[SR-Users] StriShaken module compilation

Hello all! I am struggling on compiling StriShaken module on RHEL 9.2. So far, the module was compiled as well as libstirshaken. Below are the steps used to compile (as sudo) both library and module (I hope this may help someone) and, of course, correct me if I am wrong in any step or if there is

[SR-Users] Re: Is there any funtion to route based on source phone number

Hi all! I guess you could use dispatcher group ID and depending on the invite phone number, use dispatcher group ID X *Sérgio Charrua* *www.voip.pt * Tel.: +351 91 631 11 44 Email : *sergio.char...@voip.pt * This message and any files or documents attached are

[SR-Users] Re: A question about Db_flatstore Module in Kamailio

Hi Farzaneh, I have a similar script doing log rotation too, and after executing kamcmd, you may remove the old files (*.temp in your case). In my case, instead of deleting file, I move them to another folder (to be processed by the CDR processor) right after executing kamcmd, who will create new

[SR-Users] making HTTP Async request in stateless configuration

Hi all! Just wondering what would be you opinion on the following. We are integrating Kamailio as a stateless redirect server, using SIP 300 Multiple Choices response. The script requires that prior to the redirection, a HTTP request must be made. We thought about using HTTP Async Client to

[SR-Users] Re: base64 decoding issue

Hi all! Solved the issue by using s.decode.base64t instead of s.decode.base64 ... Thanks anyway! *Sérgio Charrua* On Fri, May 10, 2024 at 9:40 AM Sergio Charrua wrote: > Hi all! > > I have been dealing with STIR/SHAKEN for a few weeks now, and while doing > some tests I have found an issue

[SR-Users] base64 decoding issue

Hi all! I have been dealing with STIR/SHAKEN for a few weeks now, and while doing some tests I have found an issue which I can't find the reason for. The script has the following route logic: route[HANDLE_STIRSHAKEN] { xlog("L_INFO", "HANDLE_STIRSHAKEN - STIR/SHAKEN Logic"); if

[SR-Users] Re: Retransmission behaviour after ACK

s.html#next-url) > > > > > > Kaufman > > > > > > *From:* Sergio Charrua via sr-users > *Sent:* Monday, May 6, 2024 6:57 AM > *To:* Kamailio (SER) - Users Mailing List > *Cc:* Sergio Charrua > *Subject:* [SR-Users] Retransmission behaviour after ACK &g

[SR-Users] Retransmission behaviour after ACK

Hi all! I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing. The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules. Request route is simple: request_route { route(HANDLE_OPTIONS);

[SR-Users] Re: Kamailio works but voice is not present during the calls!

Why are you loading rtpengine module first : > # loadmodule for RTPENGINE > loadmodule "rtpengine.so" > and later on do this: > #!ifdef WITH_NAT > loadmodule "nathelper.so" > #!ifdef WITH_RTPENGINE > loadmodule "rtpengine.so" > #!else > loadmodule "rtpproxy.so" > #!endif > #!endif doesn't make

[SR-Users] Re: Kamailio works but voice is not present during the calls!

Please share your kamailio.cfg file, if possible. Do not forget to hide any username & password that may be specified in it (i.e. db connection) *Sérgio Charrua* *www.voip.pt * Tel.: +351 91 631 11 44 Email : *sergio.char...@voip.pt * This message and any files or

[SR-Users] Re: Kamailio works but voice is not present during the calls!

I have no experience with Ubuntu Server, so i'm afraid I can't help you much on this For question #1, try this how-to https://nickvsnetworking.com/rtpengine-installation-configuration-ubuntu-20-04-22-04/ As

[SR-Users] Re: Kamailio works but voice is not present during the calls!

The best solution is to use RTPEngine. Just don't reinvent the wheel, do as "best practices". Also, setting up and configuring RTPEngine is really easy and you would just need to add a couple of lines of code on the kamailio script. Here are a couple of links to help you on that quest:

[SR-Users] Re: Kamailio works but voice is not present during the calls!

Benoit is right! Your issue is a common NAT issue: your SIP signalling is sending public IP address on headers, while the SDP content is publishing internal/private IP addresses: [image: image.png] Possible solutions: 1 - Use RTPEngine and integrate it with Kamailio 2 - let your Kamailio

[SR-Users] Re: Kamailio works but voice is not present during the calls!

