Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian
El Sat, 20 Apr 2024 17:28:09 -0000 "christian.marinelli--- via sr-users" sr-users@lists.kamailio.org escribió:
Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance
Do you bridge the audio in the rtp proxy the same way you bridge the signaling in the sip proxy?
Check your sip trace and makes sure clients connected over wireguard are not offering SDP over IP address that is not routable over wireguard or not even allowed over allowed-ips of wireguard client config.
Faced the same when clients connect over wireguard vpn long time ago and they used to send their public ip in the sdp body while i had no stun/turn/ice configured anywhere for the media to pass over public internet properly. ________________________________ From: christian.marinelli--- via sr-users sr-users@lists.kamailio.org Sent: Saturday, April 20, 2024 9:28:09 PM To: sr-users@lists.kamailio.org sr-users@lists.kamailio.org Cc: christian.marinelli@hotmail.it christian.marinelli@hotmail.it Subject: [SR-Users] Kamailio works but voice is not present during the calls!
Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
If it's not DNS , it's a firewall...
why don't you share us a nice trace of the call, the INVITE and the OK packets is what we're looking for. Ngrep/sngrep are great tools for you to achieve this.
Op za 20 apr 2024 om 21:22 schreef Mahmood Alkhalil via sr-users < sr-users@lists.kamailio.org>:
Check your sip trace and makes sure clients connected over wireguard are not offering SDP over IP address that is not routable over wireguard or not even allowed over allowed-ips of wireguard client config.
Faced the same when clients connect over wireguard vpn long time ago and they used to send their public ip in the sdp body while i had no stun/turn/ice configured anywhere for the media to pass over public internet properly.
*From:* christian.marinelli--- via sr-users sr-users@lists.kamailio.org *Sent:* Saturday, April 20, 2024 9:28:09 PM *To:* sr-users@lists.kamailio.org sr-users@lists.kamailio.org *Cc:* christian.marinelli@hotmail.it christian.marinelli@hotmail.it *Subject:* [SR-Users] Kamailio works but voice is not present during the calls!
Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hi, thank you... Did you refer to the Kamailio log file? Am i right? Christian
checkout the tool sngrep, and try to export us some call traces, they will tell us a lot
Op ma 22 apr 2024 om 16:47 schreef christian.marinelli--- via sr-users < sr-users@lists.kamailio.org>:
Hi, thank you... Did you refer to the Kamailio log file? Am i right? Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hi, thank you for your answer. I don't use FIREWALL or/and NAT on my net, so i think there aren't problems with this component.... I can help you with some configurations files or logs? Christian
Ciao Christian
Unless you use rtpengine or similar, Kamailio won't do anything with RTP.
So I fear, you have to sniff the INVITE + DSP and 200 OK + SDP of one of those calls without audio. (tcpdump -nvs0 port 5060 on the kamailio host for example)
Look at the
c=IN and m=audio lines in the SDP
The c=IN contains the IP address, the m=audio the port
Can the IP in the INVITE directly communicate wit the IP in the 200 OK Reply and vice versa, or is one of them a 'private' IP behind NAT?
Can you ping from one ip to the other? What does traceroute from one ip to the other say?
Mit freundlichen Grüssen
-Benoît Panizzon-
Hi, thank you... This is a tcpdump made during a call without voice (the problem):
https://docs.google.com/document/d/1xrxcJhTpw62z7TBymXOMecSauKqT3ZCJIc7gBldV...
And this is a packets sniff on the net:
https://drive.google.com/file/d/1emKfQmnvrpAjH-Z3KCHZl7fpipEw4Bzm/view?usp=s...
I had to upload them on Drive because there aren't a code function on this forum and the possibility to upload an image. Christian
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, 2024 at 7:11 PM christian.marinelli--- via sr-users < sr-users@lists.kamailio.org> wrote:
Hi, thank you... This is a tcpdump made during a call without voice (the problem):
https://docs.google.com/document/d/1xrxcJhTpw62z7TBymXOMecSauKqT3ZCJIc7gBldV...
