sr-users April 2026

sr-users@lists.kamailio.org

27 participants
24 discussions

can't use msg_apply_changes() on failure route?
by David Villasmil 22 Apr '26

22 Apr '26

Hello all, I need to make changes on the SDP. The first change is fine on the request route, but i only need to make changes to it in the failure route to set it properly for the failover endpoint. I restore the sdp, but my problem is i can't modify it since i can't manipulate it: # Restore original dual-transport SDP before stripping for new dest -- $avp(orig_sdp) was saved on the initial request route. set_body("$avp(orig_sdp)", "application/sdp"); # Strip unsupported transport for new destination if ( $du =~ "tls" ) { xlogl("L_INFO", "[RTF_CARRIER] TLS dest - stripping RTP/AVP\n"); sdp_remove_transport("RTP/AVP"); } else { xlogl("L_INFO", "[RTF_CARRIER] Non-TLS dest - stripping RTP/SAVP\n"); sdp_remove_transport("RTP/SAVP"); } Any ideas how to accomplish this? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com

1 0

Topos and TLS
by Martin Nyström 22 Apr '26

22 Apr '26

Hi, How am I expected to handle TLS when using TOPOS? listen=tls:192.168.5.122:5061 advertise foobar.pbx.com:5061 modparam("topos", "storage", "db") modparam("topos", "contact_host", “foobar.pbx.com") modparam("topos", "dialog_expire", 10800) When using TLS, TOPOS sets the contact header as something like this for a 200 OK returned. Contact: <sip:atpsh-69e6453f-a4-4@foobar.pbx.com> Is this valid? Or should it not be foobar.pbx.com:5061;transport=TLS It appears some vendors blindly use the contact header and returns the ACK for an example to a non TLS endpoint, despite the dialog (INVITE) starting out as TLS. /M

3 3

Kamailio v6.1.2 Released
by Daniel-Constantin Mierla 22 Apr '26

22 Apr '26

Hello, Kamailio SIP Server v6.1.2 stable release is out. This is a maintenance release of the latest stable branch, 6.1, that includes fixes since the release of v6.1.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v6.1.x. Deployments running previous v6.1.x versions are strongly recommended to be upgraded to v6.1.2. For more details about version 6.1.2 (including links and guidelines to download the tarball or from GIT repository), visit: - https://www.kamailio.org/w/2026/04/kamailio-v6-1-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 7-8, 2026 - Berlin, Germany -- kamailioworld.com

1 0

Releasing v6.1.2
by Daniel-Constantin Mierla 22 Apr '26

22 Apr '26

Hello, the release of Kamailio v6.1.2 (out of branch 6.1) is considered to be done on Wednesday, April 22, 2026. 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 (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 7-8, 2026 - Berlin, Germany -- kamailioworld.com

