September 2025 - sr-users

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

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

topoh breaks SIPS with PJSIP if PJMEDIA_SRTP_MANDATORY is set
by Mathias Schneuwly 01 Dec '25

01 Dec '25

Hi I have a question related to topoh. We are running Kamailio in version 6.0.2 with topoh enabled. The PJSIP based softphone is unable to establish a call with SIPS and SRTP if we use "PJMEDIA_SRTP_MANDATORY" in PJSIP. PJSIP complains about "Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session". The other side is sending "Contact: <sips:620150@10.165.44.117:59686;transport=tls>", but topoh modifies the contact header to "Contact: <sip:127.0.0.8;line=sr-avdFaQDvhUxGrqtxhqxThq-ZnUy6nUzGrQDZmqa1rUf6a0JTaLt4aoyV2EGQ>" without sips scheme and without transport=tls. PJSIP then refuses to establish the call and sends immediately a BYE. If topoh is disabled, the call is successfully established. Is it somehow possible to tell topoh to keep the scheme or transport=tls if the other side provided it? Thanks for your help Best regards Mathias

3 3

Kamailio Developers Meeting, Dec 8-9, 2025, in Dusseldorf
by Daniel-Constantin Mierla 24 Nov '25

24 Nov '25

Hello, Kamailio SIP Server project is organizing another meeting of its developers and community members during December 8-9, 2025 (Mon-Tue), hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and contributors in order to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join, developer or user interested in helping the project. Please note we have a limited capacity of seats in the meeting room, the main policy for accepting participants being first come first server. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

RTPEngine installation on RHEL 8.x or Rocky Linux 8.x
by Chandramouli P 21 Nov '25

21 Nov '25

Hello, I tried a lot to install RTPEngine on RHEL 8.x and as well as in Rocky Linux 8.x also using different sources. But, I am facing a lot of issues like libraries issues, dependencies issues, conflict issues etc. Anyone successfully installed RTPEngine on RHEL 8.x or Rocky Linux 8.x (Not on CentOS)? If so, could you please share the documentation step-by-step or web URL? Thanks in advance. Best regards, Chandramouli.

5 18

dynamic selection of authentication algorithm
by Juha Heinanen 28 Oct '25

28 Oct '25

How about this new function that sets the algorithm? -- Juha iff --git a/src/modules/auth/auth_mod.c b/src/modules/auth/auth_mod.c index 534ad9e20f..4e3a584d22 100644 --- a/src/modules/auth/auth_mod.c +++ b/src/modules/auth/auth_mod.c @@ -74,6 +74,10 @@ int w_consume_credentials(struct sip_msg *msg, char *s1, char *s2); * Check for credentials with given realm */ int w_has_credentials(struct sip_msg *msg, char *s1, char *s2); +/* + * Set authentication algorithm + */ +int w_auth_algorithm(struct sip_msg *msg, char *alg, char* s2); static int pv_proxy_authenticate( struct sip_msg *msg, char *realm, char *passwd, char *flags); @@ -170,6 +174,8 @@ static cmd_export_t cmds[] = { REQUEST_ROUTE}, {"pv_auth_check", (cmd_function)w_pv_auth_check, 4, fixup_pv_auth_check, 0, REQUEST_ROUTE}, + {"auth_algorithm", w_auth_algorithm, 1, fixup_spve_null, 0, + REQUEST_ROUTE}, {"bind_auth_s", (cmd_function)bind_auth_s, 0, 0, 0}, {0, 0, 0, 0, 0, 0} @@ -475,6 +481,35 @@ int w_has_credentials(sip_msg_t *msg, char *realm, char *s2) return -1; } return ki_has_credentials(msg, &srealm); + +} +/** + * + */ +int w_auth_algorithm(sip_msg_t *msg, char* alg, char* s2) +{ + if (fixup_get_svalue(msg, (gparam_t*)alg, &auth_algorithm) < 0) { + LM_ERR("failed to get algorithm value\n"); + return -1; + } + + if (strcmp(auth_algorithm.s, "MD5") == 0) { + hash_hex_len = HASHHEXLEN; + calc_HA1 = calc_HA1_md5; + calc_response = calc_response_md5; + } + else if (strcmp(auth_algorithm.s, "SHA-256") == 0) { + hash_hex_len = HASHHEXLEN_SHA256; + calc_HA1 = calc_HA1_sha256; + calc_response = calc_response_sha256; + } + else { + LM_ERR("Invalid algorithm provided." + " Possible values are \"\", \"MD5\" or \"SHA-256\"\n"); + return -1; + } + + return 1; } #ifdef USE_NC