And this is a packets sniff on the net:
https://drive.google.com/file/d/1emKfQmnvrpAjH-Z3KCHZl7fpipEw4Bzm/view?usp=s...
I had to upload them on Drive because there aren't a code function on this forum and the possibility to upload an image. Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Ciao Christian
PCAP file would be easier to read. But let's see..
1st Leg:
Linode Client1 IP: 93.189.137.22 => Kamailio Server: 44.3.44.40 (73 from HB9EUE :-) )
That Linode Client has RTP on: 192.168.1.21
2nd Leg:
Kamailio (44.3.44.40) => Client2 (88.116.10.134) Grandstream
RTP Side 1: 192.168.1.21
Grandstream => Kamaililio 200 OK + SDP
Grandstream tells: RTP on: 188.116.10.134
So, the question is: Can 188.116.10.134 <=> 192.168.1.21 communicate with each other?
I fear not! 192.168.1.21 is obviously behind NAT.
93.189.137.22 probably is a NAT router. Maybe it has a SIP ALG setting which would take care to open ports and rewrite the RTP IP address? If not, you loose!
Kamailio does nothing with the RTP audio stream. So both connected clients would need to be able to directly send UDP packets to each other.
OR: You need to 'backhaul' the rtp audio stream via some RTP proxy like rtpengine which could detect NAT and try to send the RTP stream back to the IP it received it from, instead of the one advertised in the SDP.
Mit freundlichen Grüssen
-Benoît Panizzon-
maybe you need to register to VPN IP , not to the public IP of Kamailio
*Antony* satskiy.a@gmail.com +380669197533 +48727830247
On Tue, 23 Apr 2024 at 09:59, Benoit Panizzon via sr-users < sr-users@lists.kamailio.org> wrote:
Ciao Christian
PCAP file would be easier to read. But let's see..
1st Leg:
Linode Client1 IP: 93.189.137.22 => Kamailio Server: 44.3.44.40 (73 from HB9EUE :-) )
That Linode Client has RTP on: 192.168.1.21
2nd Leg:
Kamailio (44.3.44.40) => Client2 (88.116.10.134) Grandstream
RTP Side 1: 192.168.1.21
Grandstream => Kamaililio 200 OK + SDP
Grandstream tells: RTP on: 188.116.10.134
So, the question is: Can 188.116.10.134 <=> 192.168.1.21 communicate with each other?
I fear not! 192.168.1.21 is obviously behind NAT.
93.189.137.22 probably is a NAT router. Maybe it has a SIP ALG setting which would take care to open ports and rewrite the RTP IP address? If not, you loose!
Kamailio does nothing with the RTP audio stream. So both connected clients would need to be able to directly send UDP packets to each other.
OR: You need to 'backhaul' the rtp audio stream via some RTP proxy like rtpengine which could detect NAT and try to send the RTP stream back to the IP it received it from, instead of the one advertised in the SDP.
Mit freundlichen Grüssen
-Benoît Panizzon-
I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________
Zurlindenstrasse 29 https://www.google.com/maps/search/Zurlindenstrasse+29?entry=gmail&source=g Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________ __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hi HB9EUE, 73! This is a pcap file that i got with sngrep, a simple call with two client without voice:
https://drive.google.com/file/d/1WaGvBjKY46IB_hl2cpgWXNYb11Lp9B4v/view?usp=s...
I think the problem is precisely the fact that the two clients do not communicate with each other. If you can confirm this by the pcap file, is there a way to force the clients to transmit and receive the voice throught the SIP server intead of from a direct connection? Thank you in adavance
Christian de IU6DJR
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 instance modify SDP content and replace internal/private IP addresses with the public addresses 3 - let everyone communicate via VPN and using internal/private IP addresses
I would choose option #1
*Sérgio Charrua*
On Tue, Apr 23, 2024 at 5:55 PM christian.marinelli--- via sr-users < sr-users@lists.kamailio.org> wrote:
Hi HB9EUE, 73! This is a pcap file that i got with sngrep, a simple call with two client without voice:
https://drive.google.com/file/d/1WaGvBjKY46IB_hl2cpgWXNYb11Lp9B4v/view?usp=s...
