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Job: System Engineer VoIP - Germany / HO
by Gerrit Jacobsen 25 Jul '24

25 Jul '24

VIER is looking for a System Engineer VoIP What VIER does VIER is a global contact center technology provider. VIER develops and runs its own technology in several data centers. You will be part of the Real-time Media Processing Team which is responsible for the telephony connections between carriers, applications and call-center agents. Your mission You are responsible for the support of the VoIP systems as well as the associated infrastructure Since you have sovereignty over the VoIP systems, you are also responsible for performing system upgrades and configurations. You have a lot of freedom to design, plan, and implement new VoIP system components for internal and external projects. You manage and implement change requests and are a specialist for special topics in the communication area. You show a high-sense of responsibility and foresight to keep systems running 24/7 You are prepared to work occasionally also in the evening hours to roll out new features Your skills You have several years experience in the day-to-day operation of telecom systems You have worked extensively with opensource VoIP software such as Kamailio / OpenSIPS, FreeSWITCH You can administer Linux systems such as RockyLinux and Debian You know SIP, RTP, RTCP, TLS, WebRTC protocols and can debug problems with Wireshark You have also worked with MySQL, Git You are communicative and like to work with team colleagues Further beneficial skills Experience with Ansible or Puppet for automation and configuration management. CheckMK, Homer, Hepic, Redis, CGRates, CDR Tools, Python Own VoIP software projects German language skills Please apply here: https://vier.jobs.personio.de/job/1638419?language=en&display=en

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RTJSON: No Json for routing
by felipe.nunes＠comuniki.me 24 Jul '24

24 Jul '24

Good afternoon I have the following error in the application: [rtjson_routing.c:701]: rtjson_update_branch(): no json for routing This error occurs regardless of whether there is high demand or not and the application sending the data correctly. Can you help me with this? Felipe Nunes

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ds_select from xhttp:request
by Christian Berger 24 Jul '24

24 Jul '24

Hi, we want to execute ds_select from an xhttp:request event route. The idea is to be able to get all active targets of a dispatcher set so our monitoring can check that via http which would be convenient for us. So are there any reasons why ds_select wouldn't work in an xhttp:request even route? Regards Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

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Listening on multiple TLS ports
by João Silva 23 Jul '24

23 Jul '24

Hello everyone! Is there a way to define a listen port range for TCP/TLS? Or any macro I can use to create the LISTEN commands? And finally is there a performance penalty for having multiple (2k) listening ports at the same time? I am in a strange situation where I need to register multiple credentials on a single SIP trunk, for each registration I need to use a different local port otherwise the trunk will simply overwrite the previous registrations. Furthermore, when I place a call on that trunk I have to be consistent with the port I used for registration for that specific credential. So I am looking into explicitly defining the port to be used per credential, UACREG allows me to use the contact_addr/socket for the registration part and for placing of outbound calls I can use $fsn to force the correct socket. Best regards, Joao

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IMS PCSCF force source IP address
by Anthony Blandin 23 Jul '24

23 Jul '24

Hello, I am using Kamailio pcscf with ipsec. I have one ipsec listen address which is used by the UE to connect to the PCSCF and one other listen address on port 5060. During the registration until the ipsec is not established, the UE sends the requests to the ipsec listen address and the pcscf is using the other listen address to forward the sip request to the ICSCF. That's is ok for me. But when the ipsec is established after the 200 OK REGISTER, the UE sends a SUBSCRIBE but the pcscf forwards the SUBSCRIBE to the SCSCF using its ipsec listen address instead of the other listen address. I tried to use $fs to force the socket but without success. Do you know how to force the source IP address to be used on the pcscf? Thanks for your support. Regards, Anthony

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mysql error after migration from 5.7.5 to 5.7.6
by Alejandro Guralnik 23 Jul '24

23 Jul '24

Hello, After upgraded from kamailio 5.7.5 to 5.7.6, when did start kamailio process, got the following error: WARNING: db_mysql [km_my_con.c:179]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring After downgrade to 5.7.5 dind't happened. (we don't use mysql ssl connection).