1 1

6.1 crashes
by Juha Heinanen 20 Apr '26

20 Apr '26

After updating my sip proxy to latest kamailio 6.1 branch running on Debian Trixie , I have noticed several crashes. syslog has this: 2026-04-19T16:41:19.664532+03:00 box kernel: kamailio[944355]: segfault at 0 ip 0000558b1c9ea97e sp 00007fffa1deb610 er\ ror 6 in kamailio[f297e,558b1c926000+3b0000] likely on CPU 2 (core 0, socket 4) 2026-04-19T16:41:19.664633+03:00 box kernel: Code: 00 00 00 48 8b 57 08 48 8b 07 f3 0f 7e 0d 8a 72 3f 00 48 8b 4f 10 64\ 4c 8b 24 25 28 00 00 00 4c 89 a4 24 88 00 00 00 41 89 f4 <48> 89 02 66 0f 6c ca 48 89 50 08 0f 11 07 48 83 c3 10 74 13\ 48 39 2026-04-19T16:41:19.936157+03:00 box /usr/sbin/kamailio[944455]: CRITICAL: <core> [core/pass_fd.c:284]: receive_fd(): E\ OF on 34 2026-04-19T16:41:20.435008+03:00 box /usr/sbin/kamailio[944292]: ALERT: <core> [main.c:824]: handle_sigs(): child proce\ ss 944355 exited by a signal 11 2026-04-19T16:41:20.435143+03:00 box /usr/sbin/kamailio[944292]: ALERT: <core> [main.c:828]: handle_sigs(): core was ge\ nerated 2026-04-19T16:41:20.435191+03:00 box /usr/sbin/kamailio[944292]: INFO: <core> [main.c:851]: handle_sigs(): terminating \ due to SIGCHLD 2026-04-19T16:41:20.435613+03:00 box /usr/sbin/kamailio[944405]: INFO: <core> [main.c:907]: sig_usr(): signal 15 receiv\ ed ... and core dump shows this: #0 __pthread_kill_implementation (threadid=<optimized out>, signo=signo@entry=6, no_tid=no_tid@entry=0) at ./nptl/pthread_kill.c:44 warning: 44 ./nptl/pthread_kill.c: No such file or directory (gdb) where #0 __pthread_kill_implementation (threadid=<optimized out>, signo=signo@entry=6, no_tid=no_tid@entry=0) at ./nptl/pthread_kill.c:44 #1 0x00007f7f7ef509ff in __pthread_kill_internal (threadid=<optimized out>, signo=6) at ./nptl/pthread_kill.c:89 #2 0x00007f7f7eefbcc2 in __GI_raise (sig=sig@entry=6) at ../sysdeps/posix/raise.c:26 #3 0x00007f7f7eee44ac in __GI_abort () at ./stdlib/abort.c:77 #4 0x0000558b1c926d45 in ?? () #5 <signal handler called> #6 0x00007f7f7efb4907 in __GI_sched_yield () at ../sysdeps/unix/syscall-template.S:120 #7 0x0000558b1cc6c7ed in timer_del_safe () #8 0x0000558b1c9f8899 in destroy_dns_cache () #9 0x0000558b1c932537 in cleanup () #10 0x0000558b1c9332ed in ?? () #11 0x0000558b1c933fd3 in handle_sigs () #12 0x0000558b1c9395fd in main_loop () #13 0x0000558b1c92ae91 in main () Any ideas? -- Juha

2 7

Attempt to remove unwanted media streams has no effect
by David Villasmil 20 Apr '26

