sr-users July 2025

sr-users@lists.kamailio.org

56 participants
51 discussions

rtpengine error: no rtpp_set_list->rset_first
by disa_369＠msn.com 24 Jul '25

24 Jul '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from IPPhone ----- Avaya PBX ---sip----> Kamailio -----sip----> Video System-IVR Now Im trying setup rtpengine for proxy media . Kamailio has some errors by starting service. ERROR: rtpengine [rtpengine.c:3131]: select_rtpp_set(): no rtpp_set_list->rset_first Jun 06 12:07:00 zaks-sbc01-op /usr/sbin/kamailio[635658]: NOTICE: rtpengine [rtpengine.c:1815]: mod_init(): Default rtpp set 0 NOT found Missed out on something in config?

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not enough memory for safe operation
by David Cunningham 24 Jul '25

24 Jul '25

Hello, Would anyone have insight into the following errors? We saw several thousand of them appear in a short period of time: Jul 24 10:41:36 vpbx11 /sbin/kamailio[54805]: ERROR: tls [tls_server.c:192]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: shm=67537152 threshold1=76546048 Jul 24 10:41:36 vpbx11 /sbin/kamailio[54805]: ERROR: tls [tls_server.c:192]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: shm=67536032 threshold1=76546048 However there should have been plenty of shared memory available. These were take around 30 minutes later: # kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 407525", "shmem:free_size = 4117928416", "shmem:max_used_size = 4256807952", "shmem:real_used_size = 177038880", "shmem:total_size = 4294967296", "shmem:used_size = 111803416" ], "id": 56272 } # kamcmd pkg.stats pid 54805 { entry: 121 pid: 54805 rank: -1 used: 1249880 free: 6475408 real_used: 1913200 total_size: 8388608 total_frags: 1185 } # ps -ef | grep kama | wc -l 148 Thanks in advance for any tips or insight! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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Basic Configuration
by smh2017＠zoho.com 23 Jul '25

23 Jul '25

I am trying to set up a very basic SIP server. I can only establish external-to-internal calls using the following parameters with the default configuration files: RTP Engine Config File interface=server_local_ip!public_domain_name Kamailio Config File alias="public_domain_name" listen=server_local_ip:5061 adverise public_domain_name:5061 I cannot establish internal-to-external calls nor external-to-external calls with the above configuration settings. In these failed calls, the answering external callee UA sends 200-OK, but the caller UA does not ACK. In looking at the sip trace messages, no alias for the public ip address is appended to the Contact header of the external UA's 200-OK SIP message. So perhaps the caller doesn't know where to send the ACK. The network setup has the local UAs and the Kamailio server behind a port-forwarded NAT routher. External (internet) UAs are also behind NAT routers. Is there a change or addition to the above config snippets that would solve the problem, or do I need to add some instruction to the routing section? Finally, does anyone sell or otherwise offer basic Kamailio/RTPEngine configuration files that work more or less "out-of-the box" for the described setup? Thanks, Steve Hamilton

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Configure TLS 1.3 cipher suites
by caique.cornelio＠mitel.com 22 Jul '25

22 Jul '25

Is the removal of a specific TLS 1.3 cipher supported as today? I am working on making the ciphers accepted configurable and i know it is possible with TLS <1.2, wondering if it is the same case with 1.3.

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Best Book Recommendation for Learning SIP/VoIP
by Fernando Lopes 22 Jul '25

22 Jul '25

Hi everyone, I'm not new but also not a professional to the world of SIP and VoIP but feeling lost, and I'm looking to get a solid foundational understanding. I was wondering if anyone could recommend a good book that explains SIP in a clear and practical way? Whether it's a textbook, a practical guide, or even an eBook, I’d really appreciate your recommendations! Thanks in advance!

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Setting Via IP inside script logic
by Nick Jones 22 Jul '25

22 Jul '25

G'day all, I'm working on a scenario with a Kamailio box with a private IP, with a public IP 1:1 NATed to it (but the VM does not see the public IP on the NICs). When forwarding requests to some hosts I want to set the Via address to be the public IP, but when forwarding requests to other hosts I want to leave the Via address to the internal IP address. If I set the `Advertise` parameter in the bind config, this sets the Via to the advertised IP, but I'm seeing that address used even when communicating with hosts on the private IP. Of course if the IP was on the VM itself I could use $fs or force_send_socket, but that only works if I've got the public IP bound, which I can't do. Is there a simple way to set / override what IP gets baked into the Via header? Cheers, Nick

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Kamailio as a Stateless Smart SIP Proxy
by Antonio 19 Jul '25

19 Jul '25

Hi everyone, Do we have a ready-to-use Kamailio role or configuration for: *“Kamailio as a Stateless Smart SIP Proxy”*, Specifically one that uses *Contact Header Encoding*? That is — a proxy where Kamailio does not store registration state, but encodes the client’s IP and port into the Contact header during REGISTER , and later decodes it from the INVITE to forward the call back to the client. Thanks in advance! -- *Antony* satskiy.a(a)gmail.com

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TLS certificates kamailio & Asterisk
by Fernando Lopes 18 Jul '25

18 Jul '25

Hello everyone, I have a quick question. I'm working on a local project where Kamailio acts as a proxy and Asterisk handles the media. Both are running in separate Docker containers. For tls communication between the two, should I generate a separate TLS certificate for each container, or create only one certificate and use it for both? Thanks in advance!