I think the problem is precisely the fact that the two clients do not communicate with each other. If you can confirm this by the pcap file, is there a way to force the clients to transmit and receive the voice throught the SIP server intead of from a direct connection? Thank you in adavance
Christian de IU6DJR __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hi Sergio, thank you for you support and thank you about the three posibilities you suggest to me!
I prefer the second one because the third could create problems with other services in the VPN and the first one is a little bit difficult for a neophyte user with SIP concepts. So i prefer the second one! Can you suggest me how i can change my Kamailio configuration to do this behavior? Thanks you so much in advance Christian
christian.marinelli@hotmail.it wrote:
Hi Sergio, thank you for you support and thank you about the three posibilities you suggest to me!
I prefer the second one because the third could create problems with other services in the VPN and the first one is a little bit difficult for a neophyte user with SIP concepts. So i prefer the second one! Can you suggest me how i can change my Kamailio configuration to do this behavior? Thanks you so much in advance Christian
Hi @sergio charrua, the last days i tryed to search and study how to set my Kamailio server to modify the SDP content and replace internal/private IP addresses with the public addresses, but...without success! :( Can you help me to understand how can i modify the Kamailio configuration to do this? Thank you in advance Christian
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:
Configuring RTPEngine in Kamailio: A Quick Guide – Blog for the Tech Community (voipnuggets.com) https://voipnuggets.com/2023/06/26/configuring-rtpengine-in-kamailio-a-quick-guide/
And good old Nick's headbutts with networking Kamailio Bytes – Setting up rtpengine in Kamailio to relay RTP / Media | Nick vs Networking https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengine/
*Sérgio Charrua*
On Fri, Apr 26, 2024 at 11:48 AM christian.marinelli--- via sr-users < sr-users@lists.kamailio.org> wrote:
christian.marinelli@hotmail.it wrote:
Hi Sergio, thank you for you support and thank you about the three posibilities you
suggest to me!
I prefer the second one because the third could create problems with
other services in the
VPN and the first one is a little bit difficult for a neophyte user with
SIP concepts.
So i prefer the second one! Can you suggest me how i can change my Kamailio configuration to do this
behavior?
Thanks you so much in advance Christian
Hi @sergio charrua, the last days i tryed to search and study how to set my Kamailio server to modify the SDP content and replace internal/private IP addresses with the public addresses, but...without success! :( Can you help me to understand how can i modify the Kamailio configuration to do this? Thank you in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Sergio Charrua wrote:
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:
Configuring RTPEngine in Kamailio: A Quick Guide – Blog for the Tech Community (voipnuggets.com) https://voipnuggets.com/2023/06/26/configuring-rtpengine-in-kamailio-a-quick-guide/
And good old Nick's headbutts with networking Kamailio Bytes – Setting up rtpengine in Kamailio to relay RTP / Media | Nick vs Networking https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengine/
*Sérgio Charrua*
On Fri, Apr 26, 2024 at 11:48 AM christian.marinelli--- via sr-users < sr-users(a)lists.kamailio.org> wrote:
christian.marinelli@hotmail.it wrote: Hi Sergio, thank you for you support and thank you about the three posibilities you suggest to me!
