Re: problems using kamailio
peter walber via sr-users <[email protected]>
| Newsgroups | gmane.comp.voip.ser |
|---|---|
| Message-ID | <[email protected]> |
Hello, I have tried the setup without docker. I have installed and started rtpengine . I believe, it is successful. ------------------------------------------------------------------------------------------------------------------ *peter@asterisk-PBX:~$ sudo systemctl status rtpengine * ●rtpengine-daemon.service - RTP/media Proxy Daemon Loaded: loaded (/lib/systemd/system/rtpengine-daemon.service; enabled; preset: enabled) Active: active (running)since Mon 2025-08-11 12:53:50 CEST; 10min ago Docs: man:rtpengine(8) Process: 635848 ExecStartPre=/usr/sbin/rtpengine-iptables-setup start (code=exited, status=0/SUCCESS) Main PID: 635866 (rtpengine) Tasks: 20(limit: 6857) Memory: 11.4M CPU: 209ms CGroup: /system.slice/rtpengine-daemon.service └─635866 /usr/bin/rtpengine -f -E --no-log-timestamps --pidfile /run/rtpengine/rtpengine-daemon.pid --config-file /etc/rtpengine/rtpen> Aug 11 12:53:49 asterisk-PBX systemd[1]: Starting rtpengine-daemon.service - RTP/media Proxy Daemon... Aug 11 12:53:50 asterisk-PBX rtpengine[635866]: INFO: [crypto] Generating new DTLS certificate Aug 11 12:53:50 asterisk-PBX rtpengine[635866]: INFO: [core] Startup complete, version 10.5.3.5-1 Aug 11 12:53:50 asterisk-PBX systemd[1]: Started rtpengine-daemon.service - RTP/media Proxy Daemon. Aug 11 12:55:20 asterisk-PBX rtpengine[635866]: INFO: [control] Received command 'ping' from 127.0.0.1:50208 Aug 11 12:55:20 asterisk-PBX rtpengine[635866]: INFO: [control] Replying to 'ping' from 127.0.0.1:50208 (elapsed time 0.000001 sec) lines 1-18/18 (END) ------------------------------------------------------------------------------------ I have installed kamailio using the command sudo apt-get install kamailio kamailio-mysql-modules kamailio-websocket-modules kamailio-tls-modules Then copied the privkey.pem and fullchain.pem to /etc/kamailio. The config file from the florian-h05 project to /etc/kamailio. Kamailio at least seems to start sucessfully. ---------------------------------------------------------------------------- * * *peter@asterisk-PBX:~**$ sudo systemctl status kamailio* ●kamailio.service - Kamailio - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; preset: enabled) Active: active (running)since Mon 2025-08-11 12:55:20 CEST; 12min ago Docs: man:kamailio(8) Process: 635920 ExecStart=/usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY (code=exited, status=0/SUC> Main PID: 635922 (kamailio) Tasks: 17(limit: 6857) Memory: 16.3M CPU: 1.164s CGroup: /system.slice/kamailio.service ├─635922 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635923 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635924 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635925 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635926 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635927 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635928 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635929 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635930 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635931 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635932 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635933 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635934 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635935 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 ├─635936 /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 8 lines 1-25 __________________________________________________________________________ But it doesn't work. My client (openhab widget based on JsSIP, **[email protected]) doesn't establish a connection to the doorbird **[email protected]. And it seems to be the same problem like the docker version has. The websocket from