nmreistd created an issue (kamailio/kamailio#4247)
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### 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
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#### SIP Traffic
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Tested between `kamailio 5.8.2` and `6.0.1`
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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
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#### Type Of Change
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- [ ] PR should be backported to stable branches
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- [x] Related to issue #4252
#### Description
This PR adds a new helper `reginfo_disable_publish` to `pua_reginfo` module which disabled the publishing for the current message, which is useful to control when to actually send it or not (eg in DMQ replication scenarios).
The helper is "negative" (it *disables* not enables) on purpose in order to mantain backward compatibility to current behaviour.
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/4253
-- Commit Summary --
* pua_reginfo: add helper to disable publishing for current message
-- File Changes --
M src/modules/pua_reginfo/pua_reginfo.c (4)
M src/modules/pua_reginfo/pua_reginfo.h (4)
M src/modules/pua_reginfo/usrloc_cb.c (21)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/4253.patchhttps://github.com/kamailio/kamailio/pull/4253.diff
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- tls_list() add PROTO_WSS to TLS_LIST RPC call to include WSS connections in tls.list
- tls_kill() add PROTO_WSS to handle WSS connections
Co-authored-by: Andreas Tarp <tarp(a)sipgate.de>
#### Pre-Submission Checklist
- [ ] Commit message has the format required by CONTRIBUTING guide
- [ ] Commits are split per component (core, individual modules, libs, utils, ...)
- [ ] Each component has a single commit (if not, squash them into one commit)
- [ ] No commits to README files for modules (changes must be done to docbook files
in `doc/` subfolder, the README file is autogenerated)
#### Type Of Change
- [ ] Small bug fix (non-breaking change which fixes an issue)
- [ ] New feature (non-breaking change which adds new functionality)
- [ ] Breaking change (fix or feature that would change existing functionality)
#### Checklist:
<!-- Go over all points below, and after creating the PR, tick the checkboxes that apply -->
- [ ] PR should be backported to stable branches
- [ ] Tested changes locally
- [ ] Related to issue #4167
#### Description
WSS connection is not shown in tls.list RPC command like all other TLS connections.
Also tls_kill does not handle WSS connections.
This is related to the issue 4167.
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/4262
-- Commit Summary --
* tls: add WSS to RPC funtions
-- File Changes --
M src/modules/tls/tls_rpc.c (4)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/4262.patchhttps://github.com/kamailio/kamailio/pull/4262.diff
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