I prefer the second one because the third could create problems with other services in the VPN and the first one is a little bit difficult for a neophyte user with SIP concepts. So i prefer the second one! Can you suggest me how i can change my Kamailio configuration to do this behavior? Thanks you so much in advance Christian Hi @sergio charrua, the last days i tryed to search and study how to set my Kamailio server to modify the SDP content and replace internal/private IP addresses with the public addresses, but...without success! :( Can you help me to understand how can i modify the Kamailio configuration to do this? Thank you in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions 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! Edit mailing list options or unsubscribe:
Hi @Sergio Charrua, i tried this solution in a testing environment and i can confirm you it semmes not very difficult, but....i have some doubts:
1) To install the package on a Ubuntu 22.04 server, i need to set a specific repository as you can see in this guide: https://dfx.at/rtpengine/ Moreover, i need to manually install a librery by following this step because it is not available in the standard Ubuntu repository: https://gist.github.com/joulgs/c8a85bb462f48ffc2044dd878ecaa786 Is it correct? As i can read, it is the only way to install the package without compiling it.
2) To install the package, the APT package manager need to download 250MB of packages and dependencies to a total of 700MB of space on the disk and at the end of the installation it seems to compile something. I don't know if this is correct but i think that's a bit umcomfortable (install 700MB of packages for a only one need)...
Are you sure the second solution is not possible? In the production environment i think i can't install all these things.... Thank you a lot Christian
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-... https://nickvsnetworking.com/rtpengine-installation-configuration-ubuntu-20-04-22-04/ As for question #2, certainly most of those 700Mb are related to outdated packages that need some updates.
*Sérgio Charrua*
On Fri, Apr 26, 2024 at 5:42 PM christian.marinelli--- via sr-users < sr-users@lists.kamailio.org> wrote:
Sergio Charrua wrote:
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:
Configuring RTPEngine in Kamailio: A Quick Guide – Blog for the Tech Community (voipnuggets.com) <
https://voipnuggets.com/2023/06/26/configuring-rtpengine-in-kamailio-a-quick...
And good old Nick's headbutts with networking Kamailio Bytes – Setting up rtpengine in Kamailio to relay RTP / Media | Nick vs Networking <
https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengin...
*Sérgio Charrua*
On Fri, Apr 26, 2024 at 11:48 AM christian.marinelli--- via sr-users < sr-users(a)lists.kamailio.org> wrote:
christian.marinelli@hotmail.it wrote: Hi Sergio, thank you for you support and thank you about the three posibilities
you suggest
to me!
I prefer the second one because the third could create problems with
other
services in the VPN and the first one is a little bit difficult for a neophyte user with SIP concepts. So i prefer the second one! Can you suggest me how i can change my Kamailio configuration to do
this
behavior? Thanks you so much in advance Christian Hi @sergio charrua, the last days i tryed to search and study how to set my Kamailio
server to
modify the SDP content and replace internal/private IP addresses with the public addresses, but...without success! :( Can you help me to understand how can i modify the Kamailio
configuration
to do this? Thank you in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions 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! Edit mailing list options or unsubscribe:
Hi @Sergio Charrua, i tried this solution in a testing environment and i can confirm you it semmes not very difficult, but....i have some doubts:
- To install the package on a Ubuntu 22.04 server, i need to set a
specific repository as you can see in this guide: https://dfx.at/rtpengine/ Moreover, i need to manually install a librery by following this step because it is not available in the standard Ubuntu repository: https://gist.github.com/joulgs/c8a85bb462f48ffc2044dd878ecaa786 Is it correct? As i can read, it is the only way to install the package without compiling it.
- To install the package, the APT package manager need to download 250MB
of packages and dependencies to a total of 700MB of space on the disk and at the end of the installation it seems to compile something. I don't know if this is correct but i think that's a bit umcomfortable (install 700MB of packages for a only one need)...
Are you sure the second solution is not possible? In the production environment i think i can't install all these things.... Thank you a lot Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hello,
regarding the package compilation phase at installing rtpengine, this is required if you want to use the kernel module for better performance. If you don't install the kernel module, there should be no compilation and no compiler etc.. installed. Just try if this fits more to your requirements.
Cheers,
Henning
Henning Westerholt wrote:
Hello,
regarding the package compilation phase at installing rtpengine, this is required if you want to use the kernel module for better performance. If you don't install the kernel module, there should be no compilation and no compiler etc.. installed. Just try if this fits more to your requirements.
