Hi,
First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one.
I started using kamailio in a test environment about a week ago.
for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream
During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio.
I have installed rtpproxy and it works nicely.
There is just one situation where it fails:
Carrier --> FW (NAT) Kamailio --> asterisk
There is only one way audio (from asterisk to carrier) but not from carrier to asterisk.
I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy...
Thanks
Pemp
Hello,
is Kamailio and RTPProxy (or Asterisk) using public IP addresses? Or they listen on a private address and the firewall does port forwarding of a public IP address?
Cheers, Daniel
On Mon, Sep 16, 2019 at 10:51 AM Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com> wrote:
Hi,
First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one.
I started using kamailio in a test environment about a week ago.
for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream
During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio.
I have installed rtpproxy and it works nicely.
There is just one situation where it fails:
Carrier --> FW (NAT) Kamailio --> asterisk
There is only one way audio (from asterisk to carrier) but not from carrier to asterisk.
I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy...
Thanks
Pemp
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi,
Firewall (PFsense) does port forwarding:
inbound publicIP:5060 --> private ip (192.168.150.119)
inbound PublicIP:10000 to 20000 -> private ip (192.168.150.119)
The public ip is set in the config file (kamailio.cfg) and RTP proxy is started with
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
In the cfg file NATtin is defined. (define with_nat) rtpproxy module is loaded.
This works for most audio situations, except with the carrier Also very important: the carrier i am using for testing requires authentication, so I am using the uac module.
I can see the registration with the provider, i see the inbound call in kamailio, I see the inbound call in asterisk. A connection is set up like it should, but then there is One way audio.
The weird thing is: this route works without a problem: carrier --> FW -->kamailio --> (back through firewall) FW --> user subscribed to kamaiolio. (which should imply the same NAT'ing problems?)
If you need more info, feel free to ask!
PS: What i haven't tried yet is to call from asterisk through kamailio to the carrier number, this is what i'll set up and test next)
Arnout
Op ma 16 sep. 2019 om 13:26 schreef Daniel-Constantin Mierla < miconda@gmail.com>:
Hello,
is Kamailio and RTPProxy (or Asterisk) using public IP addresses? Or they listen on a private address and the firewall does port forwarding of a public IP address?
Cheers, Daniel
On Mon, Sep 16, 2019 at 10:51 AM Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com> wrote:
Hi,
First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one.
I started using kamailio in a test environment about a week ago.
for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream
During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio.
I have installed rtpproxy and it works nicely.
There is just one situation where it fails:
Carrier --> FW (NAT) Kamailio --> asterisk
There is only one way audio (from asterisk to carrier) but not from carrier to asterisk.
I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy...
Thanks
Pemp
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - https://www.asipto.com https://twitter.com/miconda - https://www.linkedin.com/in/miconda Kamailio Advanced Training - https://www.asipto.com/u/kat _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi,
In the meantime I think i figured it out, at least partly. As said I tried setting up outbound calls and learned a lot about the rest of the example config. Especially about how the natting rules are written and how a call follows certain "routes".
In the beginning I had a lot of problems to get the outbound call with the correct headers. (correct from header, correct outbound host...). However, after i got this working i was able to route outbound call's correctly and got audio both ways.
So now I'm thinking that my inbound forwarding is missing some kind of NAT managing. (the outbound is going through the route(RELAY) while inbound calls are pretty much just dumped in the dispatcher ...
Greetings,
Arnout
Op ma 16 sep. 2019 om 14:06 schreef Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com>:
Hi,
Firewall (PFsense) does port forwarding:
inbound publicIP:5060 --> private ip (192.168.150.119)
inbound PublicIP:10000 to 20000 -> private ip (192.168.150.119)
The public ip is set in the config file (kamailio.cfg) and RTP proxy is started with
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
In the cfg file NATtin is defined. (define with_nat) rtpproxy module is loaded.
This works for most audio situations, except with the carrier Also very important: the carrier i am using for testing requires authentication, so I am using the uac module.
I can see the registration with the provider, i see the inbound call in kamailio, I see the inbound call in asterisk. A connection is set up like it should, but then there is One way audio.
The weird thing is: this route works without a problem: carrier --> FW -->kamailio --> (back through firewall) FW --> user subscribed to kamaiolio. (which should imply the same NAT'ing problems?)
