sr-users June 2020

sr-users@lists.kamailio.org

71 participants
104 discussions

by BALL SUN

Hi I have a setup using Kamailio IMS, and when trying to make a SIP call between clients, S-CSCF returns " SIP/2.0 500 Server error on LIR select next S-CSCF", do you have any idea on how to fix? - RBK

3 years, 10 months

kamailio - 5.3.2 - Possible memleak in Core Module

by José Seabra

Hello there. During my stress tests against our kamailio servers I detected that it was running out of shm memory. edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:298]: qm_find_free(): qm_find_free(0x7ff370920000, 4224); Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:432]: qm_malloc(): qm_malloc(0x7ff370920000, 4224) called from core: core/sip_msg_clone.c: sip_msg_shm_clone(496), module: core; Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/sip_msg_clone.c:499]: sip_msg_shm_clone(): could not allocate shared memory from shm pool edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1293]: new_t(): out of mem: edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1439]: t_newtran(): new_t failed edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) The output of kamctl stats shmem confirms that kamailio is running out of memory: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 36427", "shmem:free_size = 14123552", "shmem:max_used_size = 268435456", "shmem:real_used_size = 254311904", "shmem:total_size = 268435456", "shmem:used_size = 215787904" ], "id": 10901 } Then I was check what was the module that were consuming more memory with kamcmd mod.stats all shm and the output of this command showed me that Core module was consuming most of shm memory, as you can see bellow is the output of this command shows that Core - build_req_buf_from_sip_req is consuming most of shm. Module: core { sip_msg_shm_clone(496): 4872856 create_avp(175): 24768 msg_lump_cloner(986): 154376 xavp_new_value(106): 780896 build_req_buf_from_sip_req(2187): 183984624 counters_prefork_init(211): 36864 cfg_clone_str(130): 96 cfg_shmize(217): 848 main_loop(1303): 8 init_pt(106): 8 init_pt(105): 8 init_pt(104): 5256 register_timer(995): 232 cfg_register_ctx(47): 64 init_tcp(5021): 8192 init_tcp(5015): 32768 init_tcp(5007): 8 init_tcp(5000): 8 init_tcp(4993): 8 init_tcp(4987): 8 init_tcp(4975): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 timer_alloc(498): 96 init_dns_cache(361): 8 init_dns_cache(352): 16384 init_dns_cache(344): 16 init_dns_cache(336): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 qm_shm_lock_init(1202): 8 Total: 190229920 } Then I stopped the sipp that was running against kamailio in order to see if the memory consumed by *build_req_buf_from_sip_req* decreased but it didn't, so, it seems that there is a memleak in this version. If the information that I'm sending in this email is not enough to identify the root cause of this, please let me know if there is anything else that I can do here to help identify better what is the main reason. Than you Best Regards -- Cumprimentos José Seabra

3 years, 10 months

siptrace: methods blacklist modparam

by Nuno Ferreira

Hello, We are a heavy Homer/Hepic (tks QXIP team) users and on our setup, there are some SIP methods we don’t want to get into the database or send to the capture server because there’s no value-added for call troubleshooting purposes (OPTIONS, KDMQ, HTTP (from JSONRPC)…) Typically we achieve the message filtering using siptrace in “manual” mode and control what to send from the script. This works fine for some situations, but there are a few we need to have the automatic mode on: - topos enabled: using manual mode we don’t get the “wire” version of the signaling - TCP connections: there are a few situations where the destination port is traced with 0 - I believe this is due to the fact the TCP socket not established when the message is collected There are other technics to filter like using a HEP relay proxy dropping unwanted traffic, but having a native kamailio functionality would make things much easier. With that in mind, I’ve created a PR (#2374 <https://github.com/kamailio/kamailio/pull/2374>) that allows to blacklist methods we don’t want to send to the capture server when using automatic trace mode. I believe this is a very helpful change for the community as well. To block SIP OPTIONS and KDMQ we just need to declare a new modparam: modparam("siptrace", "trace_on", 1) modparam("siptrace", "methods_blacklist_auto", 16896) Let me know what you think about it. Nuno -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 years, 10 months

Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

by supreeth herle

Please try to keep the mailing list in loop so that it helps others as well facing this issue. I believe this was resolved in 5.3.3 branch not sure why you are facing this issue. Anyways, make sure that modules loaded in kamailio.cfg of PCSCF is in following order loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif loadmodule "ims_registrar_pcscf" Best Regards, Supreeth On Wed, 24 Jun 2020 at 13:36, Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi Supreeth, > > I am getting the below error > > ms_usrloc_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_usrloc_pcscf.so] > 0(455) DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: > mod_init(): Successfully bound to PCSCF Usrloc module > 0(455) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): > found export of <bind_ims_ipsec_pcscf> in module ims_ipsec_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_ipsec_pcscf.so] > 0(455) ERROR: ims_ipsec_pcscf [cmd.c:85]: bind_ipsec_pcscf(): > configuration error - trying to bind to ipsec pscscf module before being > initialized > 0(455) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while > initializing module ims_registrar_pcscf > (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_registrar_pcscf.so) > ERROR: error while initializing modules > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:305]: mod_destroy(): > Error destroying spi generator > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): > Error destroying port generator > 0(455) DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start > 0(455) DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table > 0(455) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores > 0(455) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists > > > Kindly help > > On Wed, Jun 24, 2020 at 5:03 PM supreeth herle <herlesupreeth(a)gmail.com> > wrote: > >> There is no other IPSec configuration needed apart from running Kamailio >> P-CSCF and UE sending a SIP Register with Security-Client information for >> IPSec creation. Kamailio IMS IPSec module creates IPSec tunnels using >> Netlink library. >> >> Best Regards, >> Supreeth >> >> On Wed, 24 Jun 2020 at 13:29, Pavithra Mohanraja <pavimohan3004(a)gmail.com> >> wrote: >> >>> Hi , >>> >>> Could you please tell how to configure ipsec (general configuration not >>> related to kamailio). >>> >>> Thanks, >>> Pavithra >>> >>> On Wed, Jun 24, 2020 at 2:12 PM supreeth herle <herlesupreeth(a)gmail.com> >>> wrote: >>> >>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>> >>>> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or >>>> any other HSS you are using to enter IMS user details). SCSCF is not able >>>> to find a registered user with that IMPU (i am assuming that you are >>>> getting this response while calling) >>>> >>>> Increase the debug level on S-CSCF and check out which IMPU is being >>>> referred by SCSCF >>>> >>>> Best Regards, >>>> Supreeth >>>> >>>> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja < >>>> pavimohan3004(a)gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I am facing below error in kamailio-ims 5.3.4 version. >>>>> >>>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>>> could you please help me how should i proceed with and where the >>>>> registration details will be stored in kamailio ims. >>>>> >>>>> It fails at this line in kamailio.cfg file in scscf . >>>>> >>>>> route[FINAL_TERM] { >>>>> if (lookup("location")) { >>>>> if (uri==myself) { >>>>> if (!t_newtran()) { >>>>> sl_reply_error(); >>>>> exit; >>>>> } >>>>> xlog("Destination Not Found ----\n"); >>>>> t_reply("404","Not Found - destination user >>>>> not found on this S-CSCF"); >>>>> exit; >>>>> } >>>>> >>>>> here lookup(location) means what it means. Could you please help me >>>>> >>>>> >>>>> >>>>> Thanks, >>>>> Pavithra >>>>> >>>>> On Thu, May 14, 2020 at 9:59 PM supreeth herle < >>>>> herlesupreeth(a)gmail.com> wrote: >>>>> >>>>>> Hello Pavithra, >>>>>> >>>>>> I have written a guide for how to integrate an IMS using Kamailio >>>>>> with open5gs EPC, here is the link >>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can >>>>>> find steps to setup rtpengine in that guide. >>>>>> >>>>>> Regarding requirement of rtpengine, its mandatory if you want to use >>>>>> the scripts in that example folder without modification. Also, read the >>>>>> READE of sipwise/rtpengine github repo, its very informative (in a gist >>>>>> rtpengine offer much more feature than rtpproxy) >>>>>> >>>>>> why can't we have sip client to sip client rtps flowing like how it >>>>>>> is happening without NAT. >>>>>>> >>>>>> >>>>>> For this, i would definitely suggest to read about SIP working. This >>>>>> is a perfect link https://github.com/onmyway133/blog/issues/284 >>>>>> which offers extensive material (short and very interesting to read with >>>>>> examples) related to SIP. >>>>>> >>>>>> Best Regards, >>>>>> Supreeth >>>>>> >>>>>> >>>>>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja < >>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>> >>>>>>> Hi , >>>>>>> >>>>>>> I didn't configure rtpengine ..instead of that only I used >>>>>>> rrpproxy..will it not work? >>>>>>> >>>>>>> If not please tell me how to configure rtpengine and also tell me if >>>>>>> it is really madatory .. why can't we have sip client to sip client rtps >>>>>>> flowing like how it is happening without NAT. >>>>>>> >>>>>>> That's my doubt ..please clarify .. >>>>>>> >>>>>>> Thanks. >>>>>>> >>>>>>> On Thu, May 14, 2020, 8:57 PM supreeth herle < >>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>> >>>>>>>> HI, >>>>>>>> >>>>>>>> Hold on a moment, >>>>>>>> >>>>>>>> >>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>> 10.40.10.30 - hss >>>>>>>>> >>>>>>>> >>>>>>>> So i assume you are using the examples provided in the ims folder >>>>>>>> of kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have >>>>>>>> you modified the P-CSCF configuration to work with RTPProxy? >>>>>>>> >>>>>>>> Best regards, >>>>>>>> Supreeth >>>>>>>> >>>>>>>> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja < >>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi , >>>>>>>>> I have attached working rtp pcap file. But it is getting reverted >>>>>>>>> back to me . >>>>>>>>> >>>>>>>>> As you said , i have tried restarting rtpproxy several times. but >>>>>>>>> it didnt work . >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, May 14, 2020 at 7:11 PM supreeth herle < >>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Thanks for the pcap. Sorry, to ask this but can you also send >>>>>>>>>> pcap of scenario where you see RTP packets? >>>>>>>>>> >>>>>>>>>> Also, did you restart the rtpproxy after altering the >>>>>>>>>> /etc/default/rtpproxy ? >>>>>>>>>> >>>>>>>>>> i.e. >>>>>>>>>> >>>>>>>>>> $ systemctl restart rtpproxy >>>>>>>>>> >>>>>>>>>> Best Regards, >>>>>>>>>> Supreeth >>>>>>>>>> >>>>>>>>>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja < >>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hi Supreeth, >>>>>>>>>>> I have attached pcap files . kindly check and let me know. >>>>>>>>>>> filter of rtp packets and check. >>>>>>>>>>> >>>>>>>>>>> *filter : sip || rtp* >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Thu, May 14, 2020 at 5:15 PM supreeth herle < >>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi Pavithra, >>>>>>>>>>>> >>>>>>>>>>>> Thanks for the clarification. Can you please attach a pcap of >>>>>>>>>>>> the call where you are not seeing the RTP packets? And, btw is the call >>>>>>>>>>>> getting dropped or something in that scenario? >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> Supreeth >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Thu, 14 May 2020 at 13:34, Pavithra Mohanraja < >>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi Supreeth, >>>>>>>>>>>>> >>>>>>>>>>>>> I have kamailio ims installed in 4 vms inside openstack. >>>>>>>>>>>>> >>>>>>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>>>>>> 10.40.10.30 - hss >>>>>>>>>>>>> >>>>>>>>>>>>> so all these vms are running in same subnet. >>>>>>>>>>>>> I have a floating ip associated to pcscf. >>>>>>>>>>>>> Now I have two zoiper clients outside openstack >>>>>>>>>>>>> zoiper1 - 10.0.2.15 (NAT Enabled Network adaptor) >>>>>>>>>>>>> zoiper2 - 10.0.2.15 >>>>>>>>>>>>> >>>>>>>>>>>>> when i am making a call from zoiper1 to zoiper2, call is going >>>>>>>>>>>>> properly . SIP messages are coming properly in wireshark packets. >>>>>>>>>>>>> when i filter for RTP packets in wireshark, it is not coming. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> when i have the zoiper clients inside openstack itself ,i can >>>>>>>>>>>>> able to see RTP Packets in wireshark between two sip clients without any >>>>>>>>>>>>> NAT related configurations done. >>>>>>>>>>>>> >>>>>>>>>>>>> Kindly help me. >>>>>>>>>>>>> Thanks. >>>>>>>>>>>>> >>>>>>>>>>>>> On Thu, May 14, 2020 at 4:49 PM supreeth herle < >>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Can you explain some more when you mean by "RTP alone is not >>>>>>>>>>>>>> coming."? And, how are you checking/verifying that? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:12, Pavithra Mohanraja < >>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Yes I have configured as you mentioned only . In my >>>>>>>>>>>>>>> pcscf only i have rtpproxy also configured. >>>>>>>>>>>>>>> Floating IP i have given in listen address also as >>>>>>>>>>>>>>> listen = 10.40.10.3:4060 advertise 10.45.4.22:4060 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 10.40.10.3 - private ip >>>>>>>>>>>>>>> 10.45.4.22 - floating ip >>>>>>>>>>>>>>> Still RTP alone is not coming. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Is it the only way of configuring NAT ? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Thu, May 14, 2020 at 4:37 PM supreeth herle < >>>>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hello Pavithra, >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Have you tried advertising the floating IP of the VM >>>>>>>>>>>>>>>> running RTPProxy as follows? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Edit /etc/default/rtpproxy file as follows: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> # Additional options that are passed to the daemon. >>>>>>>>>>>>>>>> EXTRA_OPTS="-l 172.24.15.30 -d DBUG:LOG_LOCAL0" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> here, "-l <PUBLIC_IP>" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hope it helps. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:01, Pavithra Mohanraja < >>>>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Could anybody please give me the approach how to configure >>>>>>>>>>>>>>>>> NAT in kamailio IMS. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I have installed kamailio IMS in openstack environment >>>>>>>>>>>>>>>>> which has private ips for all four vms (each component in one vm). >>>>>>>>>>>>>>>>> pcscf has one floating IP attached. I have enabled "WITH_NAT" and >>>>>>>>>>>>>>>>> configured rtpproxy . >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> when i make a call from outside of openstack using 2 >>>>>>>>>>>>>>>>> zoiper clients with outbound proxy keeping floating ip of pcscf, I am >>>>>>>>>>>>>>>>> getting 200 ok with all SIP packets correctly flowing but there is no RTP >>>>>>>>>>>>>>>>> Packet. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> When i try the same within openstack environment without >>>>>>>>>>>>>>>>> using public ip, I am getting RTP Packets. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly tell me the procedure of NAT Enablement or do i >>>>>>>>>>>>>>>>> need to do anything other than this. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly help because i am stuck with this for long time. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>> >>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>> >>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>> >>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> Kamailio (SER) - Users Mailing List >>>>>>> sr-users(a)lists.kamailio.org >>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>