Cheers,
Henning
Hi @Henning Westerholt, this is interesting but when i install the package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege? Thank you in advance Christian
Hello,
I think you just don't need to install the rtpengine kernel modules package, then it should not be done.
Cheers,
Henning
-----Original Message----- From: christian.marinelli--- via sr-users sr-users@lists.kamailio.org Sent: Montag, 29. April 2024 14:28 To: sr-users@lists.kamailio.org Cc: christian.marinelli@hotmail.it Subject: [SR-Users] Re: Kamailio works but voice is not present during the calls!
Henning Westerholt wrote:
Hello,
regarding the package compilation phase at installing rtpengine, this is required if you want to use the kernel module for better performance. If you don't install the kernel module, there should be no compilation and no compiler etc.. installed. Just try if this fits more to your requirements.
Cheers,
Henning
Hi @Henning Westerholt, this is interesting but when i install the package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege? Thank you in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Henning Westerholt wrote:
Hello,
I think you just don't need to install the rtpengine kernel modules package, then it should not be done.
Cheers,
Henning
-----Original Message----- From: christian.marinelli--- via sr-users <sr-users(a)lists.kamailio.org> Sent: Montag, 29. April 2024 14:28 To: sr-users(a)lists.kamailio.org Cc: christian.marinelli(a)hotmail.it Subject: [SR-Users] Re: Kamailio works but voice is not present during the calls!
Henning Westerholt wrote: Hello,
regarding the package compilation phase at installing rtpengine, this is required if you want to use the kernel module for better performance. If you don't install the kernel module, there should be no compilation and no compiler etc.. installed. Just try if this fits more to your requirements.
Cheers,
Henning Hi @Henning Westerholt, this is interesting but when i install the package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege? Thank you in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions 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! Edit mailing list options or unsubscribe:
Hi @Henning Westerholt, i made some test on a testing environment and i saw there are these packages:
rtpengine - NGCP RTP/media proxy - metapackage rtpengine-daemon - proxy for RTP and media streams used in NGCP, userspace part rtpengine-iptables - IPtables extension module for the kernel-space NGCP media proxy rtpengine-kernel-dkms - IPtables kernel module for the NGCP media proxy - DKMS rtpengine-perftest - helper tool to test rtpengine transcoding performance rtpengine-perftest-data - helper tool to test rtpengine transcoding performance - data files rtpengine-recording-daemon - recording daemon for RTP and media streams rtpengine-utils - scripts and Perl modules for NGCP rtpengine
I used to install the rtpengine package (the first one) that is in real a metapackage that install all the other ones. I tried to install only the rtpengine-daemon and indeed it seems there will be downloaded 150MB of dependencies for a total of 400MB on the disk. Maybe you refered to this? In this way there isn't any compilation....and i hope there will be all we need for the rtpengine configuration :) Christian
Hi Christian
this is interesting but when i install the package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege?
To disalbe kernel module usage:
rtpengine.conf
### for userspace forwarding only: table = -1
Mit freundlichen Grüssen
-Benoît Panizzon-
Benoit Panizzon wrote:
Hi Christian
this is interesting but when i install the package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege?
To disalbe kernel module usage:
rtpengine.conf
### for userspace forwarding only: table = -1
Mit freundlichen Grüssen
-Benoît Panizzon-
Hi @Benoit Panizzon and @Sergio Charrua, i set up rtpengine on my SIP server and i configured it with the guide you suggested to me: https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengin...
Unfortunatly, there is no voice during the calls :) This is the pcap file during the call with rtpengine configured, if it can help you: https://drive.google.com/file/d/1VQVrd8w5UicAsjl095MkwN5gNUcdjfI6/view?usp=d...
