March 2026 - sr-users - kamailio.org

Postgres connections optimization #4342
by Stephane Savonitto 17 Jun '26

17 Jun '26

Hello, On P-CSCF when running we have 200+ permanent connection to postgres db, even if there is no traffic. Is there a way to manage connections dynamically using a pool like using PgBouncer ? Regards, Stéphane

6 10

6.0.2 to 6.1.1 Migration issue: malformed From header no longer reachable in routing script
by m.cazzola＠zaleos.net 19 May '26

19 May '26

Hello, I am writing regarding a behaviour change between Kamailio 6.0.2 and 6.1.1 that is causing one of our system tests to fail. The test sends intentionally malformed SIP INVITE requests and expects a 400 response. The failing cases involve malformed From headers (e.g. an empty From). After reviewing the changelog and tracing the code, we believe the change was introduced in commit cf1c796 (cherry-picked into 6.0.4), which reworks From header parsing in get_hdr_field() to use the shared parse_to_body() function. As a result, a malformed From now causes parse_headers() to return an error earlier in receive.c, before the message reaches the routing script. Previously, the From header parsing failure was caught deep enough in execution for our sanity_check() call in the request route to detect and handle it — sending back a 400 as RFC 3261 implies. Now, the failure happens before the routing script is reached. We have attempted to intercept it in event_route[core:pre-routing] and event_route[core:receive-parse-error], but to no avail. All of our other malformed type messages (e.g. a bad Expires value or invalid RURI) work as intended. Our questions are: Is there any hook, event, or flag we could use to detect and respond to this condition before the message is dropped? If no config-level workaround exists, are there any other alternatives? Thank you in advance for your time. Best regards,  Matteo Cazzola

3 6

Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
by Ben Fitzgerald 17 Apr '26

17 Apr '26

Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: <sip:40081@kamailio:5060> Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER ========================= sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

4 8

problems using kamailio
by Peter Walber 17 Apr '26

17 Apr '26

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". 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:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:46:57.391516534Z 9(52) INFO: <script>: Current Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;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:**73@192.168.2.209:5060> 2025-08-07T07:47:00.195739350Z 9(52) INFO: <script>: START: INVITE from sip:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:47:00.195800424Z 9(52) INFO: <script>: Current Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;transport=ws;ob> 2025-08-07T07:47:00.195811886Z 9(52) INFO: <script>: Setting new Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;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:**72@speedport.ip 2025-08-07T07:47:00.196181228Z 9(52) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:**72@speedport.ip, 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:**73@speedport.ip (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:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:27.951663073Z 9(52) INFO: <script>: Current Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;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:**73@192.168.2.209:5060> 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:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:30.265125134Z 9(52) INFO: <script>: Current Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;transport=ws;ob> 2025-08-07T07:47:30.265135557Z 9(52) INFO: <script>: Setting new Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;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:**72@speedport.ip 2025-08-07T07:47:30.265664116Z 9(52) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:**72@speedport.ip, 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:**73@speedport.ip (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* *

7 15

Kamailio db_redis & usrloc: Cannot Retrieve Location via Username from redis-cli
by xf han 01 Apr '26

01 Apr '26

Dear Kamailio Community, I encounter an issue when using db_redis with the usrloc module. Below are my relevant Kamailio configurations: # kamailio 6.1.1 modparam("db_redis", "schema_path", "/usr/share/kamailio/db_redis") modparam("db_redis", "keys", "location=entry:ruid&usrdom:username,domain&timer:expires") modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", 0) modparam("usrloc", "db_url", "redis://localhost:6379/0") modparam("usrloc", "db_mode", 3) After user 1001 successfully registers, I execute the KEYS * command in redis-cli, and the output is as follows: redis-cli KEYS * 1) "location::index::timer" 2) "location:entry::uloc-69cb7f3b-499e-1" 3) "location:timer::2026-03-31 16:01:59" My question: I cannot find the corresponding user location information via the username in redis-cli. Could anyone help me solve this problem? Thank you very much! Best regards,