If you need more info, feel free to ask!
PS: What i haven't tried yet is to call from asterisk through kamailio to the carrier number, this is what i'll set up and test next)
Arnout
Op ma 16 sep. 2019 om 13:26 schreef Daniel-Constantin Mierla < miconda@gmail.com>:
Hello,
is Kamailio and RTPProxy (or Asterisk) using public IP addresses? Or they listen on a private address and the firewall does port forwarding of a public IP address?
Cheers, Daniel
On Mon, Sep 16, 2019 at 10:51 AM Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com> wrote:
Hi,
First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one.
I started using kamailio in a test environment about a week ago.
for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream
During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio.
I have installed rtpproxy and it works nicely.
There is just one situation where it fails:
Carrier --> FW (NAT) Kamailio --> asterisk
There is only one way audio (from asterisk to carrier) but not from carrier to asterisk.
I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy...
Thanks
Pemp
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - https://www.asipto.com https://twitter.com/miconda - https://www.linkedin.com/in/miconda Kamailio Advanced Training - https://www.asipto.com/u/kat _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
might be just a typing mistake, but:
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
says that -A is private ip, it should be public IP.
Then, towards asterisk, is it everything on a private network, or does it go again through some FW with public ip and port forwarding?
Cheers, Daniel
On Tue, Sep 17, 2019 at 10:23 AM Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com> wrote:
Hi,
Firewall (PFsense) does port forwarding:
inbound publicIP:5060 --> private ip (192.168.150.119)
inbound PublicIP:10000 to 20000 -> private ip (192.168.150.119)
The public ip is set in the config file (kamailio.cfg) and RTP proxy is started with
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
In the cfg file NATtin is defined. (define with_nat) rtpproxy module is loaded.
This works for most audio situations, except with the carrier Also very important: the carrier i am using for testing requires authentication, so I am using the uac module.
I can see the registration with the provider, i see the inbound call in kamailio, I see the inbound call in asterisk. A connection is set up like it should, but then there is One way audio.
The weird thing is: this route works without a problem: carrier --> FW -->kamailio --> (back through firewall) FW --> user subscribed to kamaiolio. (which should imply the same NAT'ing problems?)
If you need more info, feel free to ask!
PS: What i haven't tried yet is to call from asterisk through kamailio to the carrier number, this is what i'll set up and test next)
Arnout
Op ma 16 sep. 2019 om 13:26 schreef Daniel-Constantin Mierla < miconda@gmail.com>:
Hello,
is Kamailio and RTPProxy (or Asterisk) using public IP addresses? Or they listen on a private address and the firewall does port forwarding of a public IP address?
Cheers, Daniel
On Mon, Sep 16, 2019 at 10:51 AM Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com> wrote:
Hi,
First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one.
I started using kamailio in a test environment about a week ago.
for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream
During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio.
I have installed rtpproxy and it works nicely.
There is just one situation where it fails:
Carrier --> FW (NAT) Kamailio --> asterisk
There is only one way audio (from asterisk to carrier) but not from carrier to asterisk.
I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy...
Thanks
Pemp
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - https://www.asipto.com https://twitter.com/miconda - https://www.linkedin.com/in/miconda Kamailio Advanced Training - https://www.asipto.com/u/kat _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Yes sorry, it is a typo.
I meant public ip...
But indeed, it does go once again through the firewall. I am thinking the problem is here:
In the tcpdump (on kamailio) I see it says incoming call from:
sip:number@IP OF CARRIER
Ans it is forwarded this way to pbx
I "think" that this way the PBX will send a reply directly to the IP of the carrier for RTP. When I look at the tcpdump (on the pbx) it does seem this way. (I am still trying to comprehend the whole sip dialog, learning bit by bit)
So, I am trying to change the "from" and "to" header to use the ip of kamailio, and not of the carrier. For this I am using uac module. But no success until now...
I had to do this to get call from extension on pbx, through kamailio , to carrier to work (And this works (hurray)).
(Maybe I'm saying stupid stuff and I'm totally wrong)
Thanks a lot for the help!
Arnout
Op di 17 sep. 2019 om 11:01 schreef Daniel-Constantin Mierla < miconda@gmail.com>:
Hello,
might be just a typing mistake, but:
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
says that -A is private ip, it should be public IP.