To make the test, i use two devices (two smartphone) connected to internet via mobile network (not in the same LAN) and as i can saw from the pcap file, it seems they want to comunicate with different ports from the standard 5060 (i opened the 5060 on my router). This could be the problem? Thanks a lot in advance Christian
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 http://www.voip.pt/* Tel.: +351 callto:+351+91+104+12+6691 631 11 44
Email : *sergio.charrua@voip.pt sergio.charrua@voip.pt*
This message and any files or documents attached are strictly confidential or otherwise legally protected.
It is intended only for the individual or entity named. If you are not the named addressee or have received this email in error, please inform the sender immediately, delete it from your system and do not copy or disclose it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free.
On Tue, Apr 30, 2024 at 8:47 PM christian.marinelli--- via sr-users < sr-users@lists.kamailio.org> wrote:
Benoit Panizzon wrote:
Hi Christian
this is interesting but when i install the package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege?
To disalbe kernel module usage:
rtpengine.conf
### for userspace forwarding only: table = -1
Mit freundlichen Grüssen
-Benoît Panizzon-
Hi @Benoit Panizzon and @Sergio Charrua, i set up rtpengine on my SIP server and i configured it with the guide you suggested to me:
https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengin...
Unfortunatly, there is no voice during the calls :) This is the pcap file during the call with rtpengine configured, if it can help you:
https://drive.google.com/file/d/1VQVrd8w5UicAsjl095MkwN5gNUcdjfI6/view?usp=d...
To make the test, i use two devices (two smartphone) connected to internet via mobile network (not in the same LAN) and as i can saw from the pcap file, it seems they want to comunicate with different ports from the standard 5060 (i opened the 5060 on my router). This could be the problem? Thanks a lot in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Sergio Charrua wrote:
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 http://www.voip.pt/* Tel.: +351 callto:+351+91+104+12+6691 631 11 44
Email : *sergio.charrua(a)voip.pt <sergio.charrua(a)voip.pt>*
This message and any files or documents attached are strictly confidential or otherwise legally protected.
It is intended only for the individual or entity named. If you are not the named addressee or have received this email in error, please inform the sender immediately, delete it from your system and do not copy or disclose it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free.
On Tue, Apr 30, 2024 at 8:47 PM christian.marinelli--- via sr-users < sr-users(a)lists.kamailio.org> wrote:
Benoit Panizzon wrote: Hi Christian
this is interesting but when i install the
package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege? To disalbe kernel module usage:
rtpengine.conf
### for userspace forwarding only: table = -1
Mit freundlichen Grüssen
-Benoît Panizzon- Hi @Benoit Panizzon and @Sergio Charrua, i set up rtpengine on my SIP server and i configured it with the guide you suggested to me:
https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengi%...
Unfortunatly, there is no voice during the calls :) This is the pcap file during the call with rtpengine configured, if it can help you:
https://drive.google.com/file/d/1VQVrd8w5UicAsjl095MkwN5gNUcdjfI6/view?usp=%...
To make the test, i use two devices (two smartphone) connected to internet via mobile network (not in the same LAN) and as i can saw from the pcap file, it seems they want to comunicate with different ports from the standard 5060 (i opened the 5060 on my router). This could be the problem? Thanks a lot in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions 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! Edit mailing list options or unsubscribe:
Hi @Sergio Charrua, this is my kamailio.cfg configuration file: https://docs.google.com/document/d/1HH9Gn8f_31nDhvbpprwKFs535z65C995nVqels7I... If you will need other informations, ask me without problems... Thank you so much Christian
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 much sense...
Same here, though this is more consistent with WITH_NAT check above:
#!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif
but the following line is also unnecessary:
# RTENGINE MODULE
modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223")
On :
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
}
You should add in T_RELAY route (for example) a call to rtpengine_delete, to free up RTP ports once the call is cancelled or terminated
if(is_method("BYE|CANCEL")) { # if(check_route_param("proxy_media=yes")) { rtpengine_delete(); xlog("L_INFO", " RTPEngine Delete from BYE|CANCEL message in route(RELAY)!"); # } }
Add in failure_route a call to rtpengine_delete() call, to free up RTP ports once the call is cancelled or terminated
You should also set the advertising IP address (the public IP address?? the VPN IP address?? not sure...)
