#### Pre-Submission Checklist
- [x] Commit message has the format required by CONTRIBUTING guide
- [x] Commits are split per component (core, individual modules, libs, utils, ...)
- [x] Each component has a single commit (if not, squash them into one commit)
- [x] 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
- [x] 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:
- [x] PR should be backported to stable branches
- [x] Tested changes locally
- [ ] Related to issue #XXXX (replace XXXX with an open issue number)
#### Description
As a local IP address for TCP sending operation the Kamailio service is taking the same network_interface/IP_address, which is used by the service for TCP listening.
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/3925
-- Commit Summary --
* core: local TCP socket is bound on listening address
-- File Changes --
M src/core/tcp_main.c (24)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/3925.patchhttps://github.com/kamailio/kamailio/pull/3925.diff
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Message ID: <kamailio/kamailio/pull/3925(a)github.com>
Hello
The timer_enable() function does not exist in KEMI.
Can you implement the equivalent of this function in KEMI?
BR
Juan PASCUAL
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https://github.com/kamailio/kamailio/issues/4008
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Message ID: <kamailio/kamailio/issues/4008(a)github.com>
Hi,
I am thinking of this idea to add shm storage for topos module. Main
motivation for this is that if db backend is not available, there will be
no store/load happening.
Did anyone else thought of this or is doing something similar already?
I see 2 milestones here:
1. Add code in topos to keep a shm hash table with all the information
needed, guarded via locks. The api functions should be very similar to what
tps_storage.c has, just it will do ops directly in memory not on db.
2. Find a way to synchronize this topos hash table among multiple
kamailios. This should be similar to how htable module syncronize via DMQ.
Sync may happen either for each new cell, for batches or for entire topos
hash table.
What do you think of this? Any opinions, comments, appreciated.
Thank you,
Stefan Mititelu
### Description
I used sip_p_charging_vector() function of siputils with flag f (remove + generate).
This function works fine if call is received without a P-Charging-Vector and only generate needs to be done.
But if the INVITE already contains a P-Charging-Vector and this vector needs to be removed and replaced by a new generated P-Charging-Vector the output is messed up.
The sip_p_charging_vector() function with flag r (remove) works fine too. P-Charging-Vector is removed if INVITE contains one. So for me this looks like an issue with the combination of remove + generate.
See SIP Traffic attached. The "old" P-Charging-Vector is not deleted and additionaly instead of a new P-Charging-Vector the "old" P-Charging-Vector is added again above SDP. Also some part of the next line "Session-Exp" is added and "TH: dih".
-----
TH: dih
icid-value=0d702d06-5700-103e-00-00-00-10-6b-03-dd-04;icid-generated-at=213.xxx.xxx.xxx
Session-Exp
-----
### Troubleshooting
#### Reproduction
It looks like the issue is triggered always a p-charging-vector is received by kamailio and sip_p_charging_vector() with flag f is called.
#### 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
```
**This message is received by kamailio (Incoming):**
INVITE sip:+49xxxxxxxx;npdi;rn=+49xxxxxxxx@bob.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 213.xxx.xxx.xxx:5060;branch=z9hG4bK08Bf3722bac16f3e05b
From: <sip:+43xxxxxxxx@alice.com;user=phone>;tag=gK085abca7
To: <sip:+49xxxxxxxx0@92.xxx.xxx.xxx:5060;user=phone>
Call-ID: 168341746_132086108(a)213.xxx.xxx.xxx
CSeq: 373279 INVITE
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+43xxxxxxxx@213.xxx.xxx.xxx:5060>
P-Asserted-Identity: <sip:+43xxxxxxxx@alice.com:5060;user=phone>
Supported: timer,100rel,histinfo
P-Charging-Vector: icid-value=0d702d06-5700-103e-00-00-00-10-6b-03-dd-04;icid-generated-at=213.xxx.xxx.xxx
Session-Expires: 1800
Min-SE: 90
Content-Length: 236
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 278609 22801 IN IP4 213.xxx.xxx.xxx
s=SIP Media Capabilities
c=IN IP4 213.xxx.xxx.xxx
t=0 0
m=audio 29020 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
**This message is send out by kamailio (outgoing):**
INVITE sip:+49xxxxxxxx;npdi;rn=+49xxxxxxxx@bob.com:5060;user=phone SIP/2.0
Record-Route: <sip:172.xxx.xxx.xxx;r2=on;lr=on;ftag=gK085abca7>
Record-Route: <sip:92.xxx.xxx.xxx9;r2=on;lr=on;ftag=gK085abca7>
Via: SIP/2.0/UDP 172.xxx.xxx.xxx;branch=z9hG4bKc451.940fa2c220f96c1ef968de3c156db262.0
Via: SIP/2.0/UDP 213.xxx.xxx.xxx:5060;TH=div;branch=z9hG4bK08Bf3722bac16f3e05b
From: <sip:+43xxxxxxxx@alice.com;user=phone>;tag=gK085abca7
To: <sip:+49xxxxxxxx@92.xxx.xxx.xxx:5060;user=phone>
Call-ID: 168341746_132086108(a)213.xxx.xxx.xxx
CSeq: 373279 INVITE
Max-Forwards: 66
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+43xxxxxxxx@213.xxx.xxx.xxx:5060>
P-Asserted-Identity: <sip:+43xxxxxxxx@alice.com:5060;user=phone>
Supported: timer,100rel,histinfo
P-Charging-Vector: icid-value=0d702d06-5700-103e-00-00-00-10-6b-03-dd-04;icid-generated-at=213.xxx.xxx.xxx
Session-Expires: 1800
Min-SE: 90
Content-Length: 252
Content-Disposition: session; handling=required
Content-Type: application/sdp
**_TH: dih
icid-value=0d702d06-5700-103e-00-00-00-10-6b-03-dd-04;icid-generated-at=213.xxx.xxx.xxx
Session-Exp_**
v=0
o=Sonus_UAC 278609 22801 IN IP4 92.xxx.xxx.xxx
s=SIP Media Capabilities
c=IN IP4 92.xxx.xxx.xxx
t=0 0
m=audio 34622 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:34623
a=ptime:20
```
### Additional Information
```
version: kamailio 5.8.2 (x86_64/linux) 3fa5f4
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: 3fa5f4
compiled on 15:54:29 Jul 22 2024 with gcc 10.2.1
```
* **Operating System**:
```
Debian GNU/Linux 11
5.10.0-26-amd64 #1 SMP Debian 5.10.197-1 (2023-09-29) x86_64 GNU/Linux
```
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Hi,
we had strange behavior with kamailio when some rtpengines were
unreachable, so we looked into the the source code and found this in
select_rtpp_node_new():
3323 /* Try to enable if it's time to try. */
3324 if(node->rn_disabled && node->rn_recheck_ticks <= get_ticks()) {
3325 node->rn_disabled = rtpp_test(node, 1, 0);
3326 }
Does this mean that a ping will be done during rtpengine manage?
Is this wise? I mean there are dedicated pings in the background. Doing
this in rtpengine manage will only periodically delay the processing of
a SIP message. We fear that in high traffic situations this could occupy
all threads. Is this a bug?
As a workaround we think setting rtpengine_tout_ms to a lower value
might help as it shortens the amount of time the module delays the
processing of the SIP message.
What do you think?
Best wishes
Christian Berger
--
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Telefon: +49 (0)211-63 55 55-0
Telefax: +49 (0)211-63 55 55-22
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