20 Apr '26

Hello all, It's been a while! I'm receiving from freeswitch an INVITE with both RTP/AVP and RTP/SAVP. Providers don0t like this, they want to see a single audio stream offer, and i want to remove the non-supported stream on kamailio, since i have to mix and match on failover. I see kamailio attempting to remove the streams but it has no effect whatsoever, you'll see below enumerating the streams to remove, but doesn't actually removes them... (i didn't print out the SDP, it's done by the module, so it should be ok, furthermore that's what i see being forwarded): ``` 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 sdpops [sdpops_mod.c:1374]: sdp_remove_transport(): attempting to search for transport type: [RTP/AVP] 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 sdpops [sdpops_mod.c:1390]: sdp_remove_transport(): stream 0 of 0 - transport [RTP/SAVP] 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 sdpops [sdpops_mod.c:1390]: sdp_remove_transport(): stream 1 of 0 - transport [RTP/AVP] 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 sdpops [sdpops_mod.c:1398]: sdp_remove_transport(): removing transport stream: RTP/AVP 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 sdpops [sdpops_mod.c:1390]: sdp_remove_transport(): stream 2 of 0 - transport [RTP/SAVP] 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 sdpops [sdpops_mod.c:1390]: sdp_remove_transport(): stream 3 of 0 - transport [RTP/AVP] 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 sdpops [sdpops_mod.c:1398]: sdp_remove_transport(): removing transport stream: RTP/AVP 5(13) DEBUG: INVITE <null> 1.2.3.4:5080 59bb066c-b53d-123f-47b3-1228f25c4047 textops [textops.c:2524]: ki_set_body(): new body: [v=0 o=FreeSWITCH 1776432039 1776432040 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 25032 RTP/SAVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:SJqvkk2FLvI0tOHPkn7Ac8TkrADNB7buSM6h9ZTBZHHh1OV9uOkGV1DF1DU= a=crypto:2 AEAD_AES_256_GCM inline:jnzCFpgWPjCIAz13r3vCfqbGzxE1x30DJl/0xOVWWS1m34Ka+dD1c45GQX4= a=crypto:3 AEAD_AES_128_GCM_8 inline:B4mysNibq+BH01ASVvceC7g69HmZUHrYtMVmDA== a=crypto:4 AEAD_AES_128_GCM inline:NLVvi1aEW+oWM+AQtV/5hKgMR274BscLBZAJRw== a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:eE3dGn207RnrLSeeTvSP05vUnQax7jATdiizGrVS9mxJQo/pcvvMlhoJdSYT+w== a=crypto:6 AES_192_CM_HMAC_SHA1_80 inline:VDveo9syEdOzxaIn9dpcMfG5ygdOv68yCpAnYwA8f0YlY51lElk= a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:8025UxyTFme6casTeasbrPYUt/ta2yoZGlQFvags a=crypto:8 AES_256_CM_HMAC_SHA1_32 inline:Ux9cBuGGe1Ft8knU7ZeWqnrOETHIqmVGjOTdi78GesXti6wO1ud0gvneW2JT+Q== a=crypto:9 AES_192_CM_HMAC_SHA1_32 inline:kxmHW+oXsu0NuCOwW3uXRmlMAgT1+2GUZjK2ExPC8sD2Xckonms= a=crypto:10 AES_CM_128_HMAC_SHA1_32 inline:MigohPAYbhU2QbJCRN6UUXcKIRAdhFur6EMDAziM a=crypto:11 AES_CM_128_NULL_AUTH inline:R9Aw9RUizWe8r07U8oV02fU+nQ4FkGmkhGflcoPY a=ptime:20 m=audio 25032 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:13 CN/8000 a=ptime:20 m=video 23724 RTP/SAVP 102 b=AS:1024 a=rtpmap:102 VP8/90000 a=rtcp-fb:102 ccm fir a=rtcp-fb:102 ccm tmmbr a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli a=crypto:1 AEAD_AES_256_GCM_8 inline:EzIhXMiY/IU6pwGdiwIGs43YlIoBYrNyiK8XR1lc3auF3qf2GNCIrADZh8k= a=crypto:2 AEAD_AES_256_GCM inline:rSsEBQWi3qlC6nOcYD+xoz8Ux2dlPZDFf2zUOKR6XREkwvbExgQp1RCOjsQ= a=crypto:3 AEAD_AES_128_GCM_8 inline:nwMmmSlFJjhgrqCujnsXV2nFsCBpmkeZ9y5FMA== a=crypto:4 AEAD_AES_128_GCM inline:bAWJtV65VMLHXVvfHwbgwBEYa5Fqi0t9DCB2zg== a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:5m+7Gpqd1AIK8gEAMwti3yrS/cf5ig75Z9OSAAnjT6X8fDG9rBnYVZu0n9pyWg== a=crypto:6 AES_192_CM_HMAC_SHA1_80 inline:zbzJ7sGdUdytETroy6Txg4YU/qBjl1/JKR7VadHx8y61tfTfEaA= a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:5UiBkGVL9s/CjGIb3SPODZkTwJcfCipkiGZH4KI6 a=crypto:8 AES_256_CM_HMAC_SHA1_32 inline:KzBizz3SfxkK8n3MRvA70hCT9X95imVZldDAWzeHEjobSzzEGbAyc7YCZFYE1g== a=crypto:9 AES_192_CM_HMAC_SHA1_32 inline:hwSaAGg6n6loiNLIqg30ZAVaor5pm/ds8Llf2ti95pYR2kGPG3Y= a=crypto:10 AES_CM_128_HMAC_SHA1_32 inline:3/cf0KoXXso4L3+vpL95iBe3Nnu+QGdZxIDC1ulq a=crypto:11 AES_CM_128_NULL_AUTH inline:61izsKf/RCgzBtsNAAiTe+3w1sW/8GP6oET7cWcK m=video 23724 RTP/AVP 102 b=AS:1024 a=rtpmap:102 VP8/90000 a=rtcp-fb:102 ccm fir a=rtcp-fb:102 ccm tmmbr a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli ] ``` Any help is greatly appreciated, thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com