Also, the script seems too complex for your needs. Do you need REGISTER? do you need route PRESENCE? and route LOCATION?
*Sérgio Charrua*
*www.voip.pt http://www.voip.pt/* Tel.: +351 callto:+351+91+104+12+6691 631 11 44
Email : *sergio.charrua@voip.pt sergio.charrua@voip.pt*
This message and any files or documents attached are strictly confidential or otherwise legally protected.
It is intended only for the individual or entity named. If you are not the named addressee or have received this email in error, please inform the sender immediately, delete it from your system and do not copy or disclose it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free.
On Wed, May 1, 2024 at 8:49 AM christian.marinelli--- via sr-users < sr-users@lists.kamailio.org> wrote:
Sergio Charrua wrote:
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 http://www.voip.pt/* Tel.: +351 callto:+351+91+104+12+6691 631 11 44
Email : *sergio.charrua(a)voip.pt <sergio.charrua(a)voip.pt>*
This message and any files or documents attached are strictly
confidential
or otherwise legally protected.
It is intended only for the individual or entity named. If you are not
the
named addressee or have received this email in error, please inform the sender immediately, delete it from your system and do not copy or
disclose
it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free.
On Tue, Apr 30, 2024 at 8:47 PM christian.marinelli--- via sr-users < sr-users(a)lists.kamailio.org> wrote:
Benoit Panizzon wrote: Hi Christian
this is interesting but when i install the
package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege? To disalbe kernel
module usage:
rtpengine.conf
### for userspace forwarding only: table = -1
Mit freundlichen Grüssen
-Benoît Panizzon- Hi @Benoit Panizzon and @Sergio Charrua, i set up rtpengine on my SIP server and i configured it with the
guide you
suggested to me:
https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengi …
Unfortunatly, there is no voice during the calls :) This is the pcap file during the call with rtpengine configured, if
it can
help you:
https://drive.google.com/file/d/1VQVrd8w5UicAsjl095MkwN5gNUcdjfI6/view?usp= …
To make the test, i use two devices (two smartphone) connected to
internet
via mobile network (not in the same LAN) and as i can saw from the
pcap
file, it seems they want to comunicate with different ports from the standard 5060 (i opened the 5060 on my router). This could be the problem? Thanks a lot in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions 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! Edit mailing list options or unsubscribe:
Hi @Sergio Charrua, this is my kamailio.cfg configuration file:
https://docs.google.com/document/d/1HH9Gn8f_31nDhvbpprwKFs535z65C995nVqels7I... If you will need other informations, ask me without problems... Thank you so much Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Sergio Charrua wrote:
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 much sense...
Same here, though this is more consistent with WITH_NAT check above:
#!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif
but the following line is also unnecessary:
# RTENGINE MODULE
modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223")
On :
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
}
You should add in T_RELAY route (for example) a call to rtpengine_delete, to free up RTP ports once the call is cancelled or terminated
if(is_method("BYE|CANCEL")) { # if(check_route_param("proxy_media=yes")) { rtpengine_delete(); xlog("L_INFO", " RTPEngine Delete from BYE|CANCEL message in route(RELAY)!"); # } }
Add in failure_route a call to rtpengine_delete() call, to free up RTP ports once the call is cancelled or terminated
You should also set the advertising IP address (the public IP address?? the VPN IP address?? not sure...)
Also, the script seems too complex for your needs. Do you need REGISTER? do you need route PRESENCE? and route LOCATION?
*Sérgio Charrua*
*www.voip.pt http://www.voip.pt/* Tel.: +351 callto:+351+91+104+12+6691 631 11 44
Email : *sergio.charrua(a)voip.pt <sergio.charrua(a)voip.pt>*
This message and any files or documents attached are strictly confidential or otherwise legally protected.
