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37507 discussions

Small Question
by Fernando Lopes 03 Jun '25

03 Jun '25

Hello Everyone, I have a cloud vm with public ip: 135.2xx.1xx.1xx:5060 and locally 10.0.0.4 When I receive a SIP message the invite looks like this, it gets stuck here. SIP Invite: 2025/06/03 16:24:58.249284 192.76.120.10:5060 -> 10.0.0.4:5060 INVITE sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060 SIP/2.0 Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=765jrrKarBe3F> Record-Route: <sip:10.255.0.1;r2=on;lr;ftag=765jrrKarBe3F> Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bKbb37.28332dbc5901e729308403cd5e049713.0 v:SIP/2.0/UDP 10.239.44.224:6000;received=10.239.44.224;rport=6000;branch=z9hG4bKD4UtjN52Qrm9D Max-Forwards:55 f:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.com>;tag=765jrrKarBe3F t:<sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060> i:c64314db-75b9-4029-8147-be3f3d77da2a CSeq:99930157 INVITE m:<sip:mod_sofia@10.239.44.224:6000> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY k:timer,path u:talk,hold,conference,refer Privacy:none c:application/sdp Content-Disposition:session l:205 P-Asserted-Identity:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.eu;verstat=No-TN-Validation> v=0 o=Telnyx 1748940128 1748940129 IN IP4 50.114.150.25 s=Telnyx c=IN IP4 50.114.150.25 t=0 0 m=audio 27770 RTP/AVP 9 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:27771 IN IP4 50.114.150.25 a=ptime:20 Kamailio Code: #!define DS_LIST "/etc/kamailio/dispatcher.list" #!define LISTEN_UDP_PUBLIC udp:135.2xx.1xx.1xx:5060 #!define LISTEN_UDP_PRIVATE udp:10.0.0.4:5060 #!define LISTEN_TCP_PUBLIC tcp:135.2xx.1xx.1xx:5060 #!define LISTEN_TCP_PRIVATE tcp:10.0.0.4:5060 #!define EXTERNAL_IP "10.0.0.4" #!define KUBERNETES_NETWORK "172.19." #!define FLAG_TELNYX_CALL 12 #set module path mpath = "/usr/lib/x86_64-linux-gnu/kamailio/modules" ####### Global Parameters ######### #mhomed=1 # doesn't work with virtual IPs #!ifdef WITH_DEBUG debug=4 #!else debug=2 #!endif log_stderror=no memdbg=5 memlog=5 #log_facility=LOG_LOCAL6 log_facility=LOG_LOCAL0 fork=yes children=4 auto_aliases=no #listen=LISTEN_UDP_PUBLIC listen=LISTEN_UDP_PRIVATE listen=LISTEN_TCP_PRIVATE ####### FLAGS ###### #!define FLAG_FROM_ASTERISK 10 #!define FLAG_FROM_PEER 11 sip_warning=no

1 0

Kamailio as Proxy only
by disa_369＠msn.com 03 Jun '25

03 Jun '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have not much experience Kamailio and i dont know what need to run Kamailio as Proxy. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from Avaya PBX -------> Kamailio ----> Video System. We dont need Kamailio subscribers\peers, Just to forward a call from avaya to Video system by ip. Video system has IVR number 3000, all we need to dial number 3000@ip_of_video_system. I got questions. 1. Do we need install maria_db for that? 2. Do we need create subscriber\peer on Kamailio? 3. Do we need add modules for that? And main question what to add in kamailio.cfg is? Thx!!!!!!!!!

2 2

Kamailio Regex
by Timothy Delin 03 Jun '25

03 Jun '25

Hi everyone, I need your advice. I am extracting a SIP Body message from a User Agent Client using the regex as such "$var(calling_party)= $(var(body){re.subst,/.*Calling-party:\s*sip:(\+[0-9]+)@.*/\1/});" And I tried to log it as "xlog("L_INFO", "Extracted number 1: $var(calling_party) <><><>\n");" However, the output in my logs is only showing as below, where I cannot see any value after my variables. "1(6) INFO: {1 10 SUBSCRIBE 5017ae21204c72a3} <script>: Extracted number 1: +6512345678" I suspect this is the reason why when I tried to use this variable in a redis_cmd, it returns as null as the variable may have impacted the command input to Redis. I tried to change the $var(calling_party)="+6512345678" and it is working perfectly fine Best regards, Timothy

3 5

Re: Kamailio as Proxy only
by airsay＠duck.com 30 May '25

30 May '25

I'm not very skilled with Kamailio config. But it sounds like dsiprouter (a graphical UI built ontop of Kamailio) would be your easiest solution. On Dsiprouter, You can set up Avaya as a Carrier Group and Video conference as an Endpoint Group. Then route accordingly. Sent from my iPhone > On 30 May 2025, at 12:19, disa_369--- via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello to everyone. > I deployed Kamailio on Debian system. Install was successfull. > I have not much experience Kamailio and i dont know what need to run Kamailio as Proxy. > I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. > Outgoing calls only from > Avaya PBX -------> Kamailio ----> Video System. > We dont need Kamailio subscribers\peers, Just to forward a call from avaya to Video system by ip. > Video system has IVR number 3000, all we need to dial number 3000@ip_of_video_system. > I got questions. > 1. Do we need install maria_db for that? > 2. Do we need create subscriber\peer on Kamailio? > 3. Do we need add modules for that? > > And main question what to add in kamailio.cfg is? > > Thx!!!!!!!!! > __________________________________________________________ > 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!