1 2

Could not relay BYE to wss client
by igor.goncharovsky＠gmail.com 27 Mar '26

27 Mar '26

Hello, While developing config for WebRTC proxy. Issue I have - inability to forward BYE from UDP side to WebRTC. We have updated to 5.8.8 with no luck.Issue we have shown in log bellow: we have an error message that tells Via could not be built. Mar 20 18:42:17 pbxKamaTestSBC /usr/local/kamailio/sbin/kamailio[1390086]: DEBUG: call_1774014127984617813|BYE|<null><core> [core/msg_translator.c:1859]: check_boundaries(): no multi-part body Mar 20 18:42:17 pbxKamaTestSBC /usr/local/kamailio/sbin/kamailio[1390086]: WARNING: call_1774014127984617813|BYE|<null><core> [core/msg_translator.c:3036]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Mar 20 18:42:17 pbxKamaTestSBC /usr/local/kamailio/sbin/kamailio[1390086]: ERROR: call_1774014127984617813|BYE|<null><core> [core/msg_translator.c:2100]: build_req_buf_from_sip_req(): could not create Via header Mar 20 18:42:17 pbxKamaTestSBC /usr/local/kamailio/sbin/kamailio[1390086]: ERROR: call_1774014127984617813|BYE|<null>tm [t_fwd.c:484]: prepare_new_uac(): could not build request Mar 20 18:42:17 pbxKamaTestSBC /usr/local/kamailio/sbin/kamailio[1390086]: ERROR: call_1774014127984617813|BYE|<null>tm [t_fwd.c:1764]: t_forward_nonack(): failure to add branchest I have tried to display tcp connection id from within the branch, but with no luck. I have printed $nh(u) and $nh(P) before t_relay() and it shows the correct URI and transport to be used. TCP connection id also shows correct and connection alive for the moment of BYE sent. Is there any known issue in that part? What tools could be additionally used to debug it? $nh(u) and $nh(P): sip:10.15.151.59:43786;transport=ws ws connection id to 10.15.151.59:43786 is: 2 # kamcmd ws.dump used_asc { connections: { 2: wss:10.15.151.59:43786 -> wss:10.1.97.53:5443 (state: OPEN, last used 2s ago, sub-protocol: sip) 1: wss:10.15.151.59:43798 -> wss:10.1.97.53:5443 (state: OPEN, last used 2s ago, sub-protocol: sip) } info: { wscounter: 2 truncated: no } }

2 1

about sip message
by zhaobin 27 Mar '26

27 Mar '26

I try send sip message from freeswitch to webrtc sip client,then kamailio send back 500 error. I have some question about this: 1:image show kamailio sl module send this error.Why sip message is trans by sl module,not tm module? 2:about error show (2/SL),i know SL is sl module,and 2 is what? 3:so i must get debug log to get more infomations about this error?

