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Looking for Documentation/Guidance to Setup Kamailio and RTPProxy to Terminate Encryption for Asterisk
by Joshua Ruehlig 29 Apr '25

29 Apr '25

Hello, I am fairly new to SIP and just recently setup Asterisk for a few purposes. 1) To have my Hikvision doorbell videocall my phone when someone presses it 2) Videocalling between me and my families' phones/tablets 3) Eventually have a home phone setup for POTS calling. I currently have an ObiHai202 that is still working with GoogleVoice, so this not a priority at the moment. I wanted to have encryption for SIP and RTP outside of my network, but wanted this to all be terminated on my OPNsense router. I can install Kamailio and RTPProxy on OPNsense since they are available for FreeBSD, but it looks like Rtpengine is currently not ported. Ideally I wanted most logic to be handled on my Asterisk server, and have Kamailio/RTPProxy do the bare minimum of encrypting/decryption and rewriting headers as needed. ## Diagram of my Targeted Setup ## * Android Phones / Tablets running Linphone || SIPS(TLS) / SRTP || * OPNsense Router ~ WAN_IP(dynamic)=DOMAIN.com / LAN_IP=10.0.1.1 - Kamailio / RTPProxy || SIP(UDP) / RTP || * FreeBSD Application Server ~ LAN_IP=10.0.1.17 / Internal_IP=192.168.10.17 - Asterisk || SIP(UDP) / RTP || * Hikvision Doorbell ~ Internal_IP=192.168.10.100 #### From my reading so far I have a few questions... * To setup Kamailio > Asterisk it looks like I need to setup the dispatcher module; I also I may need to write some routing code. Am I correct, that I will need to use the dispatcher module? Will I need to modify the routing code/logic or is the default usually sufficient? * I read RTPProxy does support SRTP, but I don't see any configuration options related to encryption. I assume this is just enabled in Asterisk/Kamailio? * For Asterisk configuration I believe I just need to define the outbound_proxy. But now I am not sure how I can require SRTP between my Linphone clients and RTPProxy, while the leg between RTPProxy and Asterisk is unencrypted RTP? Thanks for any help that is provided!

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rtpengine module does not detect new nodes in table
by David Cunningham 29 Apr '25

29 Apr '25

Hello, We are using Kamailio 5.8.2 with the rtpengine module on Ubuntu 22.04. The "rtpengine" table is used to define nodes, with a configuration like this: modparam( "rtpengine", "db_url", DB_URL ) modparam( "rtpengine", "table_name", "rtpengine" ) modparam( "rtpengine", "setid_default", 1 ) It works fine if Kamailio is restarted after entries in the "rtpengine" table are inserted or deleted, but new entries in the table aren't detected automatically. Is there a way to configure Kamailio to detect them? Thank you, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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Re: SIP Call Drops After 0.32 Seconds – Asterisk on Kubernetes with Kamailio + RTPengine
by airsay＠duck.com 28 Apr '25

28 Apr '25

I will let the more experienced Kamailio folks comment on the technical/Kamailio/RTP modifications that you need to make. The SIP standard requires an ACK to a 200 OK to be sent within that 32 second window. If an ACK is not received, the session is torn down. You would need to do a packet capture to review the Contact header in the SIP and the c: in the SDP. I'm just getting bedded in with SIP, reading Alan B Johnston's "SIP: Understanding the Session Initiation Protocol". Excellent introduction to understanding SIP. Best regards Sent from my iPhone > On 23 Apr 2025, at 16:35, Fernando Lopes via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello everyone, > I'm running into an issue with SIP calls in my current setup and would really appreciate some help. > > Setup: > I have a machine named sip00 (IP: 192.168.1.75) running Kamailio + RTPengine. > Kamailio is dispatching calls to sip:192.168.1.190:32210;transport=tcp. This IP points to another machine running Asterisk inside a Kubernetes cluster. > RTPengine is configured with an RTP port range of 10000–20000, and my router is set to allow that same range. > > Asterisk Kubernetes Service Configuration: > yaml > Copy > Edit > spec: > ports: > - name: tcp-port > protocol: TCP > port: 5060 > targetPort: 5060 > nodePort: 32210 > - name: udp-port > protocol: UDP > port: 5060 > targetPort: 5060 > nodePort: 32210 > > Problem: > When I initiate a SIP call, the router forwards traffic to Kamailio + RTPengine, which then sends it to the Asterisk server on 192.168.1.190. > Everything seems fine initially, but at exactly 0.32 seconds into the call, Asterisk sends a BYE and no longer responds with 200 OK to the SIP dialog — even though I'm still receiving and sending audio. Then, at around 01:04, I get a 408 Request Timeout. > > Questions: > Do I need to explicitly expose the RTP port range (10000–20000) in the Asterisk Kubernetes service as well? > Why is Asterisk sending a BYE so early if audio is still flowing? > Could it be a signaling timeout or an issue with SIP dialog tracking? > Any help or pointers would be greatly appreciated! > > Thanks in advance! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender!

