISTER request is relayed with a modified Contact header containing the external ip:port of the client. The cleanest solution would be to use add_contact_alias() and handle_ruri_alias(). The do not change the contact but put the public address into a uri parameter. Thus, the URI seen by the client is always the one it sends: http://www.kamailio.org/docs/modules/3.2.x/modules_k/nathelper.html#id2550431 That is then stored by the registrar (FreeSWITCH in my case) and used later to originate calls for that user. The FreeSWITCH know to send those calls to Kamailio through either use of the Path header and module in Kamailio, or through static configuration of fs_path or proxy parameters in FreeSWITCH. This is fine. The works for the first INVITE to the registered client behind NAT. But that client sends back a 200 OK with a Contact header containing its private IP address, and so fix_nated_contact() needs to be invoked on that response, and normally it would be due to FLB_NATB being set, but if Kamailio was not the registrar then that flag is not set. So I need to detect NAT on the client at the time of receiving the reply, or alternatively by having the registrar store a cookie and setting it based on that. You are correct. For in-dialog messages received from SIP clients I would always use add_contact_alias() and remove the NAT flags completely. I suppose then my next question then is can I call nat_uac_test() on a UAS? Today, almost any SIP clients are SIP symmetric. This means, that they receive SIP from the some port/connection where they send. Thus, skip the NAT tests completely and always use add_contact_alias() for messages receive from SIP clients and handle_ruri_alias() for messages sent to SIP clients. Depending on if you use Freeswitch also as media relay you can also remove the media proxy stuff from the Kamailio config. On important thing is NAT-keep-alive. This is usually done by nathelper module by querying the location table for contact with NAT-flag set. Thus, either you do NAT keep-alive by freeswitch (e.g sending OPTIONS requests) or do it in your Kamailio proxy (e.g. set the nat_bflag http://www.kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2541477 and call save(location,0x02) before relaying REGISTER to freeswitch. Then Kamailio will do NAT-pinging. Note, if you want to keep-alive only for successfully registered clients, you man want to call save() in the reply-route of the REGISTER request). regards Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Git Error on Kamailio-3.3 Install
Hello, I'm setting up new Kamailio 3.3 from git ref-link: http://www.kamailio.org/wiki/install/3.3.x/git I'm using CentOS 5.8 and when I execute the line # git checkout -b 3.3 origin/3.3 it gives me error. fatal: git checkout: updating paths is incompatible with switching branches. Did you intend to checkout 'origin/3.3' which can not be resolved as commit? Can I skip it ! o need to do something about it !! Regards, Sammy G. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