Christian, instead of sharing screenshots, please share the SIP PCAP (use tshark/wireshark or sngrep) file with SIP signaling, it will ease the analysis and get you better help. I can't understand why messages are repeated exactly 3 times. Retransmissions?! *Sérgio Charrua* On Mon, Apr 22,

[SR-Users] Re: making HTTP requests in stateless redirects

via sr-users < sr-users@lists.kamailio.org> wrote: > > > > On Apr 8, 2024, at 11:12 AM, Sergio Charrua via sr-users < > sr-users@lists.kamailio.org> wrote: > > > > Also, as this is a stateless script, is there another way of using async > http or making

[SR-Users] Re: making HTTP requests in stateless redirects

*Sérgio Charrua* On Mon, Apr 8, 2024 at 6:20 PM Nick Digalakis wrote: > I didn't test this, but why are you calling t_newtran(); if you only want > to send the 300 response? > > On Apr 8, 2024 18:12, Sergio Charrua via sr-users < > sr-users@lists.kamailio.org&

[SR-Users] making HTTP requests in stateless redirects

Hi all! For testing purposes, while I am waiting for the ST/SH REST API to be available from other teams, I developed a small python REST API that returns a mockup of a JSON object with some of the required values. The kamailio script, completely stateless as I do not need to keep track of the

[SR-Users] Re: low performance with no apparent reason

s this > is per process. This much is never needed. > > On the other hand, you should probably increase the shared memory if you > are having a lot of transactions going on, TLS etc.. This is per server, so > you can configure more. > > > > Cheers, > > > > Hennin

[SR-Users] Re: low performance with no apparent reason

Thank you all for helping! I wasn't expecting such a large number of replies! I ended up partially solving the issue with a different approach. Modifying the size of the UDP Buffer did not reveal any improvement. However, modifying the memory management did improve a lot: from 330 CPS to 1800 CPS

[SR-Users] Re: low performance with no apparent reason

t; > > > I assume that you are using udp. > > > Please increase the length of the udp queue: > > > > https://medium.com/@CameronSparr/increase-os-udp-buffers-to-improve-performance-51d167bb1360 > > > > > > Regards. > > > Ovidiu Sas > > &g

[SR-Users] Re: low performance with no apparent reason

lists.kamailio.org> wrote: > >> I assume that you are using udp. >> Please increase the length of the udp queue: >> >> https://medium.com/@CameronSparr/increase-os-udp-buffers-to-improve-performance-51d167bb1360 >> >> Regards. >> Ovidiu Sas >> &

[SR-Users] low performance with no apparent reason

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216

[SR-Users] Re: direct media between UACs

I have found the issue! For future reference, here is the explanation. The JSON object returned from the Routing Logique Engine is the standard object as per module's description, but with the "extra" property filled with an extra header value. The resulting SIP Message is : INVITE

[SR-Users] Re: direct media between UACs

Hi all! some additional details for this issue. Currently, Kamailio is using RTJSON to get routes from the routing engine and forward calls to the correct route. Please note that the 2 testing endpoints and Kamailio are all in the same network, no NAT involved, and firewalls are disabled!

[SR-Users] direct media between UACs

Hi all! got a weird behavior that I cannot understand the reason for... In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and chan_sip) and 1 Kamailio 5.7 in between. All servers are in the same network, so, there is no NAT involved. No RTPEngine either. Network is

[SR-Users] Re: DMQ module failure

<http://www.voip.pt/>* On Fri, Mar 1, 2024 at 3:52 PM Fred Posner wrote: > > On Mar 1, 2024, at 10:30 AM, Sergio Charrua via sr-users < > sr-users@lists.kamailio.org> wrote: > > > > Hi all! > > > > I'm testing the DMQ module, but having difficulties set

[SR-Users] DMQ module failure

Hi all! I'm testing the DMQ module, but having difficulties setting it up. [...] My kamailio.cfg is as follows : [...] listen=udp:0.0.0.0:5060 listen=tcp:0.0.0.0:5060 [...] modparam("dmq", "server_address", "sip:10.20.0.2:5062") #DMQ_SERVER_ADDRESS modparam("dmq", "notification_address",

[SR-Users] Re: best approach to setup SIP Session sharing between servers

< > sr-users@lists.kamailio.org> wrote: > > > > That would require transaction replication, which Kamailio doesn't have. > > > > Most messages can be processed statelessly, so this isn't a huge > obstacle. However, CANCELs won't work. > > > >> O

[SR-Users] best approach to setup SIP Session sharing between servers

Hi all! I am having some difficulties/doubts on what would be the best approach to let multiple Kamailio servers share their SIP sessions between each other. The goal is to have HA on Kamailio cluster such that if a call is received and initiated on Kamailio #1, and during any moment of the call

[SR-Users] Re: Setting kamailio to use another DB

Hi! I prefer to let Kamailio do what it does best (SIP) and let other tools manage whatever you need to do additionally. In your case, I would use a Keepalived service, shared between the 2 database nodes (or any other number of nodes) and create a Virtual IP (VIP) shared between DB nodes. By