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Kamailio TLS Listener Not Binding to Port (5061 or 10061)
by steven.muchwe 28 Apr '25

28 Apr '25

Hello Kamailio Community, I’ve been troubleshooting an issue with Kamailio where the TLS listener fails to bind to any specified ports (5061, 10061, etc.), despite valid configurations and certificates. Here’s a summary of my setup and steps taken: 1. *Environment Details*: - Kamailio version: 5.7.4 - OpenSSL version: 3.0.13 - Operating System: Ubuntu (Noble Release) - TLS module (tls.so) is installed and loaded. 2. *Issue Details*: - Configurations validate successfully (config file ok). - OpenSSL works perfectly when testing certificates and keys with s_server, binding to ports (5061 and others). - Kamailio fails to bind TLS listeners (ss -tulnp shows no activity on the specified ports). 3. *Steps Already Taken*: - Simplified TLS configuration (minimal_tls.cfg) with: plaintext listen=tls:10.14.202.39:5061 loadmodule "tls.so" modparam("tls", "certificate", "/home/localtech/vicissl/868a963bc33d5eae.crt") modparam("tls", "private_key", "/home/localtech/vicissl/private.key") - Tested multiple ports (5061, 10061, 15061). - Checked firewall settings (iptables) and confirmed no restrictions. - Rebuilt Kamailio from source and ensured TLS modules are linked to OpenSSL. - Ran Kamailio with maximum debugging (-ddd) to examine logs—no binding-related errors appeared. 4. *Log Excerpts*: (Attach relevant logs showing TLS initialization or lack of binding activity.) 5. *Question*: What additional steps or configurations should I explore to resolve this issue? Could this be a compatibility problem between Kamailio 5.7.4 and OpenSSL 3.0.13? Any guidance or insights would be greatly appreciated! Best regards, Steven Muchwe Njoroge

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OT: Client DNS resolver crashing on NAPTR lookup
by Benoit Panizzon 28 Apr '25

28 Apr '25

Hi List I hope it is OK to post a slightly off topic question. I have not found any similar reports via google. During testing of our development platform, I found two independent clients whose resolver process crashes randomly, when performing NAPTR lookups. Mitel/Aastra 68XXi Phones with Firmware >= 6.4 up to most recent. Inalp/Patton SNXX (Trinity OS) SIP Gateway with Firmware >= 3.14.X up to most recent. Both vendors reply they have no prior reports of this issue. The hostname used as registrar / proxy points to a NAPTR entry, which in turn points to 3 SRV entries für SIP+D2T, SIP+D2U, SIPS+D2T Each SRV entry then points to 4 IP Addresses (2x ipv4 and 2x ipv6) So quite a complex lookup and quite large replies. Anyone else observed this issue? When using a hostname that merely points to the A and AAAA records, no crashes. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Kamailio Redis integration
by Timothy Delin 28 Apr '25

28 Apr '25

Hey team, I am facing an issue where my Kamailio and Redis are not integrated. I am using a VM with RHEL 9 OS, Kamailio in 5.4.7 version, and Redis DB in 7.2.7 version. Both my Kamailio and Redia are using Docker. We have installed the libhiredis and the ndb_redis in Kamailio When we tried to do a simple print "Hello World" from the Redis in Kamailio, we are receiving this error "ERROR: could not open module, missing loadmodule" even though we have confirmed in kamcmd Best regards, Timothy