2 1

cannot clone xavi from non existing parsed_pv_headers
by Mathias Schneuwly 18 Jul '25

18 Jul '25

Hi I'm getting the error "Jul 15 15:15:54 ttel /usr/sbin/kamailio[812316]: ERROR: {1 382 BYE 860005579-5060-39(a)BA.A.B.BCJ} pv_headers [pvh_xavp.c:485]: pvh_clone_branch_xavi(): cannot clone xavi from non existing parsed_pv_headers" on BYE messages. Interestingly, only on the first attempt to read the from user. I have the following snippet in my `ksr_branch_manage` KSR.log('notice', f'ksr_branch_manage: 1') KSR.log('notice', f'ksr_branch_manage: 2: {KSR.pv.get('$x_fU')}') KSR.log('notice', f'ksr_branch_manage: 3: {KSR.pv.get('$x_fU')}') The result in the log is Jul 15 15:25:03 ttel /usr/sbin/kamailio[816424]: NOTICE: {1 402 BYE 1640171511-5060-41(a)BA.A.B.BCJ} <core> [core/kemi.c:155]: sr_kemi_core_log(): ksr_branch_manage: 1 Jul 15 15:25:03 ttel /usr/sbin/kamailio[816424]: ERROR: {1 402 BYE 1640171511-5060-41(a)BA.A.B.BCJ} pv_headers [pvh_xavp.c:485]: pvh_clone_branch_xavi(): cannot clone xavi from non existing parsed_pv_headers Jul 15 15:25:03 ttel /usr/sbin/kamailio[816424]: NOTICE: {1 402 BYE 1640171511-5060-41(a)BA.A.B.BCJ} <core> [core/kemi.c:155]: sr_kemi_core_log(): ksr_branch_manage: 2: 200102 Jul 15 15:25:03 ttel /usr/sbin/kamailio[816424]: NOTICE: {1 402 BYE 1640171511-5060-41(a)BA.A.B.BCJ} <core> [core/kemi.c:155]: sr_kemi_core_log(): ksr_branch_manage: 3: 200102 Even if there is an ERROR, it seems to be able to read the from user from pv_headers. Is this error just informative and uses the wrong severity, or what does it mean? I have no modparam for pv_headers. It uses the default configuration and I'm on 5.8.6. Regards Mathias

1 1

TOPOS module routing loop with rogue Route headers
by David Tek 17 Jul '25

17 Jul '25

Hello, Setup: I activated the TOPOS module in my setup that looks something like this: SIP Device ---> Proxy1(w/ topos) ---> Proxy2(no topos) ---> ??? ---> Freeswitch. I am using Kamailio 6.0.1 (or 6.0.2, not sure). Both my proxies and the other nodes I don't control in ??? part all add Record-Route/Route headers for in-dialog requests. So an INVITE coming from the SIP Device will go through these proxies, on the 200 OK there are 3+ Record-Route headers which are consumed by TOPOS on Proxy1. With my baresip and Yealink phones, the phones send a 200 OK with NO Route headers and the Route headers are readded by the TOPOS module before it hits the routes in kamailio. Problem: Some SIP devices(I used an older Aastra device to reproduce the issue) like to add a Route header with the target proxy, and so Kamailio receives something like this(some headers stripped for brevity): ACK sip:atpsh-68769b30-2b-1@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.34.122:5060;branch=z9hG4bKe3a6fe355f721ae5b Route: <sip:banana.mydomain.tld;lr> From: "12015550000" <sip:12015550000@banana.mydomain.tld>;tag=ba05cb9826 To: <sip:13105550000@banana.mydomain.tld:5060;user=phone>;tag=1SvNyXFeQjUaN Call-ID: 149a3a344dda2a1c CSeq: 12259 ACK With this, Kamailio receives the packet, internally somewhere runs the myself check and sees banana.mydomain.tld isn't a local domain. So it routes to that domain, which happens to resolve to this same Kamailio. So Kamailio routes the packet to itself until Max-Forwards hits 0 and fails. Solutions I tried: So I thought maybe alias_subdomains would help here: loadmodule "corex.so" modparam("corex", "alias_subdomains", "mydomain.tld") I ran this and found that it fixed the routing loop on the ACK, but then I got a routing loop on the BYE instead. So then I thought, maybe I can squeeze this out earlier: event_route[topos:msg-incoming] { if ( proto != TLS || !$tls_peer_verified || $Rp != "5061" ) { xlog("L_NOTICE", "=======> User packet this one $rm\n"); remove_hf("Route"); } else{ xlog("L_NOTICE", "=======> Server packet this one $rm\n"); } This didn't work, it only ever is invoked for SIP OPTIONS. So I tried with event_route[topos:msg-receiving] and saw messages like this: 16(1347) CRITICAL: <core> [core/ip_addr.c:234]: ip_addr2sbuf(): unknown address family 0 I guess $Rp isn't available in this event route. and $proto is always null. If I disable TOPOS completely, the Aastra phone will load the ACK with the Route headers defined by the Record-Route headers in the 200 OK and the call completes normally. I added this to the top of my request_route: xlog("L_NOTICE","$si - $rm - $hfl(Route) $hfl(Via)\n"); and the only route header that appeared for the ACK packet was the single route header. It would seem TOPOS can be 'defeated' by a rogue Route header. I started reading through the source code on this but could not pinpoint any code that would 'not restore TOPOS stripped headers when a rogue Route header appears because the Route header doesn't match the myself condition of Kamailio'. I even tried adding to topos:msg-sending: append_hf("Route: <sip:invalid.tld>\r\n"); To trick the phone into something that is predictable so I can strip it before TOPOS parses the headers. I'm not sure what to do next, I don't want to go back to TOPOH because I need the packet size savings(some users have issues with big packets ). I've seen other user-agents in the wild beyond these Aastras add Route headers in unexpected places, so this is not a single issue. I even went so far as to ask AI some help, but it just hallucinated the entire time. (AI says this should say 'I also explored AI assistance but didn't find relevant solutions for this specific scenario.' ). Any suggestions on what to do next ? Thanks, David

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sr-users July 2025