5 13

SIP Trunk registration In Kamailio
by tanmoykarmakar10＠gmail.com 14 Oct '25

14 Oct '25

I want to register SIP Trunk directly in Kamailio. How can I register the SIP trunk in Kamailio. Please find the below sample SIP trunk details. Network IP 10.2.2.2 Subnet Mask 255.255.255.252 Customer IP 10.3.3.3 Gateway IP 10.1.1.1 SBCIP 10.4.4.4 Username 12345678 Password 1234 Please help me with configuration to register SIP trunk in kamailio.

4 4

using dlg_refer
by Robert Boardman 09 Oct '25

09 Oct '25

Hi All, I'm trying to use dlg_refer to separate to legs of a call and send them to another device, I think I have setup the system correctly, everything seems to look ok but the call to dlg_refer doesn't do anything , it doesnt even show in the journal that it is failing, just nothing happens Here is the code: #### REFER caller leg if ($dlg_var(caller_contact) != $null) { xlog("L_INFO", "Sending dlg_refer to caller: $var(conf_uri)\n"); #dlg_refer("caller", "$var(conf_uri)"); xlog("L_INFO", ">>> Attempting dlg_refer to caller: $dlg_var(caller_contact)\n"); if (!dlg_refer("caller", "$var(conf_uri)")) { xlog("L_ERR", "dlg_refer to caller FAILED\n"); } else { xlog("L_INFO", "dlg_refer to caller triggered\n"); } } The dialog state is 4 before the refer is initiated and can be seen from the kamcmd dlg.list. The caller end point allows REFER I've ran out of ideas as to where to look, any pointers would be a fantastic help I'm using Kamailio V5.8.6 Thanks Rob Robert Boardman Solution Architect Tel: 03330147908 Email: Robert.Boardman(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at The Scalpel, 18th Floor, 52 Lime Street, London EC3M 7AF and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

5 5

Kamailio SBC using RTPproxy
by Ibe Van de Veire 01 Oct '25

01 Oct '25

Dear Kamailio community, I am currently developing a VoIP system using 2 instances of Kamailio. One as SBC and one as SIP server. The SBC is configured to work in bridge mode (one NIC on the public side, one NIC on the private side). The SIP server works as expected. But the SBC server seems to have some difficulties to route the RTP traffic. I am using RTPproxy to handle NAT traversals. Currently, I handled the following situations: * Internal to internal call * External (both NAT and CG-NAT) to internal call * Internal to external (both NAT and CG-NAT) call The situation that does not work is the following: * External (both NAT or CG-NAT) to External (both NAT or CG-NAT) The SBC seems to always translate the IN IP4 field from external to internal (as desired when calling from external to internal), but this seems not to be desired when calling from external to external. This way, both external devices try to send the RTP data to the internal (inaccessible) address of the SBC. How can I configure the SBC to stop translating the IN IP4 field when both devices are external? I have the following configuration that talks to the RTPproxy service: ========================== if(nat_uac_test("8")) { if($Ri == "MY_PUBLIC_IP") { rtpproxy_manage("coiew"); } else { rtpproxy_manage("faei"); } } else { rtpproxy_manage("cor"); } ========================== Also, is this the desired way to handle NAT traversal with RTPproxy? Kind regards Ibe Van de Veire System Engineer -- Disclaimer -- Vlaamse Radio- en Televisieomroeporganisatie Auguste Reyerslaan 52 1043 Brussel nv van publiek recht BTW BE 0244.142.664 RPR Brussel VRT Gebruikersvoorwaarden <http://www.vrt.be/gebruiksvoorwaarden>

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