It is intended only for the individual or entity named. If you are not the named addressee or have received this email in error, please inform the sender immediately, delete it from your system and do not copy or disclose it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free.
On Wed, May 1, 2024 at 8:49 AM christian.marinelli--- via sr-users < sr-users(a)lists.kamailio.org> wrote:
Sergio Charrua wrote:
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 callto:+351+91+104+12+6691 631 11 44
Email : *sergio.charrua(a)voip.pt
<sergio.charrua(a)voip.pt>*
This message and any files or
documents attached are strictly
confidential
or otherwise legally protected.
It is intended only for the individual
or entity named. If you are not
the
named addressee or have received this email in error, please inform the sender immediately, delete it from your system and do not copy or
disclose
it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free.
> > > >
On Tue, Apr 30, 2024 at 8:47 PM christian.marinelli--- via sr-users <
sr-users(a)lists.kamailio.org> wrote:
Benoit Panizzon wrote:
Hi Christian
>
this is interesting but when i install the
package all start automatically. How can i disable this function (compilation for kernel module)? Maybe i need to install a different packege? To disalbe kernel
module usage:
>
rtpengine.conf
> ###
for userspace forwarding only:
table = -1
> > > > Mit freundlichen Grüssen >
-Benoît Panizzon-
Hi @Benoit Panizzon and @Sergio Charrua, i set up rtpengine on my SIP server and i configured it with the
guide you
suggested to me:
> >
https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengi
…
>
Unfortunatly, there is no voice during the calls :)
This is the pcap file during the call with rtpengine configured, if
it can
help you:
> >
https://drive.google.com/file/d/1VQVrd8w5UicAsjl095MkwN5gNUcdjfI6/view?usp=
…
> To
make the test, i use two devices (two smartphone) connected to
internet
via mobile network (not in the same LAN) and as i can saw from the
pcap
file, it seems they want to comunicate with different ports from the standard 5060 (i opened the 5060 on my router). This could be the problem? Thanks a lot in advance Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions 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! Edit mailing list options or unsubscribe:
Hi @Sergio Charrua,
this is my kamailio.cfg configuration file:
https://docs.google.com/document/d/1HH9Gn8f_31nDhvbpprwKFs535z65C995nVqels7%...
If you will need other informations, ask me without problems... Thank you so much Christian __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions 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! Edit mailing list options or unsubscribe:
Hi @Sergio Charrua, i followed the guide at this link: https://nickvsnetworking.com/kamailio-bytes-rtp-media-proxying-with-rtpengin... i tought it was all i need for my problem, i added the rtpengine loadmodule because i didn't see it was present. I can remove it and the modparam instruction for rtpengine since it's already there.
In the link above, it suggested to modify the onreply_route[MANAGE_REPLY] and the route[RELAY] functions by adding the rtpengine_manage() call, do you think it's wrong? You suggested me to add a call to rtpengine_delete function on BYE|CANCEL but, where i need to add the SIP server IP ("the advertising IP address")? Thanks in adavnce Christian
Ciao Christian
I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN.
Maybe, provide more insight....
Your Kamailio-Server also acts as the WireGuard Server? All your SIP clients connect using WireGuard?
Can you ping from one WireGuard client to another?
An usual WireGuard Set-Up would look like:
WG-Server: 192.168.10.1/24
Client 1: 192.168.10.2/24 allowed IP: 0.0.0.0/0 (if you route all trafic through the tunnel or 192.168.10.0/24 if you only want to route that network)
Client 2: 192.168.10.3/24 and same allowed IP.
If you by mistake configured a client with a /32 netmask, it will not be able to communicate with the other client, will not be able to exchange direct RTP data.
Also make sure, Kamailio indeed listens to 192.168.10.1 and not to a different IP.
Mit freundlichen Grüssen
-Benoît Panizzon-