### Description
Hi!
We are in the progress of upgrading from Kamailio 5.5 to 5.8. During our testing we have noticed a new error being reported from Kamailio. We don’t see any other errors following it.
```jsx
/usr/sbin/kamailio[201]: CRITICAL: <core> [core/tcp_main.c:5544]: tcp_timer_check_connections(): message processing timeout on connection id: 67896 (state: 3) - closing
```
It does seem to be [new code](https://github.com/kamailio/kamailio/blob/master/src/core/tcp_main.c#… in Kamailio reporting this issue.
Given that this is a fairly expected thing, cleaning up a connection which receives no traffic within the given time, is there a need for it to be reported on CRITICAL?
I’d also expect it to be caught by
```
event_route[tcp:timeout] {
xlog("L_INFO","connection $conid timeouts (unanswered keepalives)");
}
```
given that [the description](https://www.kamailio.org/docs/modules/stable/modules/tcpops.ht… of this one is `Called for connection timeouts (unanswered keepalives).`.
### Troubleshooting
We don't have any way to reproduce it, we are still investigating it to figure out the cause. It happens around every 2 hours, so we think there might some some scheduled test or job running in our test system causing this.
### Additional Information
* **Kamailio Version** - 5.8.3
* **Operating System**: Debian 12 AMI from the AWS Marketplace.
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### Description
I encountered an issue with a specific device that is receiving a T.38 fax. The call gets established towards the device normally with inband media, then the device requests T.38 media and tries to remove the audio media via sending port 0 in the ReInvite SDP. Also it sends connection-information only on the media level and there is no "c=" on the session level.
Reading through the related RFCs and the kamailio code and discussing this on the mailing list, we came to the conclusion that kamailio is doing everything correct as per RFC. Though we were wondering if it would be possible to adjust the parser to be not as strict as it is right now, and allow a missing "c=" line on the media-level if the stream is removed/rejected via port 0, since i see no sense in requiring connection information. Of course we are also in contact with the vendor to hopefully adjust on their side.
Here are the related RFCs:
1. RFC8866 5.7
`A session description MUST contain either at least one "c=" line in each media description or a single "c=" line at the session level. It MAY contain a single session-level "c=" line and additional media-level "c=" line(s) per-media-description, in which case the media-level values override the session-level settings for the respective media.`
2. RFC3264 8.2
`Existing media streams are removed by creating a new SDP with the port number for that stream set to zero. The stream description MAY omit all attributes present previously, and MAY list just a single media format.`
At first i was especially confused by the RFC3264 8.2 part, since it seemed correct what the device is sending, but if you read carefully and keep the wording for SDP in mind only attributes ("a=") MAY be omitted. So a "c=" line should still be in the SDP for the removed media if it's not included on the session-level. Or do you see this differently?
### Expected behavior
Kamailio allows and parses the SDP when there is no session-wide "c=", a media stream is being removed via port zero and there is no "c=" for this media stream and only the remaining media streams include a "c=" line.
#### Actual observed behavior
Kamailio throws an error when trying to parse the SDP.
#### Log Messages
```
<core> [core/parser/sdp/sdp.c:523]: parse_sdp_session(): can't find media IP in the message
```
#### SIP Traffic
```
v=0
o=xmserver 1726638425 1726638427 IN IP4 169.254.1.1
s=xmserver
t=0 0
m=audio 0 RTP/AVP 8
m=image 56002 udptl t38
c=IN IP4 169.254.1.1
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
```
### Possible Solutions
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 5.8.3 (x86_64/linux)
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_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: unknown
compiled with gcc 12.2.0
```
* **Operating System**:
```
Debian 12
Linux hostname 6.1.0-20-amd64 #1 SMP PREEMPT_DYNAMIC Debian 6.1.85-1 (2024-04-11) x86_64 GNU/Linux
```
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### Description
Some UAC (notably Aastra Phones) use the sips: uri scheme without ;transport=tls when TLS is being used.
I observed kamailio 5.7.6 and most probably also 5.8.4 to insert the sips URI in the wrong Record-Route header causing replies not to reach the destination.
#### Reproduction
Find a UAC which uses the sips URI scheme when configured to use TLS. (AAstra IP Phones for example).
Important: The Registration looks like this, contains no transport attribute but the sips scheme with port 5061 and a path pointing to the SBC.
```
"Contact" : {
"Address" : "sips:315996608@157.161.4.237:5061",
"CFlags" : 0,
"CSeq" : 1446407115,
"Call-ID" : "3abf8e6af6391e7b",
"Expires" : 56,
"Flags" : 0,
"Instance" : "<urn:uuid:00000000-0000-1000-8000-00085D49FB42>",
"KA-Roundtrip" : 0,
"Keepalive" : 0,
"Last-Keepalive" : 1735827284,
"Last-Modified" : 1735827284,
"Methods" : 16095,
"Path" : "<sip:157.161.23.249;lr;r2=on>,<sip:157.161.23.249:5061;transport=tls;lr;r2=on>",
"Q" : 0.44,
"Received" : "[not set]",
"Reg-Id" : 0,
"Ruid" : "uloc-677695ae-20472d-23",
"Server-Id" : 0,
"Socket" : "udp:157.161.23.4:5060",
"State" : "CS_DIRTY",
"Tcpconn-Id" : -1,
"User-Agent" : "Aastra 6869i/6.3.0.1020"
}
```
Using this set-up:
UAC (TLS) - Kamailio SBC (rtpengine / nat) udp - Kamailio Registrar udp - Core Infrastructure
Call the UAC. Each hop is adding a Record-Route header which is then used by the UAC when creating a new transaction within the dialog.
In this example:
Transaction 1:
CORE => UAC: INVITE 100rel supported
UAC => CORE: 180 Ringing 100rel required
---
Transaction 2:
CORE => UAC: PRACK
During the first transaction, Record-Route header are accumulated to indicate the Route set to be taken by the in-dialog PRACK request.
In the INVITE, the UAC is presented with those Record-Route Header:
The first two header are added by the SBC (IP .249) with r2=on to indicate the change from UDP to TLS
The third header was added by the Registrar (IP .5)
The fourth header was added by the Core (IP .2)
```
02.01.2025 15:17:54.250 +01:00: 157.161.23.249:5061 -> 157.161.4.237:5061
INVITE sips:315996608@157.161.4.237:5061 SIP/2.0
Record-Route: <sips:157.161.23.249:5061;transport=tls;r2=on;lr;ftag=3944816274-1416360442>
Record-Route: <sips:157.161.23.249;r2=on;lr;ftag=3944816274-1416360442>
Record-Route: <sips:157.161.23.5;lr;ftag=3944816274-1416360442>
Record-Route: <sip:157.161.23.2;lr;ftag=3944816274-1416360442;did=e77.89e5>
```
UAC is ringing:
```
02.01.2025 15:17:54.368 +01:00: 157.161.4.237:5061 -> 157.161.23.249:5061
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 157.161.23.249:5061;branch=z9hG4bK5b72.d42e9bf4e341f690d8b383a0a96edeb3.0
Via: SIP/2.0/UDP 157.161.23.5;rport=5060;branch=z9hG4bK5b72.621bba23d60b529f876b92bf2ab718de.0
Via: SIP/2.0/UDP 157.161.23.2;branch=z9hG4bK5b72.2d88c4e6630f19da5ae24efb523564bf.0
Via: SIP/2.0/UDP 157.161.10.230:5060;branch=z9hG4bK100567609d7e7e9517ddfb1eaa7d684c
Record-Route: <sips:157.161.23.249:5061;transport=tls;r2=on;lr;ftag=3944816274-1416360442>, <sips:157.161.23.249;r2=on;lr;ftag=3944816274-1416360442>, <sips:157.161.23.5;lr;ftag=3944816274-1416360442>, <sip:157.161.23.2;lr;ftag=3944816274-1416360442;did=e77.89e5>
[...]
Contact: "Somebody" <sips:315996608@157.161.4.237:5061>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D49FB42>"
```
Origin of call takes this Record-Route to create the source Route header for new PRACK transaction:
```
02.01.2025 15:17:54.393 +01:00: 157.161.10.230:5060 -> 157.161.23.2:5060
PRACK sips:315996608@157.161.4.237:5061 SIP/2.0
Max-Forwards: 66
Route: <sip:157.161.23.2;lr;ftag=3944816274-1416360442;did=e77.89e5>
Route: <sips:157.161.23.5;lr;ftag=3944816274-1416360442>
Route: <sips:157.161.23.249;r2=on;lr;ftag=3944816274-1416360442>
Route: <sips:157.161.23.249:5061;transport=tls;r2=on;lr;ftag=3944816274-1416360442>
RAck: 1 1 INVITE
```
According to this route set, the call shall be routed from 157.161.23.2 to 157.161.23.5 via 'sips'
This fails as there is no TLS socket bound between those two instances.
`prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket)`
IMHO the issue is caused by the wrong URI scheme (sips instead of sip) being used when the registrar is adding it's Record-Route header. (And maybe also by the SBC when adding the double (r2=on) RR to indicate the change of transport.
I suspect this happens, because the UAC has registered it's contact with the sips uri scheme.
Repeating the same situation with an UAC which registers a sip:user@domain:5061;transport=tls URI when using TLS demonstrates this works fine. It only happens when an UAC uses the sips URI scheme.
If any more information is required like a SIP trace of the issue, I am glad to provide it.
-Benoît-
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Hello all,
As in previous years, I'm organising the VoIP dinner for FOSDEM (1 Feb @
19:00)
However unlike previous years (since the restaurant got upset about
splitting the bill 30 ways). I will ask everyone to order their meal and
pay in advance. (and I will pay the bill to the restaurant with my card)
If you are wanting to attend the dinner please fill out the following form
and follow the instructions for payment.
https://form.jotform.com/250012543310033
Contact details are also in the form if you want to reach out to me with
any further questions.
Kind regards
Torrey
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### Description
When starting Kam inside a docker container and the `auto_bind_ipv6` option is enabled, kam fails to start, see below for error logs.
Instead, inside the very same container, by executing it from a shell, it works. The difference is that starting from the entrypoint kam has pid 1, while starting from a shell as a different pid.
### Troubleshooting
#### Reproduction
Start kamailio inside a docker container,with network=host and the `auto_bind_ipv6` option enabled.
#### Log Messages
<!--
Check the syslog file and if there are relevant log messages printed by Kamailio, add them next, or attach to issue, or provide a link to download them (e.g., to a pastebin site).
-->
```
2025-01-02T15:58:29.655278523Z 0(1) DEBUG: <core> [core/socket_info.c:1764]: add_interfaces(): If: lo Fam: 2 Flg: 10049 Adr: 127.0.0.1
2025-01-02T15:58:29.655282371Z 0(1) DEBUG: <core> [core/socket_info.c:1764]: add_interfaces(): If: eth0 Fam: 2 Flg: 11043 Adr: 188.34.182.80
2025-01-02T15:58:29.655286664Z 0(1) DEBUG: <core> [core/socket_info.c:1764]: add_interfaces(): If: docker0 Fam: 2 Flg: 1003 Adr: 172.17.0.1
2025-01-02T15:58:29.655290864Z 0(1) DEBUG: <core> [core/socket_info.c:1764]: add_interfaces(): If: br-1653739d678f Fam: 2 Flg: 11043 Adr: 172.18.0.1
2025-01-02T15:58:29.655295002Z 0(1) DEBUG: <core> [core/socket_info.c:1764]: add_interfaces(): If: br-kamnet Fam: 2 Flg: 11043 Adr: 172.23.42.1
2025-01-02T15:58:29.655299578Z 0(1) ERROR: <core> [core/socket_info.c:1349]: nl_bound_sock(): could not bind NETLINK sock to sockaddr_nl
2025-01-02T15:58:29.655303602Z 0(1) ERROR: <core> [core/socket_info.c:1666]: add_interfaces_via_netlink(): Could not get network interface list
2025-01-02T15:58:29.655482917Z 0(1) ERROR: <core> [core/socket_info.c:1789]: fix_hostname(): could not resolve 'matteo-ord-devel'
2025-01-02T15:58:29.655515080Z 0(1) ERROR: <core> [core/socket_info.c:2392]: fix_all_socket_lists(): fix_socket_list udp failed
2025-01-02T15:58:29.655519918Z failed to initialize list addresses
```
### Possible Solutions
I think that it boils down to the function [nl_bound_sock](https://github.com/kamailio/kamailio/blob/master/src/core/so… function, which sets the netlink struct Port ID with `getpid()` ( https://github.com/kamailio/kamailio/blob/master/src/core/socket_info.c#L13… ).
For some reason this does not work when kam is forking from pid 1.
According to [man 7 netlink](https://man7.org/linux/man-pages/man7/netlink.7.html) should be safe to set that pid to 0, and let the kernel assign one, since if assigned from the process, it must guarantee that is unique if when multiple sockets are requested.
I've tried to set that port id to `0` and works correctly:
```diff
diff --git a/src/core/socket_info.c b/src/core/socket_info.c
index 2557a83b1e..662a3741c2 100644
--- a/src/core/socket_info.c
+++ b/src/core/socket_info.c
@@ -1343,7 +1343,7 @@ static int nl_bound_sock(void)
bzero(&la, sizeof(la));
la.nl_family = AF_NETLINK;
la.nl_pad = 0;
- la.nl_pid = getpid();
+ la.nl_pid = 0;
la.nl_groups = 0;
if(bind(sock, (struct sockaddr *)&la, sizeof(la)) < 0) {
LM_ERR("could not bind NETLINK sock to sockaddr_nl\n");
```
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
5.8.4
* **Operating System**:
<!--
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Kernel details (output of `lsb_release -a` and `uname -a`)
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Docker container based on ubuntu noble.
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Creating an empty file `src/modules/topos_htable/README` allows the installation to complete without errors.
```
$ ninja install
[..]
CMake Error at src/modules/cmake_install.cmake:1588 (file):
file INSTALL cannot find
"/usr/src/kamailio/src/modules/topos_htable/README":
No such file or directory.
Call Stack (most recent call first):
src/cmake_install.cmake:96 (include)
cmake_install.cmake:47 (include)
FAILED: CMakeFiles/install.util
cd /usr/src/kamailio/build && /usr/bin/cmake -P cmake_install.cmake
ninja: build stopped: subcommand failed.
$ touch ../src/modules/topos_htable/README && ninja install && echo $?
[..]
0
```
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Module: kamailio
Branch: master
Commit: fc182198c7182759cc2c10b4afa3569336a438c9
URL: https://github.com/kamailio/kamailio/commit/fc182198c7182759cc2c10b4afa3569…
Author: Daniel-Constantin Mierla <miconda(a)gmail.com>
Committer: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2025-01-03T12:02:21+01:00
presence_dfks: added README file
- initial import, it is going to be autmatically generated from now on
---
Added: src/modules/presence_dfks/README
---
Diff: https://github.com/kamailio/kamailio/commit/fc182198c7182759cc2c10b4afa3569…
Patch: https://github.com/kamailio/kamailio/commit/fc182198c7182759cc2c10b4afa3569…
---
diff --git a/src/modules/presence_dfks/README b/src/modules/presence_dfks/README
new file mode 100644
index 00000000000..d965a6e4c6a
--- /dev/null
+++ b/src/modules/presence_dfks/README
@@ -0,0 +1,78 @@
+presence_dfks Module
+
+Maja Stanislawska
+
+ <maja.stanislawska(a)yahoo.com>
+
+Victor Seva
+
+ Sipwise GmbH
+ <vseva(a)sipwise.com>
+
+Edited by
+
+Victor Seva
+
+ Sipwise GmbH
+ <vseva(a)sipwise.com>
+
+ Copyright �� 2014 Maja Stanislawska
+
+ Copyright �� 2024 Victor Seva
+ __________________________________________________________________
+
+ Table of Contents
+
+ 1. Admin Guide
+
+ 1. Overview
+ 2. Dependencies
+
+ 2.1. Kamailio Modules
+ 2.2. External Libraries or Applications
+
+ 3. Parameters
+
+Chapter 1. Admin Guide
+
+ Table of Contents
+
+ 1. Overview
+ 2. Dependencies
+
+ 2.1. Kamailio Modules
+ 2.2. External Libraries or Applications
+
+ 3. Parameters
+
+1. Overview
+
+ This module for handling as-feature-event presence messages Provides
+ support for 'Device FeatureKey Synchronization', as described in
+ http://community.polycom.com/polycom/attachments/polycom/VoIP/2233/1/De
+ viceFeatureKeySynchronizationFD.pdf This 'protocol' was developed at
+ Broadsoft, and is used in Linksys/Cisco and Polycom phones and probably
+ some other too. It allows to set up features like dnd (No Not Disturb)
+ and call forwarding using phone interface and have that status updated
+ on aplication server/ proxy or otherway, set those features on
+ aplication server using some gui, and have this data provisioned on
+ phone. For Shared Line Appreance application this let You share status
+ on all phones.
+
+2. Dependencies
+
+ 2.1. Kamailio Modules
+ 2.2. External Libraries or Applications
+
+2.1. Kamailio Modules
+
+ The following modules must be loaded before this module:
+ * sl. presence. pua.
+
+2.2. External Libraries or Applications
+
+ The following libraries or applications must be installed before
+ running Kamailio with this module loaded:
+ * libxml2 - xml handling library.
+
+3. Parameters