3 2

Authentication qop auth-int
by 13715209697＠139.com 28 Apr '25

28 Apr '25

Hello! Kamailio 5.7.6 Debian 10.12 Here is my authentication route: ``` route[AUTH] { if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "2"); # flag not 0 exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } ... } ``` start eyebeam Version 1.5.14.4 ``` eyebeam --->REGISTER ----> kamailio <--401 (WWW-Authenticate: Digest realm="192.168.3.10", nonce=..., qop="auth-int") --->REGISTER (Authorization: Digest username="08001",realm="192.168.3.10",nonce=...,uri=...,response=...,cnonce=...,nc=00000001,qop=auth-int,algorithm=MD5) <--401 (auth_challenge() return -2 ) ``` OpenSIPS 3.4.12 register to kamailio, got the same result as eyebeam Any suggestions? Thanks!

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Kamailio v6.0.1 and SUBSCRIBE processing
by Blair, Steve 26 Apr '25

26 Apr '25

I’m coming back to Kamailio after about a 10 year absence. I have v6.0.1 running handling registrations (unique devices) and inbound/outbound PSTN calling correctly. What I need is to process subscribe requests. I’m looking for direction on the correct configuration to 1. Authenticate subscribe messages. This appears to be working with following snippet. #eDevice: Need to authenticate subscribes before proceeding xlog("L_INFO", "eDevice: Start initial SUBSCRibe check. \n"); if (is_method("SUBSCRIBE")) { record_route(); if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); } # eDevice SUB Change consume_credentials(); route(PRESENCE); } 1. After authenticating forward any subscribe to an external presence server 2. Relay any response from external presence server to client device. Only the above authentication is working so direction on proper processing of subscribes when Kamailio is not the presence server will be appreciated. Thanks

1 0

Re: RTPengine media starts for intiating caller only
by airsay＠duck.com 23 Apr '25

23 Apr '25

Again, not super skilled yet with Kamailio/RTPEngine, but my first instinct is: "why you are advertising myproxyname.domain.net which is possibly a public IP". Your entire setup is in your LAN so you don't need to advertise any public IP. Look forward to others contributing and clarifying that Sent from my iPhone > On 23 Apr 2025, at 20:42, smh2017--- via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello, > > I have a problem with my basic Kamailio setup using RTPengine. I'm testing initially with two client UAs in the same LAN as the SIP server behind the firewall. The initiating UA starts the media proxy as soon as the receiving UA accepts the call, indicated by the call timer starting. But after accepting the call the receiving UA's timer doesn't start and after 10-15 secs hangs up with a NO ACK error message. > > I am using the kamailio default config file that comes with the Debian package and have made minimal changes recommended by various How-Tos for using Kamailio behind a firewall. Because the clients can successfully register over TLS, I'm assuming the problem is in the rtpengine configuration. The two rtp listen parameters I've tried are: > > interface = internal/192.168.1.100;external/192.168.1.100!myproxyname.domain.net > --or-- > interface = internal/192.168.1.100;external/myproxyname.domain.net > listen-ng = localhost:2223 > > I'm not sure why some example rtpengine configs list both the private and public addresses as external. > > Any suggestions where to start troubleshooting? > > Thanks, > > Steve > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender!

2 1

RTPengine media starts for intiating caller only
by smh2017＠zoho.com 23 Apr '25

23 Apr '25

Hello, I have a problem with my basic Kamailio setup using RTPengine. I'm testing initially with two client UAs in the same LAN as the SIP server behind the firewall. The initiating UA starts the media proxy as soon as the receiving UA accepts the call, indicated by the call timer starting. But after accepting the call the receiving UA's timer doesn't start and after 10-15 secs hangs up with a NO ACK error message. I am using the kamailio default config file that comes with the Debian package and have made minimal changes recommended by various How-Tos for using Kamailio behind a firewall. Because the clients can successfully register over TLS, I'm assuming the problem is in the rtpengine configuration. The two rtp listen parameters I've tried are: interface = internal/192.168.1.100;external/192.168.1.100!myproxyname.domain.net --or-- interface = internal/192.168.1.100;external/myproxyname.domain.net listen-ng = localhost:2223 I'm not sure why some example rtpengine configs list both the private and public addresses as external. Any suggestions where to start troubleshooting? Thanks, Steve

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