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Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…

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Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria <a href="https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…">aws</a>

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SHM Memory leak?
by Ihor Olkhovskyi 23 Jul '24

23 Jul '24

Hello! I'm trying to get where all the SHM memory gone. For the moment I'm using these settings: -m 8192 -M 1024 which means 8G of SHM meory. Kamailio parameters are the following fork=yes children=8 tcp_children=12 enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 And having ~1000 TLS/WSS clients I'm getting this after 3 days of running: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 2796", "shmem:free_size = 3737947072", (3.7 G) "shmem:max_used_size = 4857418512", (4.8 G) "shmem:real_used_size = 4851987520", "shmem:total_size = 8589934592", "shmem:used_size = 4838988096" (4.8 G) ], "id": 984479 } Means half of SHM memory is gone When running kamcmd mod.stats all shm I'm getting (output truncated for the largest values) Module: core { ... tcpconn_new(1201): 130669280 (0.13 G) ... Total: 131657632 (0.13 G) } Module: sl { ... Total: 7520 } Module: siptrace { ... Total: 13520 } Module: rr { Total: 0 } Module: dialog { ... Total: 146080 } Module: permissions { ... Total: 62144 } Module: htable { .... Total: 3359552 (3.3 M) } Module: rtpengine { ... Total: 31456 } Module: textopsx { Total: 0 } Module: tsilo { ... Total: 75072 } Module: tm { .... Total: 3459504 (3.4 M) } Module: usrloc { .... Total: 1217616 } Module: pua_dialoginfo { ... Total: 8720 } Module: pua { ... Total: 150848 } Module: pike { ... Total: 9504 } Module: websocket { wsconn_add(198): 54139856 (54 M) ... Total: 54148096 (54 M) } Module: debugger { ... Total: 21872 } Module: tmx { Total: 512 } Module: kex { .... Total: 1888 } Module: tls { ... ser_malloc(364): 87246112 (87 M) ... Total: 87997168 (87 M) } Module: secfilter { ... Total: 768 } Module: exec { ... Total: 16 } Module: dispatcher { ... Total: 2992 } Module: cfgutils { ... Total: 48 } Module: app_python3 { ... Total: 32 } So, I'm guessing, where all the memory gone? Is there any way to get more detailed info on SHM? Kamailio 5.8.2 -- Thanks in advance, Ihor

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Not responding to OPTIONS if existing tcp connection
by Jon Bonilla (Manwe) 22 Jul '24

22 Jul '24

Hi all version 5.7.6 Debian 12 I'm having issues with a kamailio server not sending 200 to some OPTIONS requests. It does send 200 to some subscribers but not 100% of the time to 100% of subs. I have enabled debug=3 in that server and this is what I found: If the log says that it found existing tcp connection and it's reusing it I can't see the 200ok in a sngrep capture. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 2 <core> [core/tcp_main.c:2614]: tcpconn_send_put(): send from reader (360359 (18)), reusing fd <core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending... <core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c= 0x7fb0faa893d8 n=384 fd=14 <core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf=#012SIP/2.0 200 Keepalive#015#012Via: SIP/2.0/TCP 10.35.190.105:5060;branch=z9hG4bK15ebf764;rport=55894;received=1.1.1.186#015#012From: "Unknown" <sip:contesta01@10.35.190.105>;tag=as5311c3a7#015#012To: <sip: 1.1.1.138>;tag=054524a74bcccf444a0f08f67c5b7657.d07b5401#015#012Call-ID: 15dcd38c46f915576ded132a11d8ff4c@10.35.190.105:5060#015#012CSeq: 102 OPTIONS#015#012Server: aaa SBC#015#012Content-Length: 0#015#012#015#012 <core> [core/receive.c:531]: receive_msg(): request-route executed in: 302 usec On the other side, if kamailio doesn't find an existing tcp connection (at least nothing in the logs about that) I can see the 200 ok in sngrep. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/receive.c:531]: receive_msg(): request-route executed in: 295 usec All subs use tcp. And the kamailio.cfg is very simple regarding OPTIONS: Sanity, check maxfwd and sl_send_reply(200). Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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