5 6

Failover-Failback without any Load-Balancing
by Kama User 19 Apr '26

19 Apr '26

Hi. I'm trying to make failover-failback without any Load-Balancing with kamailio paired with HiPath 3800. Ubuntu 20.04, kamailio 5.5.7 # dispatcher.list 1 sip:192.168.1.10:5060 0 0 p=primary 1 sip:192.168.1.11:5060 0 0 p=secondary # IP Addresses of Kamailio Server listen=udp:10.0.0.203:5060 listen=tcp:10.0.0.203:5060 listen=tls:10.0.0.203:5060 # --- Module Parameters --- modparam("tm", "fr_timer", 10000) modparam("tm", "fr_inv_timer", 60000) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_mode", 1) # Probes all nodes, even if inactive modparam("dispatcher", "ds_probing_threshold", 2) modparam("dispatcher", "ds_inactive_threshold", 5) # --- Routing Logic --- request_route { if (is_method("INVITE|REGISTER")) { if (!ds_select_dst("1", "8")) { sl_send_reply("503", "No servers available"); exit; } xlog*("Selected Destination $rd/$du \n"); t_on_failure("FAILOVER"); t_relay(); } } failure_route[FAILOVER] { if (t_check_status("503|408")) { xlog("Failed Destination $rd/$du \n"); if (ds_next_dst()) { t_on_failure("FAILOVER"); t_relay(); } } } When both PBXs are online I check the log file and see that Selected Destination is 10.0.0.203/sip:10.0.0.65. I also check via sngrep: if both PBXs are up I see OPTIONS to both of them 10.0.0.65 and 10.0.0.75 When I shutdown 10.0.0.65, log file says Failed Destination is 10.0.0.203/sip:10.0.0.65 and that's it. ds_next_dst not triggering at all. Thanks in advance.

4 13

Re: how to get all non-execluded modules to single kamailio Debian package?
by Juha Heinanen 19 Apr '26

19 Apr '26

Just to complete this topic, I managed to do it by: 1) removing all other packages except kamailio from debian/control 2) added LOTS of modules to debian/rules EXCLUDED_MODULES list 3) added my local module to MOD_LIST_UTILS list The process would have been MUCH easier, if there would have been INCLUDED_MODULES list. Also, when new modules are introduced, I would not need to do anything. -- Juha

2 2

Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
by Ben Fitzgerald 17 Apr '26

17 Apr '26

Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: <sip:40081@kamailio:5060> Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER ========================= sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

4 8

problems using kamailio
by Peter Walber 17 Apr '26

17 Apr '26

Hello, I try to use kamailio as webrtc-sip-gateway between my openhab smart home system and my doorbird doorbell with phone and camera. I try to use the product of "https://github.com/florian-h05/webrtc-sip-gw". Which integrates kamailio together with rtpengine to the gateway. I want to use openhab as SIP Client and have a Telekom Speedport Smart 4 Router on the other side. The webrtc-sip-gw from florian-h05 shall be the software between Openhab and the speedport smart 4. On the speedport smart router the SIP Server is enabled and configured. I can sucessfully call with zoiper extern phone numbers. The doorbird als is configured correctly. I can call the doorbird with my zioper softphone using the SIP interface of the doorbird. The runtime system is a docker container on debian 12. I want to call the doorbird by SIP from my openhab SIP client. I can initiate the call, but it doesn't establish Here is the log from the docker container. --------------------------------------------------------------------------------------------------- 2025-08-07T07:46:57.384687165Z 9(52) INFO: <script>: HTTP Request Received 2025-08-07T07:46:57.391479050Z 9(52) INFO: <script>: START: REGISTER from sip:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:46:57.391516534Z 9(52) INFO: <script>: Current Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:243e461d-f0cb-4c09-9a63-f57c9bba8d6b>";expires=600 2025-08-07T07:46:57.391526535Z 9(52) INFO: <script>: Setting new Contact header: <sip:**73@192.168.2.209:5060> 2025-08-07T07:47:00.195739350Z 9(52) INFO: <script>: START: INVITE from sip:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:47:00.195800424Z 9(52) INFO: <script>: Current Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;transport=ws;ob> 2025-08-07T07:47:00.195811886Z 9(52) INFO: <script>: Setting new Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;transport=ws;ob;alias=192.168.2.185~39510~6;alias=192.168.2.185~39510~6> 2025-08-07T07:47:00.196163528Z 9(52) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:**72@speedport.ip 2025-08-07T07:47:00.196181228Z 9(52) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:**72@speedport.ip, method:INVITE, status:<null>, extra_id: z9hG4bK44082210, rtpengine_manage: replace-origin replace-session-connection via-branch=extra rtcp-mux-demux SDES-off ICE=remove RTP/AVP 2025-08-07T07:47:00.196608235Z INFO: [penvq8f06ehafutk9ivf]: [control] Received command 'offer' from 127.0.0.1:50790 2025-08-07T07:47:00.196625247Z NOTICE: [penvq8f06ehafutk9ivf]: [core] Creating new call 2025-08-07T07:47:00.198519843Z INFO: [penvq8f06ehafutk9ivf]: [control] Replying to 'offer' from 127.0.0.1:50790 (elapsed time 0.001979 sec) 2025-08-07T07:47:01.927528447Z 9(52) INFO: <script>: START: CANCEL from sip:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:47:01.927577111Z 9(52) INFO: <script>: Current Contact header: <null> 2025-08-07T07:47:01.927586439Z 9(52) INFO: <script>: Setting new Contact header: <null> 2025-08-07T07:47:09.169489716Z 7(50) INFO: <script>: WebSocket connection from 192.168.2.185:39510 has closed *2025-08-07T07:47:27.355185671Z 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found* 2025-08-07T07:47:27.914769340Z 9(52) INFO: <script>: HTTP Request Received 2025-08-07T07:47:27.951608347Z 9(52) INFO: <script>: START: REGISTER from sip:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:27.951663073Z 9(52) INFO: <script>: Current Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:39eac0ca-0aea-4526-a5c7-0516bc8a2d71>";expires=600 2025-08-07T07:47:27.951915520Z 9(52) INFO: <script>: Setting new Contact header: <sip:**73@192.168.2.209:5060> 2025-08-07T07:47:30.168546446Z 2(45) INFO: <script>: BRANCH FAILED: z9hG4bK4408221 + 0INFO: [penvq8f06ehafutk9ivf]: [control] Received command 'delete' from 127.0.0.1:36296 2025-08-07T07:47:30.168587880Z INFO: [penvq8f06ehafutk9ivf]: [core] Deleting call branch '' (via-branch 'z9hG4bK44082210') 2025-08-07T07:47:30.168597011Z INFO: [penvq8f06ehafutk9ivf]: [core] Call branch '' (via-branch 'z9hG4bK44082210') deleted, no more branches remaining 2025-08-07T07:47:30.168605215Z INFO: [penvq8f06ehafutk9ivf]: [core] Deleting entire call 2025-08-07T07:47:30.168612883Z INFO: [penvq8f06ehafutk9ivf]: [core] Final packet stats: 2025-08-07T07:47:30.168620604Z INFO: [penvq8f06ehafutk9ivf]: [core] --- Tag 'utenlull77', created 0:30 ago for branch '' 2025-08-07T07:47:30.168628359Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscribed to '' 2025-08-07T07:47:30.168636445Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscription for '' 2025-08-07T07:47:30.168645446Z INFO: [penvq8f06ehafutk9ivf]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec 2025-08-07T07:47:30.168653691Z INFO: [penvq8f06ehafutk9ivf]: [core] --------- Port 192.168.2.209:23484 <> 192.168.2.185:60650, SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:47:30.168661939Z INFO: [penvq8f06ehafutk9ivf]: [core] --- Tag '', created 0:30 ago for branch 'z9hG4bK44082210' 2025-08-07T07:47:30.168669844Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscribed to 'utenlull77' 2025-08-07T07:47:30.168677713Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscription for 'utenlull77' 2025-08-07T07:47:30.168685661Z INFO: [penvq8f06ehafutk9ivf]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec 2025-08-07T07:47:30.168693705Z INFO: [penvq8f06ehafutk9ivf]: [core] --------- Port 192.168.2.209:23472 <> :0 , SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:47:30.168718608Z INFO: [penvq8f06ehafutk9ivf]: [core] --------- Port 192.168.2.209:23473 <> :0 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:47:30.168728065Z INFO: [penvq8f06ehafutk9ivf]: [control] Replying to 'delete' from 127.0.0.1:36296 (elapsed time 0.000505 sec) *2025-08-07T07:47:30.169042099Z 2(45) INFO: <script>: Failure: <null> 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found* 2025-08-07T07:47:30.265075785Z 9(52) INFO: <script>: START: INVITE from sip:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:30.265125134Z 9(52) INFO: <script>: Current Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;transport=ws;ob> 2025-08-07T07:47:30.265135557Z 9(52) INFO: <script>: Setting new Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;transport=ws;ob;alias=192.168.2.185~44816~6;alias=192.168.2.185~44816~6> 2025-08-07T07:47:30.265484502Z 9(52) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:**72@speedport.ip 2025-08-07T07:47:30.265664116Z 9(52) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:**72@speedport.ip, method:INVITE, status:<null>, extra_id: z9hG4bK16860370, rtpengine_manage: replace-origin replace-session-connection via-branch=extra rtcp-mux-demux SDES-off ICE=remove RTP/AVP 2025-08-07T07:47:30.265819591Z INFO: [412n0o8dedgair81t70p]: [control] Received command 'offer' from 127.0.0.1:50790 2025-08-07T07:47:30.265979425Z NOTICE: [412n0o8dedgair81t70p]: [core] Creating new call 2025-08-07T07:47:30.267497852Z INFO: [412n0o8dedgair81t70p]: [control] Replying to 'offer' from 127.0.0.1:50790 (elapsed time 0.001599 sec) 2025-08-07T07:47:30.469565385Z 9(52) INFO: <script>: START: CANCEL from sip:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:30.469629711Z 9(52) INFO: <script>: Current Contact header: <null> 2025-08-07T07:47:30.469645116Z 9(52) INFO: <script>: Setting new Contact header: <null> 2025-08-07T07:47:41.207672567Z 7(50) INFO: <script>: WebSocket connection from 192.168.2.185:44816 has closed 2025-08-07T07:47:57.917754216Z 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found 2025-08-07T07:48:00.230412646Z 2(45) INFO: <script>: BRANCH FAILED: z9hG4bK1686037 + 0INFO: [412n0o8dedgair81t70p]: [control] Received command 'delete' from 127.0.0.1:36296 2025-08-07T07:48:00.230457143Z INFO: [412n0o8dedgair81t70p]: [core] Deleting call branch '' (via-branch 'z9hG4bK16860370') 2025-08-07T07:48:00.230466718Z INFO: [412n0o8dedgair81t70p]: [core] Call branch '' (via-branch 'z9hG4bK16860370') deleted, no more branches remaining 2025-08-07T07:48:00.230475640Z INFO: [412n0o8dedgair81t70p]: [core] Deleting entire call 2025-08-07T07:48:00.230483968Z INFO: [412n0o8dedgair81t70p]: [core] Final packet stats: 2025-08-07T07:48:00.230491909Z INFO: [412n0o8dedgair81t70p]: [core] --- Tag 'hc7h5imu96', created 0:30 ago for branch '' 2025-08-07T07:48:00.230518917Z INFO: [412n0o8dedgair81t70p]: [core] --- subscribed to '' 2025-08-07T07:48:00.230528302Z INFO: [412n0o8dedgair81t70p]: [core] --- subscription for '' 2025-08-07T07:48:00.230537387Z INFO: [412n0o8dedgair81t70p]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec 2025-08-07T07:48:00.230545867Z INFO: [412n0o8dedgair81t70p]: [core] --------- Port 192.168.2.209:23400 <> 192.168.2.185:46431, SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:48:00.230554949Z INFO: [412n0o8dedgair81t70p]: [core] --- Tag '', created 0:30 ago for branch 'z9hG4bK16860370' 2025-08-07T07:48:00.230563124Z INFO: [412n0o8dedgair81t70p]: [core] --- subscribed to 'hc7h5imu96' 2025-08-07T07:48:00.230570851Z INFO: [412n0o8dedgair81t70p]: [core] --- subscription for 'hc7h5imu96' 2025-08-07T07:48:00.230578827Z INFO: [412n0o8dedgair81t70p]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec 2025-08-07T07:48:00.230587091Z INFO: [412n0o8dedgair81t70p]: [core] --------- Port 192.168.2.209:23494 <> :0 , SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:48:00.230595631Z INFO: [412n0o8dedgair81t70p]: [core] --------- Port 192.168.2.209:23495 <> :0 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:48:00.230603868Z INFO: [412n0o8dedgair81t70p]: [control] Replying to 'delete' from 127.0.0.1:36296 (elapsed time 0.000253 sec) *2025-08-07T07:48:00.230753116Z 2(45) INFO: <script>: Failure: <null> 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found* --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- especially the bold lines confuse me. Why is the TCP/TLS closed and can not be found ? Could that be the problem ?* * * * Thanks for some help and best regards Peter* *

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sr-users April 2026