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Re: rtp lost package in 300 concurrent calls and above
by Henning Westerholt 16 Jan '23

16 Jan '23

Hello, please keep the list in CC. Ok, the machine seems to not really using all its cores. Regarding playing a file, this can be also done with rtpengine: https://kamailio.org/docs/modules/5.5.x/modules/rtpengine.html#rtpengine.f.… Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com> Sent: Monday, January 16, 2023 3:35 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] rtp lost package in 300 concurrent calls and above thanks for your reply my kamailio server has 8GB of ram and 4core*2.6 on esxi(dedicated)Ubuntu 22.04.1 LTS. is it low? htop on server shows load average 1.04 at max and I didn't see more than 10% load on any core of the server. I've selected rtp media server because of playback of a voice file over incoming calls in the future usage On Mon, Jan 16, 2023 at 5:45 PM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, do you observe (too) high load on the system with the standard system management tools? If you want to just proxy calls, my recommendation would be to use rtpengine. Here you should be able to achieve higher concurrent calls rates, if you have a few CPUs on a decent machine. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com<mailto:keshavarzianpoorm@gmail.com>> Sent: Monday, January 16, 2023 2:50 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] rtp lost package in 300 concurrent calls and above Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

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rtp lost package in 300 concurrent calls and above
by Mohammad Reza Keshavarzianpoor 16 Jan '23

16 Jan '23

Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

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Authenticated ACK: $au not set?
by Benoit Panizzon 16 Jan '23

16 Jan '23

Hi Gang I have this code snipplet: if (has_credentials("$fd")) { xlog("L_INFO", "$cfg(route): got $rm with credentials. Validate them!\n"); if ($aU == $null) { xlog("L_INFO", "$cfg(route): no auth user, send challenge\n"); } auth_challenge("$fd", "0"); exit; } } Please don't ask if this makes sense, it's a constructed snipplet to show the situation. If $rm is INVITE, this works fine, no challenge is being sent, $au is populated. If $rm is ACK (some CPE seem to send the Credentials with each message) then $au is $null, auth_challenge is called and fails. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Parallel call forking with call authorization
by Mathias Schneuwly 13 Jan '23

13 Jan '23

Hi As a first notice; I already wrote to the mailing list, but without registration. I read afterwards that I should register first because of spam reasons. If this is a duplicate, then I'm sorry. I'm new to mailing lists... I've an issue with parallel call forking together with call authentication. I hope somebody can give me some hints about what the problem could be and how I could solve this problem. I'm using Kamailio 5.5.4 with the python KEMI binding. Kamailio is in front of an Asterisk, which requires call authorization. RTPEngine is also inplace which does SRTP to RTP bridging. Simple calls from and to Asterisk through Kamailio and RTPEngine work without any issue. So the basic setup seems to work fine. The problem starts when I do parallel call forking where both calls are sent to Asterisk. If I use "KSR.tm.t_on_failure("trunk_authentication")" for authentication, the first branch is authenticating and the device rings, but the second branch does not authenticate. Well, that's not 100% true, if the SIP timeout appears on the first branch, I can see in the log, that the second branch enters the trunk_authentication method, but then, the call is already terminated. Call forking to two devices (not behind Asterisk and without call authentication) works also as expected. I made some trials with "KSR.tm.t_on_branch_failure("trunk_authentication")" but in this case, the behavior is strange. Then the authentication for both branches work and both devices are ringing. If a device hooks up, the second device does not stop ringing. And after a few seconds, the first device cancels the call, because it did not get the ACKs. I can also see that the CANCEL is sent from Kamailio to Asterisk, but Asterisk does not react to it. For trunk authentication, I'm using the implementation from the well known examples plus a few own stuff. Btw, NAT is not enabled...: def trunk_authentication(self, msg): scode = KSR.pv.get("$T_reply_code") self.log('notice', 'trunk_authentication {}'.format(scode)) if WITH_NAT and self.ksr_route_natmanage(msg) == -255: return 1 # Call manage_rtp_engine to destroy RTPEngine if it has been used. call_context = self.get_call_context(msg=msg) callee_object = call_context.callee.get_callee_object() if callee_object.connection_type: call_context.manage_rtp_engine() if KSR.tm.t_is_canceled() > 0: return 1 if scode in [401, 407]: KSR.pv.sets('$avp(auser)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_username)')) KSR.pv.sets('$avp(apass)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_password)')) if KSR.uac.uac_auth(): # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if KSR.is_method_in('IBSU'): if KSR.tm.t_is_set("branch_route") < 0: KSR.tm.t_on_branch("ksr_branch_manage") self.log('notice', 'Authentication Asterisk trunk') if KSR.tm.t_relay() < 0: KSR.sl.sl_reply_error() return 1 Any ideas? Best regards Mathias

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src/modules.lst - difference between skipping a module from compile list and excluding a module from compile list
by Дилян Палаузов 13 Jan '23

13 Jan '23

Hello, the generated src/modules.lst file contains: # the list of modules to skip from compile list skip_modules= … # the list of modules to exclude from compile list exclude_modules= … For me excluding a module from the compile list and skipping the module when compiling is the same. Please describe what is the difference between skipping and excluding, either in modules.lst, at https://www.kamailio.org/wikidocs/devel/makefile-system/ or at https://kamailio.org/docs/tutorials/devel/kamailio-install-guide-git/ . Thanks Дилян

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Newbie question
by Joao Paulo Vaz 12 Jan '23

12 Jan '23

Hello masters, I am a newbie who recently got to know Kamailio. I have a scenario and I would like to know if this tool would suit: a central asterisk server with all user accounts and three remote units with Kamailio acting as a proxy. Is there any module or configuration that if the central asterisk server fails to respond, would the remote proxies act in survival mode, allowing local communication by users until the central asterisk server returns a response? Att, Joao Paulo

