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38 participants
37512 discussions

LCR GW
by Antoine DEBARRE 20 May '22

20 May '22

Hi, Is there a way to specify an outbound proxy for just a few lcr gateways added in table lcr_gw ? Thanks in advance ! Antoine

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Path MTU issues over UDP/IPv6
by Rick van Rein 20 May '22

20 May '22

Hello, IPv6 routers never fragment packets. Rather, they drop a packet that is too large for a (local) MTU and send back ICMPv6 "Packet Too Big". This seems to cause loss of larger SIP messages when an ISP tunnels their IPv6 at the expense of the MTU. The pmtu_discovery flag sets Don't Fragment in IPv4 traffic; in IPv6 this is an implied property. Does Kamailio learn a lower MTU from any "Packet Too Big" for IPv6 even if pmtu_discovery is not set? Future resends can then be fragemented appropriately. The udp_mtu setting diverts to another protocol, but that would be a setting as low as the worst peer, impacting all. It would be a weird struggle with a telco serving many. PMTU would be better to rely on, but how does it work in Kamailio? Details on https://www.rfc-editor.org/rfc/rfc3542#section-11.3 https://stackoverflow.com/questions/38817837/how-does-mtu-retransmission-wo… Thanks, -Rick

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Forwarding SIP status code to HTTP client
by Tim Anderson 19 May '22

19 May '22

Hi, I have a kamailio that is has an HTTP message server. Is it possible to send the SIP status code back in the HTTP reply? Is it possible to suspend an xhttp request while some other stuff happens and then send the reply later? The relevant bit of the config currently looks like event_route[xhttp:request]{ if($hu == "/sms"){ $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:1234@example.com"; $uac_req(furi)="sip:4567@example.com"; $uac_req(turi)="sip:1234@example.com"; $uac_req(body)=$rb; uac_req_send(); xhttp_reply("200", "OK", "text/plain", "Sent") } } Regards, Tim

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Kamailio Stop Responding Randomly
by M Arqum CH 19 May '22

19 May '22

Kamailio first gives timeout on ctl unix socket as I am exporting data using prometheus exporter, after a few minutes it stops responding to any sip request on default/production udp port binded with multiple IP addresses for sip traffic. The other listing port which is used by keepalived check script for bing-pong keeps responding to option messages. Its very random behaviour occurs after a few days or after a few months, the system having max load 350 concurrent calls with load average 200-300. I am attaching core dump which is collected before restarting the process. Please let me know if more information is required. version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 Thankyou in advance. -- Regards M Arqum

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Relay wss to Asterisk
by Michal Hajek 18 May '22

18 May '22

Hello, can i configure kamailio for relay wss to asterisk/pjsip? WebRTC client ( chrome ) --> WSS -> Kamailio --> WSS -> Asterisk with pjsip Asterisk accept wss://ip_address:wss_port/ws Thanks Michal Hajek

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Split audio and video into different calls
by Rüdiger, Christoph 18 May '22

18 May '22

Hi everyone, we are running a call center with really good to use hardphones and audio only up to this point. Because of changing regulation, we soon have to deal with video calls as well, but only for a fraction of the calls (below 1%). Instead of switching everyone to clumsy softphones for the rare case of a video call, I’m exploring the ability to branch an incoming call into a pure audio call to the hardphone and optionally a pure video call to the softphone. Has anyone done this in the past and can share some experiences/ideas on 1. Can it be done purely with Kamailio and rtpengine? I don’t have a clear picture of how to do it purely with Kamailio on the SIP side, because we end up with two independent legs (to the hardphone and the softphone). Or do I need to deploy an Asterisk, Yate, freeswitch, etc. and doing it as a 3-way conference where one party gets both RTP streams (the trunk) and the other two parties only get one RTP stream (either audio or video)? 2. Was it worth the effort or a steady candidate for troubles? I’m looking forward to your thoughts, especially on question b). Have a nice weekend, Christoph --- Christoph Rüdiger Digital Operations Architect DTX OU DACH M +49 152 3828 0376 | christoph.ruediger(a)tkelevator.com<mailto:christoph.ruediger@tkelevator.com> TK Aufzüge GmbH | Münchener Str. 100c | 45145 Essen | Deutschland | www.tkelevator.com/de<http://www.tkelevator.com/de> VORSITZENDE DES AUFSICHTSRATS Inge Delobelle | GESCHÄFTSFÜHRUNG Burkhard Schlenker (Vorsitzender) | Christian Fongern | Katja Schrafft | SITZ DER GESELLSCHAFT Neuhausen a. d. F. | REGISTERGERICHT Stuttgart HRB 731152 This e-mail (including any attachments) may contain confidential and/or privileged information. Any unauthorized use or dissemination of this message in whole or in part is strictly prohibited. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail.

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kamailio WebSocket not respond sometimes
by dolphin 18 May '22

18 May '22

Hello, kamailio could not respond to the question: This machine is on aws, there is no high load, there is an external ip listening to the WebSocket 443 port, and sometimes there are some unknown detection signals "SSTP_DUPLEX_POST" that will cause port 443 to fail to respond. Need to restart kamailio Then, it stopped responding again after about a month. Attach the detailed log details: root@sip1-aws:~# /home/pkg/kamailio/sbin/kamailio -V version: kamailio 5.3.9 (x86_64/linux) a5fd0a flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: a5fd0a compiled on 22:31:18 Mar 31 2022 with gcc 5.4.0 Thanks.

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RabbitMQ hangs if connection lost
by Avremy Cohen 17 May '22

17 May '22

Hi When I Publish to RabbitMQ and the connection is down, the server just hangs and there is no timeout. I am using KEMI with python3 and have tried different timeout functions to circumvent the hanging but it doesn't seem to work 1. Is there a way to tell if RabbitMQ is connected? 2. using python is there a way to timeout a KSR function? Thanks guys!

1 0

Kamailio Crashed
by M Arqum CH 17 May '22

17 May '22

Dear Team, Kamailio instance got crashed, I found this error in logs. Please give advice on how I can avoid this in future. *kernel: [52641555.303165] kamailio[123814]: segfault at 30 ip 00007f42f70db770 sp 00007ffd5d95a9f0 error 4 in dialog.so[7f42f7033000+c9000]* version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Thank you in advance. Regards M Arqum

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Failing Audio stream on Kamailio+webrtc with integration to asterisk server
by Kelvin Ezenwaka 16 May '22

16 May '22

Summary: We have a Kamailio server as our sip proxy server, sip firewall with WebSocket and RTP engine configured on it (the Kamailio Server). But we experience one way audio, no audio, hang up after 30s, when we try making calls between internal extensions (eg extension 100 to call extension 105) and external calls (eg extension 100 to call mobile number 09056925668) as well. The call flow is as: Webrtc client<------->Kamailio1+rtpengine<-----Asterisk---->Kamailio2<-------->Telco Provider The asterisk communication between all three boxes is via local IP (All ports open between them). The Kamailio 1 box where we have our sip registration cache, has rtp ports and wss port open on the internet. Asterisk 18.9.0 Kamailio 5.5.3 + RTPengine 10.4.0.0 Debian 11 bullseye Kindly see attached below a diagram depicting the VoIP network flow and also attached are logs files for the webrtc client and server side. Ezenwaka Kelvin Arinze VOIP Engineer Tel: (+234) 9056925668 DISCLAIMER: The message and its attachments are for designated recipient(s) only and may contain privileged, proprietary and private information. If you have received it in error, kindly delete it and notify the sender immediately. PressOne accepts no liability for any loss or damage resulting directly and indirectly from the transmission of this e-mail message.

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