3 years, 10 months

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)

by Martin W Woscek

Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

3 years, 10 months

Kamailio IMS using returned SRV records for transport slelection

by BALL SUN

Hi All I am new to Kamailio IMS, and I just setup a testing environment where having P-CSCF, I-CSCF and S-CSCF. I would like to use make below works 1. SIP client <-> P-CSCF (either UDP or TCP, based on the transport tag) 2. P-CSCF <-> I-CSCF (TCP by default) 3. I-CSCF <-> S-CSCF (TCP by default) For point 2, I observed that P-CSCF actually will send DNS SRV query to DNS server for resolving _sip._udp. and _sip._tcp. I believed we can just disable the "_sip._udp." so that P-CSCF will always use TCP to communication wih I-CSCF. However, for point 3, I didnt notice I-CSCF send DNS SRV query, My question is how we can force I-CSCF to do that or is there any configuration that we should enable such that I-CSCF always use TCP to S-CSCF? - RBK

3 years, 10 months

SECFILTER reload from Siremis

by Social Boh

Hello list, is posible to reload SECFILTER data directly from Siremis? or the only way is: kamcmd secfilter.reload Regards -- --- I'm SoCIaL, MayBe

3 years, 10 months

dialog's track_cseq_updates seems to break uac_auth() in this configuration

by George Diamantopoulos

Hello again, This might turn out to be a bug, but I'm posting here first because my config is fairly complicated and it might not be the sanest it could be. The problem manifests as follows: with the sole change being the enabling of track_cseq_updates for the dialog module, kamailio fails to process the 100 Trying it receives following authentication initiated by the uac_auth() function. Without the track_cseq_updates option enabled, the call proceeds successfully, albeit with the CSeq for the auth-carrying INVITE having the same value as the original. Here's a simplified version of the configuration routes involved in this: request_route { ... route(DISPATCH_PROVIDER); ... } route[DISPATCH_PROVIDER] { if ( !ds_select_dst("10", "8") ) { t_send_reply("503", "Downstream carrier unavailable"); exit; } t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_branch("PROVIDER_BRANCH"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } failure_route[DISPATCH_PROVIDER_FAILURE] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(arealm) = "authrealm.com"; $avp(auser) = "authusername"; $avp(apass) = "verys3kr1t"; uac_auth(); t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll explain why this is here later in this e-mail t_relay(); exit; } } branch_route[PROVIDER_BRANCH] { uac_replace_from("$dlg_var(my_new_from)"); uac_replace_to("$dlg_var(my_new_to)"); $rU = "my new RURI user"; } event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { return; # next DST - only for 5xx or local timeout } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout() && !t_branch_replied()) ) { if ( ds_next_dst() ) { t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } else { xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER: Failed to route request to PROVIDER! Giving Up. Negative response will be sent upstream.\n"); } } } One will rightfully wonder why use both branch-failure routes and failure_route to handle things. Well, it's not obvious from this because the relevant parts are removed for brevity, but I'm generally using branch failure event routes to try other destinations in the same destination set, and failure routes to additionally try other downstream "providers" in case all entries is the current destination set have failed. I don't think these actions/routes are relevant because the problem manifests without any of these failover mechanisms (of switching over to the next provider) engaging. Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);" in the failure_route, which might seem a little odd, this was added because without it, uac_auth() will send the auth-carrying INVITE always to the first destination in the destination set even if ds_next_dst has been called from the branch-failure event route. Unfortunately I haven't been able to determine if this actually helped, because I have only received a 5xx error once from the downstream peer before the "fix" and haven't been able to reproduce this since then. But that's another issue unrelated to the one I'm asking about here. So what happens with track_cseq_updates enabled is, after uac_auth() successfully sends the authenticated INVITE and the peer starts sending provisional responses, kamailio doesn't seem to acknowledge them. Instead, it will retransmit the auth INVITE as if there was a firewall preventing the 1xx responses from being admitted. Then, the timeout will engage and the configuration script will behave as if t_branch_timeout returns true (it will do ds_next_dst). So I was wondering if this is to be expected with this configuration, or if this should be reported as a bug. Thanks! Best regards, George

3 years, 10 months

Kamailio-5.3.3. NAT and RTP not working

by Pavithra Mohanraja

Hi , Could anybody please give me the approach how to configure NAT in kamailio IMS. I have installed kamailio IMS in openstack environment which has private ips for all four vms (each component in one vm). pcscf has one floating IP attached. I have enabled "WITH_NAT" and configured rtpproxy . when i make a call from outside of openstack using 2 zoiper clients with outbound proxy keeping floating ip of pcscf, I am getting 200 ok with all SIP packets correctly flowing but there is no RTP Packet. When i try the same within openstack environment without using public ip, I am getting RTP Packets. Kindly tell me the procedure of NAT Enablement or do i need to do anything other than this. Kindly help because i am stuck with this for long time. Thanks.

3 years, 10 months

multiple domains with microsoft team

by Leanderson Correia

I'm testing here to use multiple domains with microsoft teams and I'm having trouble making and receiving calls. It seems that when connecting a call, kamailio keeps the tls connection active and reuses it, making the domain parameters used the wrong ones for new call. Can someone help me, or give a tip? -- Att, Lean

3 years, 10 months

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sr-users June 2020