4 7

doubt using rtpengine_delete
by descartin＠bts.io 26 Mar '26

26 Mar '26

Hi, We are observing an unexpected behavior when using Kamailio together with rtpengine and would like to understand if this is expected or if there is a way to control it. Environment: Kamailio version: 6.0.4 rtpengine version: 12.5.1.48 (mr12.5.1 branch) Transport: UDP NG control socket Observation: From the rtpengine logs, we consistently see that every delete command is preceded by a query command for the same call, coming from the same Kamailio instance and even the same source port. Mar 19 08:51:05 mia-rtpengine-6 rtpengine[109401]: INFO: [SD7kus901-8c2067aa09bf3bfafd4b18ca3df0eeb9-v300g00010]: [control] Received command 'query' from 192.168.9.56:49983 Mar 19 08:51:05 mia-rtpengine-6 rtpengine[109401]: INFO: [SD7kus901-8c2067aa09bf3bfafd4b18ca3df0eeb9-v300g00010]: [control] Replying to 'query' from 192.168.9.56:49983 (elapsed time 0.000043 sec) Mar 19 08:51:05 mia-rtpengine-6 rtpengine[109401]: INFO: [SD7kus901-8c2067aa09bf3bfafd4b18ca3df0eeb9-v300g00010]: [control] Received command 'delete' from 192.168.9.56:49983 Mar 19 08:51:05 mia-rtpengine-6 rtpengine[109401]: INFO: [SD7kus901-8c2067aa09bf3bfafd4b18ca3df0eeb9-v300g00010]: [control] Replying to 'delete' from 192.168.9.56:49983 (elapsed time 0.000036 sec) We are not explicitly calling rtpengine_query() anywhere in our Kamailio configuration. Our usage is limited to rtpengine_offer(), rtpengine_answer(), and rtpengine_delete(). Is this query before delete expected behavior (e.g. implicitly triggered by rtpengine_delete() to retrieve statistics)? If so, is there any way to disable this behavior so that rtpengine_delete() does not trigger a query? related issue ?¿ Seems under high load conditions, we have observed that the NG control socket (UDP) occasionally becomes congested. For example, we have seen cases where the rtpengine server receive queue (Recv-Q) for the control socket grows significantly (hundreds of thousands of bytes), indicating that rtpengine is not draining the socket fast enough. ss -lunp | grep 7772 UNCONN 214272 0 192.168.9.129:7772 0.0.0.0:* users:(("rtpengine",pid=3320,fd=9)) this issue could have some relation with an overload of control commands, in some particular scenarios if we have these parameters set for rtpengine instances selection? modparam("rtpengine", "queried_nodes_limit", 2) modparam("rtpengine", "rtpengine_retr", 2) modparam("rtpengine", "rtpengine_tout_ms", 350) Thanks in advance for your help. David Escartín

4 3

Kamailio with 2 daisy-chained rtpengine
by Lyubomir Yatanski 24 Mar '26

24 Mar '26

Hi, I have an IMS setup with Kamailio, where the attached phone is in private network. 2 rtpengines are required to bridge between networks: 1. with 2 network interfaces - internal and external 2. with single interface in the external network (required as this interface is known to a firewall, not under my control) I'm trying to daisy-chain the 2 rtpengine instances so media path could be established, by following the documentation: modparam("rtpengine", "rtpengine_sock", "1 == udp:X.X.X.X:9910") modparam("rtpengine", "rtpengine_sock", "2 == udp:Y.Y.Y.Y:9910") after that in the script on INVITE: set_rtpengine_set("1", "2"); rtpengine_manage("internal, external"); and on the answer: set_rtpengine_set("1", "2"); rtpengine_manage("external, internal"); The behaviour I was expecting was in the offer to see only the IP of rtpengine of setid 2 and on the answer to see the internal address of the rtpengine of setid 1. What I'm actually observing is both addresses of the 2 rtpengines get appended on both directions and this seems to confuse the phone as no RTP is flowing. I'll appreciate any help, suggestions or guidance. Regards, Lyubomir Yatanski

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possible performance regression in Kamailio 6.1 release related to TCP/TLS
by Henning Westerholt 24 Mar '26

24 Mar '26

Hello, we have noticed a possible performance regression related to TCP/TLS in the Kamailio 6.1.x release when installed from packages or from source with cmake, and in Kamailio 6.0 releases only when installed from source with cmake. There was a change introduced by mistake that disabled certain polling functionality related to the internal TCP stack. It can be observed at startup, or by executing "kamailio -I". See below for further information. This was fixed today in git master, 6.1 and 6.0 branches. Commit ids are: 54572ce96, f1a333b37b, b906c409. I believe that it should be only noticeable if you are having a high number of connected devices and/or a high connection rate. The mentioned fix will be included in the next minor releases of 6.1 and 5.8., as usual. Best regards, Henning Westerholt 1. Verification in log files * Affected: kamailio: INFO: <core> [core/tcp_main.c:5264]: init_tcp(): using poll as the io watch method (auto detected) * Unaffected: kamailio: INFO: <core> [core/tcp_main.c:5264]: init_tcp(): using epoll_lt as the io watch method (auto detected) 1. Verification in Kamailio binary: * Affected: $ kamailio -I | grep "poll methods" TCP poll methods: poll * Unaffected: $ kamailio -I | grep "poll methods" TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select -- Kamailio & VoIP services - https://gilawa.com<https://gilawa.com/>

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