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
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__________________________________________________________
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Important: keep the mailing list in the recipients, do not reply only to the
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Important: keep the mailing list in the recipients, do not reply only to the
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Important: keep the mailing list in the recipients, do not reply only to the
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