the client side seems to close immediately. The SIP Client of my doorbird (**[email protected]) definitly works. I can call it from zoiper successfully. The option #!WITH_DEBUG also doesn't give me a hint. ----------------------------------------------------------------------------- peter@asterisk-PBX:~$ sudo journalctl -r | grep kamailio | more *Aug 11 12:56:24 asterisk-PBX /usr/sbin/kamailio[635932]: INFO: <script>: WebSocket connection from 192.168.2.151:36894 has closed * Aug 11 12:56:17 asterisk-PBX /usr/sbin/kamailio[635936]: INFO: <script>: START: REGISTER from sip:**[email protected] (IP:192.168.2.151:36894) Aug 11 12:56:01 asterisk-PBX /usr/sbin/kamailio[635935]: INFO: <script>: START: ACK from sip:**[email protected] (IP:192.168.2.151:36894) Aug 11 12:56:01 asterisk-PBX /usr/sbin/kamailio[635935]: INFO: <script>: START: INVITE from sip:**[email protected] (IP:192.168.2.151:36894) Aug 11 12:55:22 asterisk-PBX /usr/sbin/kamailio[635934]: INFO: <script>: START: REGISTER from sip:**[email protected] (IP:192.168.2.151:36894) Aug 11 12:55:22 asterisk-PBX /usr/sbin/kamailio[635934]: INFO: <script>: HTTP Request Received Aug 11 12:55:20 asterisk-PBX systemd[1]: Started kamailio.service - Kamailio - the Open Source SIP Server. Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635930]: INFO: ctl [io_listener.c:213]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635923]: INFO: rtpengine [rtpengine.c:2929]: rtpp_test(): rtpengine instance <udp:127.0.0.1:22222> found, support for it enabled Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:735]: set_verification(): TLSc<default>: Server MUST present valid certificate Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:389]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:386]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:382]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:379]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:372]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:364]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:361]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: ca_path='(null)' Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:328]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:750]: set_verification(): TLSs<default>: No client certificate required and no checks performed Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: NOTICE: tls [tls_domain.c:1144]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... Aug 11 12:55:20 asterisk-PBX /usr/sbin/kamailio[635922]: INFO: tls [tls_domain.c:389]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 ---------------------------------------------------------------------- My openhab client doesn't register successfully at the speedport smart router. When I open the network console of the browser (ctrl-shift-J) I see the following when I try to dial **[email protected] --------------------------------------------------- ACK sip:**[email protected] SIP/2.0 Via: SIP/2.0/WSS vd1ol3bgpp3k.invalid;branch=z9hG4bK1779381 Max-Forwards: 69 To: <sip:**[email protected]>;tag=5532e97d005227a861814c798ae9fc63-f7e00000 From: <sip:**[email protected]>;tag=69mva3r0pg Call-ID: r4ivvc0u742gafkurin7 CSeq: 3667 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY Supported: outbound User-Agent: JsSIP 3.10.1 Content-Length: 0 +0ms app.725b24ffcb85453fcf4b.js:1 JsSIP:WebSocketInterface send() +2ms app.725b24ffcb85453fcf4b.js:1 JsSIP:RTCSession receiveInviteResponse() +6ms app.725b24ffcb85453fcf4b.js:1 JsSIP:RTCSession session failed +0ms app.725b24ffcb85453fcf4b.js:1 JsSIP:RTCSession emit "_failed" +0ms app.725b24ffcb85453fcf4b.js:1 JsSIP:RTCSession close() +0ms app.725b24ffcb85453fcf4b.js:1 JsSIP:RTCSession close() | closing local MediaStream +1ms app.725b24ffcb85453fcf4b.js:1 JsSIP:RTCSession emit "failed" +0ms *app.06ad2752b5cdbdf8cc97.js:2 oh-sipclient: Call failed. Reason: Not Found* app.725b24ffcb85453fcf4b.js:1 JsSIP:InviteClientTransaction Timer D expired for transaction z9hG4bK1779381 +1m ------------------------------------ * * I don't understand the message *app.06ad2752b5cdbdf8cc97.js:2 oh-sipclient: Call failed. Reason: Not Found* * * *Best regards* * * *Peter * * * Am 08.08.25 um 16:54 schrieb Richard Robson via sr-users: > > the Github link you originally posted with does have a basic > websockets, tls and stp engine script in it > https://github.com/florian-h05/webrtc-sip-gw/blob/main/config/kamailio/kamailio.cfg > > maybe try without Docker and get things running in an environment that > you can change easily. you can then get access to all the ports and > traffic you need without extra docker configuration. > > I use a docker environment for testing, which uses sipp to emulate > traffic and it spits out a set of logs and pcaps in needed, but > again that a set of proffestional level tools, I use just to confirm > the scripts sun as expected. each test needs to be configured and > reads the lastest script I provide. also I need to setup myslq in the > docker to provide the requied configs for the modules being used, > dialplan, drouting, dispatcher, etc. > > try running kamailio on 1 vm and rtpengine on another, then setup the > network for the VMs. have your endpoints connect o the VMs LAN. check > the traffic is working before hand, via Pings and telnet. once your > platform is stable, then try and get the different elements working. > > tls can be a pain to get working initally, make sure you have valid > certs etc. then get RTPengine working and then the websockets. > > I'm afraid Its going to be difficult to aid you further without doing > it all for you. Like Henning said, Its a steep learning curve and can > be pretty daunting. > > Think of Kamailio as a framework not an end solution. You need to > build the end solution from the framework provided. Again think of the > github repo as a starting point for your solution. Once you get the > desired outcome from your testing, you could probably dockerize that > with the egress and ingress ports open to your local LAN. > > > > Hope you get your solution working. > > > Richard > > On 08/08/2025 08:49, Peter Walber via sr-users wrote: >> >> Hi, >> >> >> I can imagine that the problems come from the configuration within >> the docker container. IP address and ports are somehow automatically >> defined and I'm not sure if these paramerts are correct. And I can >> not enter the routing scripts for HOMER intergration in the docker >> container. >> >> Therefore , I want to install and run kamailio with rtpengine >> directly outside a docker container on my debian vm. >> >> But I'm pretty much confused by the default kamailio configuration >> file with all the parameters, directives and loadmodules. >> >> Can somebody send me a basic configuration file for *kamailio with >> websockets, TLS Support and rtpengine. * >> >> Thanks and regards >> >> Peter >> Am 07.08.25 um 19:24 schrieb Richard Robson via sr-users: >>> >>> Hi, >>> >>> You could run up a Homer server and send the traffic to there from >>> Kamailio with the sipcapture module. It should give you the ingress >>> and egress packets unencrypted from the script. >>> >>> Hope that helps >>> >>> Richard >>> >>> On 07/08/2025 17:35, peter walber via sr-users wrote: >>>> >>>> Hello, >>>> >>>> >>>> I can use wireshark to capture and filter packets now. But the >>>> payload ist encrypted and I don't know how to decrypt the app-data. >>>> >>>> I can observe a lot of packets which are trevelling from the SIP >>>> Client to the webrtc-sip-gw. But can not decrypt the payload. >>>> >>>> >>>> I have opened the network console of google chrom. I can see that >>>> the client starts the SIP request using JsSIP >>>> >>>> **72 is my doorbird >>>> >>>> **73 is my openhab SIP Client >>>> >>>> --------------------------------------------------------- >>>> >>>> >>>> INVITE sip:**[email protected] SIP/2.0 >>>> Via: SIP/2.0/WSS r64og133ol2n.invalid;branch=z9hG4bK2639027 >>>> Max-Forwards: 69 >>>> To: <sip:**[email protected]> >>>> From: <sip:**[email protected]>;tag=0l5gn2o04p >>>> Call-ID: k57isvj3jnib42sone96 >>>> CSeq: 6684 INVITE >>>> Contact: <sip:[email protected];transport=ws;ob> >>>> Content-Type: application/sdp >>>> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY >>>> Supported: ice,replaces,outbound >>>> User-Agent: JsSIP 3.10.1 >>>> Content-Length: 1705 >>>> v=0 >>>> o=- 8248900579316048806 2 IN IP4 127.0.0.1 >>>> s=- >>>> t=0 0 >>>> a=group:BUNDLE 0 >>>> a=extmap-allow-mixed >>>> a=msid-semantic: WMS c8e94492-8d01-4aa1-9023-497510a9054b >>>> m=audio 43416 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 >>>> c=IN IP4 192.168.2.151 >>>> a=rtcp:9 IN IP4 0.0.0.0 >>>> a=candidate:1715666646 1 udp 2122260223 192.168.2.151 43416 typ >>>> host generation 0 network-id 1 >>>> a=candidate:2923572520 1 udp 2122194687 172.17.0.1 34664 typ host >>>> generation 0 network-id 2 >>>> a=candidate:2565352002 1 tcp 1518280447 192.168.2.151 9 typ host >>>> tcptype active generation 0 network-id 1 >>>> a=candidate:1357445564 1 tcp 1518214911 172.17.0.1 9 typ host >>>> tcptype active generation 0 network-id 2 >>>> a=ice-ufrag:vZWG >>>> a=ice-pwd:NWYzQwxQ4nP4FdcL6XpCl66/ >>>> a=ice-options:trickle >>>> a=fingerprint:sha-256 >>>> 80:22:F4:1E:B4:94:EF:F8:44:FA:E9:CA:72:DA:75:F3:99:1F:10:8D:B4:11:B3:65:3D:F9:44:67:28:31:CB:E5 >>>> a=setup:actpass >>>> a=mid:0 >>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level >>>> a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time >>>> a=extmap:3 >>>> http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 >>>> a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid >>>> a=sendrecv >>>> a=msid:c8e94492-8d01-4aa1-9023-497510a9054b >>>> 8cd550e3-9b06-4e06-af55-f31b5392bafd >>>> a=rtcp-mux >>>> a=rtcp-rsize >>>> a=rtpmap:111 opus/48000/2 >>>> a=rtcp-fb:111 transport-cc >>>> a=fmtp:111 minptime=10;useinbandfec=1 >>>> a=rtpmap:63 red/48000/2 >>>> a=fmtp:63 111/111 >>>> a=rtpmap:9 G722/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:13 CN/8000 >>>> a=rtpmap:110 telephone-event/48000 >>>> a=rtpmap:126 telephone-event/8000 >>>> a=ssrc:1295466586 cname:7i+lr9FQ7qgsLxhh >>>> a=ssrc:1295466586 msid:c8e94492-8d01-4aa1-9023-497510a9054b >>>> 8cd550e3-9b06-4e06-af55-f31b5392bafd >>>> --------------------------------------------- >>>> >>>> But I also see the obviously kamilio responding with timeout >>>> >>>> ------------------------------------------------------------------------------------------ >>>> >>>> SIP/2.0 408 Request Timeout >>>> Via: SIP/2.0/WSS >>>> r64og133ol2n.invalid;branch=z9hG4bK2315704;rport=58076;received=192.168.2.151 >>>> To: >>>> <sip:**[email protected]>;tag=5532e97d005227a861814c798ae9fc63-c1c00000 >>>> From: <sip:**[email protected]>;tag=kfjl6vah0u >>>> Call-ID: p928jbnb7d2spbvdgoepjt >>>> CSeq: 1 REGISTER >>>> Server: kamailio (5.6.3 (x86_64/linux)) >>>> Content-Length: 0 >>>> -------------------------------------------------------------------- >>>> >>>> does it help to analyse ? >>>> >>>> Regards >>>> >>>> Peter >>>> >>>> >>>> Am 07.08.25 um 14:07 schrieb Henning Westerholt via sr-users: >>>>> >>>>> Hello, >>>>> >>>>> have a look e.g. with tcpdump and wireshark what exactly happens. >>>>> Maybe the connection is closed from one party. Sometimes TCP >>>>> keep-alive or SIP REGISTER time intervals needs to be adapted to >>>>> keep a connection open, e.g. in NAT scenarios. >>>>> >>>>> Cheers, >>>>> >>>>> Henning >>>>> >>>>> *From:*Peter Walber via sr-users <[email protected]> >>>>> *Sent:* Donnerstag, 7. August 2025 09:56 >>>>> *To:* [email protected] >>>>> *Cc:* Peter Walber <[email protected]> >>>>> *Subject:* [SR-Users] problems using kamailio >>>>> >>>>> 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" >>>>> <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:**[email protected] (IP:192.168.2.185:39510) >>>>> 2025-08-07T07:46:57.391516534Z 9(52) INFO: <script>: Current >>>>> Contact header: <sip:[email protected];transport=ws> >>>>> <sip:[email protected];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:**[email protected]:5060> >>>>> <sip:**[email protected]:5060> >>>>> 2025-08-07T07:47:00.195739350Z 9(52) INFO: <script>: START: >>>>> INVITE from sip:**[email protected] (IP:192.168.2.185:39510) >>>>> 2025-08-07T07:47:00.195800424Z 9(52) INFO: <script>: Current >>>>> Contact header: >>>>> <sip:[email protected];transport=ws;ob> >>>>> <sip:[email protected];transport=ws;ob> >>>>> 2025-08-07T07:47:00.195811886Z 9(52) INFO: <script>: Setting new >>>>> Contact header: >>>>> <sip:[email protected];transport=ws;ob;alias=192.168.2.185~39510~6;alias=192.168.2.185~39510~6> >>>>> <sip:[email protected];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:**[email protected] >>>>> 2025-08-07T07:47:00.196181228Z 9(52) INFO: <script>: NATMANAGE >>>>> branch_id:0 ruri: sip:**[email protected], 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:**[email protected] (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:**[email protected] (IP:192.168.2.185:44816) >>>>> 2025-08-07T07:47:27.951663073Z 9(52) INFO: <script>: Current >>>>> Contact header: <sip:[email protected];transport=ws> >>>>> <sip:[email protected];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:**[email protected]:5060> >>>>> <sip:**[email protected]: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:**[email protected] (IP:192.168.2.185:44816) >>>>> 2025-08-07T07:47:30.265125134Z 9(52) INFO: <script>: Current >>>>> Contact header: >>>>> <sip:[email protected];transport=ws;ob> >>>>> <sip:[email protected];transport=ws;ob> >>>>> 2025-08-07T07:47:30.265135557Z 9(52) INFO: <script>: Setting new >>>>> Contact header: >>>>> <sip:[email protected];transport=ws;ob;alias=192.168.2.185~44816~6;alias=192.168.2.185~44816~6> >>>>> <sip:[email protected];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:**[email protected] >>>>> 2025-08-07T07:47:30.265664116Z 9(52) INFO: <script>: NATMANAGE >>>>> branch_id:0 ruri: sip:**[email protected], 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:**[email protected] (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 >>>>> >>>>> >>>>> __________________________________________________________ >>>>> Kamailio - Users Mailing List - Non Commercial Discussions [email protected] >>>>> To unsubscribe send an email [email protected] >>>>> Important: keep the mailing list in the recipients, do not reply only to the sender! >>>> >>>> __________________________________________________________ >>>> Kamailio - Users Mailing List - Non Commercial Discussions [email protected] >>>> To unsubscribe send an email [email protected] >>>> Important: keep the mailing list in the recipients, do not reply only to the sender! >>> >>> __________________________________________________________ >>> Kamailio - Users Mailing List - Non Commercial Discussions [email protected] >>> To unsubscribe send an email [email protected] >>> Important: keep the mailing list in the recipients, do not reply only to the sender! >> >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions [email protected] >> To unsubscribe send an email [email protected] >> Important: keep the mailing list in the recipients, do not reply only to the sender! > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions [email protected] > To unsubscribe send an email [email protected] > Important: keep the mailing list in the recipients, do not reply only to the sender! __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- [email protected] To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender!