nmreistd created an issue (kamailio/kamailio#4247)
<!--
Kamailio Project uses GitHub Issues only for bugs in the code or feature requests. Please use this template only for bug reports.
If you have questions about using Kamailio or related to its configuration file, ask on sr-users mailing list:
* https://lists.kamailio.org/mailman3/postorius/lists/sr-users.lists.kamailio…
If you have questions about developing extensions to Kamailio or its existing C code, ask on sr-dev mailing list:
* https://lists.kamailio.org/mailman3/postorius/lists/sr-dev.lists.kamailio.o…
Please try to fill this template as much as possible for any issue. It helps the developers to troubleshoot the issue.
Note that an issue report may be closed automatically after about 2 months
if there is no interest from developers or community users on pursuing it, being
considered expired. In such case, it can be reopened by writing a comment that includes
the token `/notexpired`. About two weeks before considered expired, the issue is
marked with the label `stale`, trying to notify the submitter and everyone else
that might be interested in it. To remove the label `stale`, write a comment that
includes the token `/notstale`. Also, any comment postpone the `expire` timeline,
being considered that there is interest in pursuing the issue.
If there is no content to be filled in a section, the entire section can be removed.
You can delete the comments from the template sections when filling.
You can delete next line and everything above before submitting (it is a comment).
-->
### Description
rtpengine module is now trying to ping previous permanently disabled nodes after commits:
```
* e183a3e25c rtpengine: Re-enable down servers (but not disabled ones)
* 3823479733 rtpengine: Add timer to ping rtpengine
```
### Troubleshooting
#### Reproduction
Just create a set of nodes and also set at least one to be permanently disabled. After that do RPC `rtpengine.reload` or equivelent and after a `rtpengine.show all` and `disabled: 1(permanent)` will never show as before the aforementioned additions.
#### Debugging Data
<!--
If you got a core dump, use gdb to extract troubleshooting data - full backtrace,
local variables and the list of the code at the issue location.
gdb /path/to/kamailio /path/to/corefile
bt full
info locals
list
If you are familiar with gdb, feel free to attach more of what you consider to
be relevant.
-->
```
(paste your debugging data here)
```
#### 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).
-->
```
(paste your log messages here)
```
#### SIP Traffic
<!--
If the issue is exposed by processing specific SIP messages, grab them with ngrep or save in a pcap file, then add them next, or attach to issue, or provide a link to download them (e.g., to a pastebin site).
-->
```
(paste your sip traffic here)
```
### Possible Solutions
<!--
If you found a solution or workaround for the issue, describe it. Ideally, provide a pull request with a fix.
-->
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
Tested between `kamailio 5.8.2` and `6.0.1`
```
* **Operating System**:
<!--
Details about the operating system, the type: Linux (e.g.,: Debian 8.4, Ubuntu 16.04, CentOS 7.1, ...), MacOS, xBSD, Solaris, ...;
Kernel details (output of `lsb_release -a` and `uname -a`)
-->
```
debian bookworm
```
--
Reply to this email directly or view it on GitHub:
https://github.com/kamailio/kamailio/issues/4247
You are receiving this because you are subscribed to this thread.
Message ID: <kamailio/kamailio/issues/4247(a)github.com>
IgorrG created an issue (kamailio/kamailio#4266)
### Description
While implementing mid-register inspired by pull request (https://github.com/kamailio/kamailio/pull/3360) we found the case when it goes impossible to handle responses for registrar servers. In case some of registrars busy and respond with 401 after 200ok received - it is not possible using any routes and configuration options to handle responses received in branch transactions.
According to documentation I think that setting wt_timer should allow to absorb and handle responses on failure-route while transaction kept in memory. In fact wt_timer does not affect handling responses in any way.

### Troubleshooting
To reproduce this case we have configured sipp to response instant with 200ok on registration request.
#### Reproduction
To reproduce issue SIPP could be setup as one of the dispatcher nodes with instant 200ok reply to register. Kamailio should use following configuration snippet: https://github.com/ovoshlook/kamailio-mid-registrar-config-snippets/blob/ma…
#### Log Messages
REGISTER|401tm [t_lookup.c:912]: t_reply_matching(): t_reply_matching: hash 12337 label 0 branch 1
REGISTER|401tm [t_lookup.c:986]: t_reply_matching(): reply (0x7f1fc67f83a8) matched an active transaction (T=0x7f1fc0f0f740)!
REGISTER|401tm [t_lookup.c:1122]: t_check_msg(): msg (0x7f1fc67f83a8) id=13/1915035 global id=13/1915035 T end=0x7f1fc0f0f740
REGISTER|401tm [t_reply.c:2363]: reply_received(): transaction found - T:0x7f1fc0f0f740 branch:1
REGISTER|401tm [t_reply.c:2376]: reply_received(): original status uas=200, uac[1]=0 local=0 is_invite=0)
REGISTER|401tm [t_reply.c:1363]: t_should_relay_response(): ->>>>>>>>> T_code=200, new_code=401
REGISTER|401tm [t_reply.c:1374]: t_should_relay_response(): final reply already sent
REGISTER|401tm [t_reply.c:1625]: t_should_relay_response(): finished with rps discarded - uas status: 200
### Possible Solutions
<!--
If you found a solution or workaround for the issue, describe it. Ideally, provide a pull request with a fix.
-->
### Additional Information
* **Kamailio Version**
```
version: kamailio 5.6.4 (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, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 16:35:08 Jun 5 2023 with gcc 10.2.1
```
* **Operating System**:
```
No LSB modules are available.
Distributor ID: Debian
Description: Debian GNU/Linux 11 (bullseye)
Release: 11
Codename: bullseye
Linux pbxKamaTest 5.10.0-23-amd64 #1 SMP Debian 5.10.179-1 (2023-05-12) x86_64 GNU/Linux
```
--
Reply to this email directly or view it on GitHub:
https://github.com/kamailio/kamailio/issues/4266
You are receiving this because you are subscribed to this thread.
Message ID: <kamailio/kamailio/issues/4266(a)github.com>
jaspreet-eltropy created an issue (kamailio/kamailio#4270)
**I’m running into an issue when using Yealink SIP phones in a setup involving Kamailio + RTPengine + FreeSWITCH/Fusionpbx.**
**Setup Overview:**
SIP Proxy: Kamailio (handling NAT traversal, registration, and routing)
Media Relay: RTPengine (with replace-origin, replace-session-connection, media-address, and other options)
Media Server: FreeSWITCH (handles RTP/audio and call logic)
Phone: Yealink T54w Series (but likely affects other models too)
Transport: UDP (with force_rport() and NAT detection routes in place)
Environment: Mixed NAT/public IPs with STUN/TURN enabled on WebRTC clients(SIP.js) (where applicable)
Problem Description:
When a call is placed from/to a Yealink phone, the call signaling works fine (INVITE, 200 OK, ACK are all seen correctly), but there is no audio or sometimes one-way audio. This only happens with Yealink; other SIP clients (Zoiper, Linphone, SIP.js etc.) work fine with the same setup.
From packet captures and Kamailio logs, I can see the following:
RTPengine receives the call and processes it.
The rtpengine_manage() is called with these flags:
May 30 05:38:13 ip-10-7-37-179 /usr/local/sbin/kamailio[62354]: INFO: <script>: BRIDGING: TRYING TO BRIDGE udp sip:5555@192.168.1.10:12561;transport=TCP
May 30 05:38:13 ip-10-7-37-179 /usr/local/sbin/kamailio[62354]: INFO: <script>: NATMANAGE branch_id:0 ruri: sip:5555@192.168.1.10:12561;transport=TCP, method:INVITE, status:<null>, extra_id: z9hG4bKcSBpr6vNyyepe0, rtpengine_manage: replace-origin replace-session-connection media-address=52.24.225.81 trust-address via-branch=extra
The Yealink phone sends SDP with correct IPs, but it doesn’t receive RTP (confirmed via tcpdump).
Interestingly, if I bypass RTPengine + kamilio (media goes directly to/from FreeSWITCH), audio works.
What I’ve Tried:
Enabling/disabling trust-address and via-branch options.
Using set_contact_alias() for REGISTER/NATed clients.
Ensured that Contact header aliasing is applied.
Verified that Yealink NAT settings are configured correctly.
Using force_rport() and fix_nated_register() in Kamailio.
Ensured RTP ports are open and media_address in RTPengine is set correctly.
My Questions:
Is there a known compatibility issue with Yealink phones and RTPengine in relayed mode?
Does Yealink require specific NAT handling rules that differ from softphones?
Are there any recommended rtpengine_manage() flags or Kamailio routing practices specifically for Yealink endpoints?
Any suggestions to debug further — SIP traces seem clean, but RTP is not received by the Yealink.
Sample Logs (Kamailio):
NATMANAGE branch_id:0 ruri: <null>, method:INVITE, status:100, extra_id: z9hG4bK4b1.ffbb20e21cff840bb08800b46398f55b.00, rtpengine_manage: replace-origin replace-session-connection media-address=52.24.225.81 trust-address via-branch=extra
Yealink SIP Config:
NAT Traversal: Enabled (STUN)
Keep-Alive: 30s
UDP port: 5060
SIP registration: works
Audio: not working when RTPengine is in the path
Any help or suggestions would be greatly appreciated. Let me know if logs or traces are needed.
```
-[ RECORD 1 ]-+----------------------------------------------------------------------
id | 3259
ruid | uloc-683943f5-f3a0-1
username | 5555
domain |
contact | sip:5555@192.168.1.10:12561;transport=TCP
received | sip:182.77.49.201:12561;transport=tcp
path |
expires | 2025-05-30 05:44:04
q | -1
callid | [2_689921921@192.168.1.10](mailto:2_689921921@192.168.1.10)
cseq | 6
last_modified | 2025-05-30 05:42:04
flags | 0
cflags | 64
user_agent | Yealink SIP-T54W 96.86.0.75
socket | tcp:10.7.37.179:5060
methods | 16383
instance |
reg_id | 0
server_id | 0
connection_id | 4
keepalive | 1
partition | 28
```
Thanks in advance
--
Reply to this email directly or view it on GitHub:
https://github.com/kamailio/kamailio/issues/4270
You are receiving this because you are subscribed to this thread.
Message ID: <kamailio/kamailio/issues/4270(a)github.com>