sr-users May 2021

sr-users@lists.kamailio.org

57 participants
66 discussions

Roadmap to the next major release - v5.5.0
by Daniel-Constantin Mierla 16 Jun '21

16 Jun '21

Hello, it is the time to plan a bit the road to the next major release, to be versioned 5.5.0. The 5.4.0 was release by end of July 2020, therefore we are approaching the end of our usual development cycle, also during the last online devel meeting we considered the 2nd part of spring 2021 as a target time frame for 5.5. To narrow it down, I would propose to freeze the development at the end of March 2021, test during April and maybe beginning of May, then release v5.5.0. So far we have a lot of development to existing components, by the freeze date I expect 3-5 new modules to be in the repo as well. If anyone wants a different time line towards 5.5.0, let's discuss and choose the one that suits most of the developers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Problem NOTIFY in Presence module
by hamed r 07 Jun '21

07 Jun '21

hello dears ,i am using presence module and NOTIFY message are working properly. but when i put the database on another server , NOTIFY message will not be sent to subscribers. what is problem?

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handle_ruri_alias() and multiple aliases
by Denys Pozniak 02 Jun '21

02 Jun '21

Hello! I need help understanding how the *handle_ruri_alias()* function works. *Call-flow:* Upstream operator -> Kamailio Proxy -> Edpoint. The upstream operator in the initial SIP INVITE in the Contact field sends us an alias parameter, in turn, our Kamailio Proxy adds its own too. The Contact after the Proxy looks something like this: *Contact: <sip:10.0.0.115:5060;alias=10.0.0.115~5060~1;alias=3.3.3.3~5060~1>* When the Endpoint sends us a SIP BYE, then the *handle_ruri_alias()* absorbs not the last alias, but the first one, and this leads to the incorrect formation of a $du. The RURI after the Proxy looks something like this: *Request-Line: BYE sip:10.0.0.115:5060;alias=3.3.3.3~5060~1 SIP/2.0* The first thing that comes to mind is a modification of the original Contact field with storing it's within a dialogue ... Is it possible to elegantly solve this problem? -- BR, Denys Pozniak

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STIR/SHAKEN tests
by David Villasmil 02 Jun '21

02 Jun '21

Hello guys, I want to test secsipid, but i don't yet have the certificate. So i thought i'd create a cert like: openssl req -new -newkey rsa:4096 -nodes -keyout snakeoil.key -out snakeoil.csr openssl x509 -req -sha256 -days 365 -in snakeoil.csr -signkey snakeoil.key -out snakeoil.pem Then i'm simply doing: $var(rc) = secsipid_add_identity("$fU", "$rU", "A", "", " https://somedomain.com/stir/$rd/cert.pem <https://kamailio.org/stir/$rd/cert.pem>", "/etc/kamailio/snakeoil.pem"); if ( $var(rc) ) { xlog("L_ERR", "[STIR/SHAKEN][$ci] Shaken authentication added (SIP Identity Header created)\n"); } else { xlog("L_ERR", "[STIR/SHAKEN][$ci] Failed\n"); } But no matter what i do it silently fails: INVITE d54c2919-39b6-123a-95a7-0e29a5289b8d} <script>: [STIR/SHAKEN][d54c2919-39b6-123a-95a7-0e29a5289b8d] Failed I have debug on 6, but i don't get more info regarding the error. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Releasing Kamailio v5.4.6
by Daniel-Constantin Mierla 02 Jun '21

02 Jun '21

Hello, I am considering to release Kamailio v5.4.6 sometime next week, likely on Wednesday or Thursday (June 2 or 3, 2021). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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IMS IPSec does use UDP socket?
by H Yavari 01 Jun '21

01 Jun '21

Hi to all, I am interested in the implementation of IPsec module. When I run Kamailio, it is creating processes based on the below formulate: UDP children * IPSEC_MAX_CONN * interfaces (v4 and v6) + (TCP+UD) processes for each interface. For example: children = 16tcp_children=16 IPSEC_MAX_CONN = 25and two interface : IPv6 and V4 Kamailio process count = (16 * 25 * 2 ) + 16 + 16 = 832 But why IPsec module is listening to UDP not TCP? this is an internal connection? Thank you. Regards,Hossein

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SDP body, add maxptime attribut
by Victor Velo 01 Jun '21

01 Jun '21

Hello, I would like to add the maxptime attribut to the sdp body. How and where should I do insertion? This kamailio is also using RTPEngine, so is it better to do the insertion using RTPEngine? And the source of the SDP is actually FreeSwitch :), maybe there to tweak? Of course, I don't know either way of doing it so please advise. Thanks, Victor

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New MariaDB server "limiting" Kamailio CPS
by Stephen Lokker 01 Jun '21

01 Jun '21

Good afternoon All, Im hoping for some advise or direction with the below. We are running a Kamailio SBC used for high call volumes and recently decided to upgrade the Database servers as the existing servers were struggling. Both the old and new databases are using MariaDB(version details all below), the problem we face is that since migrating to the new database there appears to a limitation to the amount of DB connections that can be processed or accepted. The SBC functions normally until the CPS starts to exceed around 100cps at which point the SBC stops responding to invites(in bursts which resembles flood prevention). We have checked both the Kamailio logs and the MySQL logs on the DB, both log files have no indication of errors that would cause the problem experienced(the logs attached are seen when the SBC is cut back to use the old DB, not sure they are related to the fault), the connections/processes in mysql do not drastically increase or get stuck either. The fault points to a db connection limit of some kind but all users have no limitations set. If anyone has experienced a similar problem or has any advise it would be greatly appreciated. Some info on the Kamialio and DB versions New DB root@voice-jhb-db01:~# mysql --version mysql Ver 15.1 Distrib 10.3.27-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2 Old DB [root@isadb01 ~]# mysql --version mysql Ver 15.1 Distrib 10.0.36-MariaDB, for Linux (x86_64) using readline 5.1 Kamailio SBC [root@sbc01 ~]# kamailio -v version: kamailio 5.1.4 (x86_64/linux) Kind regards, Stephen Voice Engineer Tel: 2404 Fax: 3468 Email:

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How To setup Provider SIP trunk with registration
by zak ast 31 May '21

31 May '21

Hi, I know I can you uacreg to register a SIP trunk as a SIP client with username/password. But how can I set up a SIP trunk as a provider and get the other end (SIP client) to register with username/password. Regards, Zak.

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Architecture - your opinion is welcome and your experience too
by Daian Conrad 28 May '21

28 May '21

Hi all, I am studying and improving my understanding of how kamailio works, I have used asterisk for a few years(and starting learning freeswitch too), and I use an application to manage extensions, trunks and other media services. I'm building a new scenario where kamailio is facing the internet and the asterisk(s) are internally on private networks, would like the opinion of you who are more experienced with more complex environments than me. I cannot change the application that uses asterisk for now, due to the effort to develop event monitoring, cdr and other features that I use today. I imagined and working on making kamailio functional by saving the registration of extensions with location on DB, and with the UAC module making the registration of extensions on asterisk(s), replacing the registration address with the address of kamailio, its functional at this point, but data replication with extensions turns things some hard to mantain manually. I think about using the DNS domain for each asterisk and make this forwarded, each asterisk response for a fqdn and its extensions, like ( pbx1.example.com is forwarded to asterisk1, pbx2.example.com to asterisk2) and so on. Read about the dispatcher, rr, htable,carrierroute module to identify the domain and forward based on that , none made me sure to be chosen for the role, however, everyone has the resources to do it. I am very wrong to follow this path, which option in your opinion is "less" painful for a beginner apprentice like me? I think about simple proxy based on domain requests(all messages and dialogs) to asterisk where is responsible for the domain(realm) but, loss the ability to use great security features of kamailio, is what I understand now with the knowledge I have. Sorry and forgive my english, i'm not very good at writing. -- ================ **Daian Conrad** E-mail: daian.conrad(a)gmail.com OpenS Team (DaCoD) Linux user: #248912

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sr-users May 2021