1 0

Hyperthreading and high request volumes in modern times
by Brooks Bridges 30 May '25

30 May '25

So, I have a platform that handles in the order of around 2500 or more call attempts per second. There are 2 servers that are built that handle this traffic, "old" and "new". The "old" design is a bit heavy, has multiple redis calls to external IO, does a lot of "work" on each invite before passing it along, etc. The "new" design is extremely lightweight, only a couple of htable comparisons and a dispatcher check on each invite before passing it along. The 2 servers are running different versions of Kamailio, however they're both in the 5.x train and I've found nothing in the changelogs that I believe would explain a significant difference in how child process CPU scheduling is performed, especially a detrimental change. I'm more than happy to be proven wrong about that though! The "old" system is running 5.4.8: This system is running 4 x 8 core CPUs (32 cores) with no hyperthreading (4 x E5-4627 v2 to be exact) and has multiple external IO interactions with things like redis, lots of "work" being done on each invite checking various things before allowing it to proceed. This is currently running 32 child processes and again, has no issues handling the requests whatsoever and has been running at this level and higher for literal years. The "new" system is running 5.8.3: This system is running 2 x 8 core CPUs (16 cores) *with* hyperthreading (2 x E5-2667 v4 to be exact) and has no external IO interactions with anything, and only does a couple of checks against some hash table values and dispatcher before allowing it to proceed. This is also currently running 32 child processes, however I am experiencing spikes in context switching and odd memory access issues (e.g. backtraces show things like "<error: Cannot access memory at address 0x14944ec60>" in udp_rvc_loop) with similar loads. What I'm looking for here is empirical knowledge about things specifically like context switching and if hyperthreading is detrimental in cases where I'm doing very high volume, low latency (e.g. no waiting on external replies, etc) work. Is it possible there are additional issues with the way hyperthreading works and the concepts behind it if the individual calls for CPU resources are so "quick" as to overwhelm the scheduler and cause congestion in the CPU's thread scheduler itself? Have you run into anything like this? Have you discovered that very low latency call processing can incur more hyperthreading overhead than the benefit you get from it? Do you have additional data I may be missing to help me justify to the C suite to change out the system architecture to provide a higher physical core count? Can you share it? Thanks!

6 14

Sca in cluster/load balancer
by mauro.celli76＠gmail.com 29 May '25

29 May '25

We have two kamailio in load balancing, mysql is in cluster mode with galera. We need to use sca, but kamailio db_cluster is not supported (i need to specify connection string to sql not cluster name) and if phone1 is registered on kamailio1 and phone2 on kamailio2, sca not work. It is possibile to implement cluster?

3 5

How to count the first re-register request?
by James Morrison 26 May '25

26 May '25

I want to count the first REGISTER request during an initial register or re-register from UE. In my scscf instance, I have defined att_init_reg and att_rereg counters to count the number of initial registers and re-registers respectively. In my early tests, the following piece of logic worked as expected to filter the very first register request (one that is sent from UE before getting challenged). However, more tests showed that during re-registers, UE sends a REGISTER request, containing an authorization header and non-empty response, in that case the code is unable to distinguish a re-register attempt and the related counter fails to increment. Is there a way to fix this issue? #!ifdef WITH_XHTTP_PROM /* Only count the first register request */ if (!is_present_hf("Authorization") || search_hf("Authorization","response=\"\"","f")) { if ($var(impu_registered) != 1) { prom_counter_inc("att_init_reg", "1"); } else { prom_counter_inc("att_rereg", "1"); } } #!endif

3 5

5.7.7 release, tag and docker build
by Henning Westerholt 23 May '25

23 May '25

Hello, I have noticed that the git tag for release 5.7.7 was missing, I've just set it. The 5.7.7-bullseye docker image is also missing from https://github.com/kamailio/kamailio-docker/pkgs/container/kamailio Not sure if this is done automatically, if not - please can somebody please trigger the workflow to create it? Thank you, Henning

1 0

Help Adding WebRTC to Existing SIP Setup
by Fernando Lopes 22 May '25

22 May '25

Hi everyone, I currently have a working SIP setup, but I'm trying to get WebRTC working alongside it and haven’t had much success so far. Does anyone have tips or best practices for enabling both SIP and WebRTC in the same configuration? Is it possible to handle both within a single Kamailio config? Or would I need some kind of conditional logic (like if/else) to manage how the config handles different types of clients? My setup looks like this: Router → Kamailio → Kubernetes (Asterisk inside the cluster) I was having some issues with the Contact header, but I managed to resolve that. Now I'm stuck figuring out how to make WebRTC work cleanly alongside SIP. Any advice would be appreciated! Thanks!

2 1

Kamailio v5.7.7 Released
by Daniel-Constantin Mierla 21 May '25

21 May '25

Hello, Kamailio SIP Server v5.7.7 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.7. Note that 5.7 is now the third last stable branch, v5.7.7 being the last planned released in 5.7.x series. The latest two stable branch are 5.8 and 6.0, with v6.0.1 being released a while ago. For more details about version 5.7.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/05/kamailio-v5-7-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

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