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unsubscribe.pls
by Stanly Levyy 11 Jan '23

11 Jan '23

*** THIS E-MAIL MESSAGE AND ANY DOCUMENTS ATTACHED IS/ARE INTENDED ONLY FOR THE USE OF THE INDIVIDUAL(S) OR ENTITY TO WHICH IT IS ADDRESSED; AND MAY CONTAIN INFORMATION THAT IS LEGALLY PRIVILEGED, CONFIDENTIAL AND/OR EXEMPT FROM DISCLOSURE UNDER APPLICABLE LAW, including the Electronic Communications Privacy Act, 18 U.S.C. Secs 2510-2521 *** On Wednesday, January 11, 2023, 01:36:44 AM PST, <sr-users-request(a)lists.kamailio.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Performing async_http calls on stateless Loadbalancers (Julius Flohr) 2. Re: Releasing v5.6.3 (Daniel-Constantin Mierla) 3. Performing async_http calls on stateless Load balancers (Lewis Hutchinson) ---------------------------------------------------------------------- Message: 1 Date: Tue, 10 Jan 2023 18:50:44 +0100 From: Julius Flohr <flohr(a)sipgate.de> Subject: [SR-Users] Re: Performing async_http calls on stateless Loadbalancers To: Henning Westerholt <hw(a)gilawa.com> Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Alles wird besser <alleswirdbesser(a)sipgate.de> Message-ID: <34C7487C-0900-4C49-BBB4-49A0291D83E6(a)sipgate.de> Content-Type: multipart/alternative; boundary=Apple-Mail-0AF7C672-1C85-4A98-9DDC-EED95FFB4B7E A message part incompatible with plain text digests has been removed ... Name: not available Type: text/html Size: 7117 bytes Desc: not available ------------------------------ Message: 2 Date: Wed, 11 Jan 2023 08:30:17 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: [SR-Users] Re: Releasing v5.6.3 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org> Message-ID: <454ae3b5-a810-f393-4196-e71c4482e9d3(a)gmail.com> Content-Type: text/plain; charset=UTF-8 Hello, the usual note that preparing the release of Kamailio v5.6.3 starts soon, therefore any commits to branch 5.6 should be announced first on sr-dev mailing list and wait for the ack before pushing to repo. Once the announcement is out, commits can go in the branch directly. Cheers, Daniel On 06.01.23 16:44, Daniel-Constantin Mierla wrote: > Hello, > > I am considering to release Kamailio v5.6.3 (out of branch 5.6) next > week (likely on Wednesday, Jan 11, 2023). If anyone is aware of > issues not yet on the bug tracker, report them there asap in order to > have a better chance to be fixed. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com ------------------------------ Message: 3 Date: Wed, 11 Jan 2023 09:20:28 +0000 From: Lewis Hutchinson <lewis.hutchinson(a)missionlabs.co.uk> Subject: [SR-Users] Performing async_http calls on stateless Load balancers To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Cc: "flohr(a)sipgate.de" <flohr(a)sipgate.de>, "alleswirdbesser(a)sipgate.de" <alleswirdbesser(a)sipgate.de> Message-ID: <CWXP265MB2517CB6E3E8188F3E3BB5B2AC5FC9(a)CWXP265MB2517.GBRP 265.PROD.OUTLOOK.COM> Content-Type: text/plain; charset="us-ascii" Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list -- sr-users(a)lists.kamailio.org To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org ------------------------------ End of sr-users Digest, Vol 212, Issue 33 *****************************************

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Performing async_http calls on stateless Load balancers
by Lewis Hutchinson 11 Jan '23

11 Jan '23

Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis

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Releasing v5.6.3
by Daniel-Constantin Mierla 11 Jan '23

11 Jan '23

Hello, I am considering to release Kamailio v5.6.3 (out of branch 5.6) next week (likely on Wednesday, Jan 11, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

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Help with rtpengine_delete()
by Jose Figueroa 10 Jan '23

10 Jan '23

Hello everyone! I've been struggling with a kamailio scenario with the rtpengine_delete. My scenario is the following: I've setup a Kamailio multihomed, since some of my vendors has provided us a direct link using a different local subnet in eth2 and I have my public IP in eth1. The kamailio i've been working on is using app_lua module. SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth2) 10.203.5.8 <------> Vendor 10.203.5.20 100.XXX.XXX.22 Basically, I receive a call from public IP eth1 and according to the logic from the vendor I forward statefully the call to 10.203.5.20 in second branch. I've been reading a lot of issues and documentation such as:[#840 rptengine], [#875 rtpengine] and [#2222 kamailio], but I don't really get working the rtpengine_delete() correctly, since I go to netstat -plan|grep rtpengine and I see some sessions still that are not freed. My second case is the following when I receive a 503 from another GW: SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth1) 200.XXX.XXX.14 <-----------> Vendor GW 233.XXX.XXX.19 100.XXX.XXX.22 In that case I still see the rtpengine ports, mostly with my public IP 200.XXX.XXX.14 but some of them with the LAN 10.203.5.8 IP. Also, something curious I see on my 2nd use case is the following: https://pastebin.com/raw/B4vCgExX So, it would be great if someone can point me to the right direction, how and when I should use rtpengine_delete in order to finish the rtpengine sessions correctly and the ports can be freed up. Any help will be great! [840] https://github.com/sipwise/rtpengine/issues/840 [875] https://github.com/sipwise/rtpengine/issues/875 [2222] https://github.com/kamailio/kamailio/issues/2222 -- Regards, Jose Figueroa

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