Then, towards asterisk, is it everything on a private network, or does it go again through some FW with public ip and port forwarding?
Cheers, Daniel
On Tue, Sep 17, 2019 at 10:23 AM Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com> wrote:
Hi,
Firewall (PFsense) does port forwarding:
inbound publicIP:5060 --> private ip (192.168.150.119)
inbound PublicIP:10000 to 20000 -> private ip (192.168.150.119)
The public ip is set in the config file (kamailio.cfg) and RTP proxy is started with
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
In the cfg file NATtin is defined. (define with_nat) rtpproxy module is loaded.
This works for most audio situations, except with the carrier Also very important: the carrier i am using for testing requires authentication, so I am using the uac module.
I can see the registration with the provider, i see the inbound call in kamailio, I see the inbound call in asterisk. A connection is set up like it should, but then there is One way audio.
The weird thing is: this route works without a problem: carrier --> FW -->kamailio --> (back through firewall) FW --> user subscribed to kamaiolio. (which should imply the same NAT'ing problems?)
If you need more info, feel free to ask!
PS: What i haven't tried yet is to call from asterisk through kamailio to the carrier number, this is what i'll set up and test next)
Arnout
Op ma 16 sep. 2019 om 13:26 schreef Daniel-Constantin Mierla < miconda@gmail.com>:
Hello,
is Kamailio and RTPProxy (or Asterisk) using public IP addresses? Or they listen on a private address and the firewall does port forwarding of a public IP address?
Cheers, Daniel
On Mon, Sep 16, 2019 at 10:51 AM Arnout Van Den Kieboom < arnoutvdkieboom@gmail.com> wrote:
Hi,
First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one.
I started using kamailio in a test environment about a week ago.
for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream
During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio.
I have installed rtpproxy and it works nicely.
There is just one situation where it fails:
Carrier --> FW (NAT) Kamailio --> asterisk
There is only one way audio (from asterisk to carrier) but not from carrier to asterisk.
I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy...
Thanks
Pemp
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - https://www.asipto.com https://twitter.com/miconda - https://www.linkedin.com/in/miconda Kamailio Advanced Training - https://www.asipto.com/u/kat _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - https://www.asipto.com https://twitter.com/miconda - https://www.linkedin.com/in/miconda Kamailio Advanced Training - https://www.asipto.com/u/kat _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello Arnout,
don't worry, everybody started somewhere. Great that it works now. :-)
About the changing the from and to header - there is indeed functionality in the uac module for that, look for the "replace" functions there.
Cheers,
Henning
Am 17.09.19 um 13:35 schrieb Arnout Van Den Kieboom: Yes sorry, it is a typo.
I meant public ip...
But indeed, it does go once again through the firewall. I am thinking the problem is here:
In the tcpdump (on kamailio) I see it says incoming call from:
sip:number@IP OF CARRIER
Ans it is forwarded this way to pbx
I "think" that this way the PBX will send a reply directly to the IP of the carrier for RTP. When I look at the tcpdump (on the pbx) it does seem this way. (I am still trying to comprehend the whole sip dialog, learning bit by bit)
So, I am trying to change the "from" and "to" header to use the ip of kamailio, and not of the carrier. For this I am using uac module. But no success until now...
I had to do this to get call from extension on pbx, through kamailio , to carrier to work (And this works (hurray)).
(Maybe I'm saying stupid stuff and I'm totally wrong)
Thanks a lot for the help!
Arnout
Op di 17 sep. 2019 om 11:01 schreef Daniel-Constantin Mierla <miconda@gmail.commailto:miconda@gmail.com>: Hello,
might be just a typing mistake, but:
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
says that -A is private ip, it should be public IP.
Then, towards asterisk, is it everything on a private network, or does it go again through some FW with public ip and port forwarding?
Cheers, Daniel
On Tue, Sep 17, 2019 at 10:23 AM Arnout Van Den Kieboom <arnoutvdkieboom@gmail.commailto:arnoutvdkieboom@gmail.com> wrote: Hi,
Firewall (PFsense) does port forwarding:
inbound publicIP:5060 --> private ip (192.168.150.119)
inbound PublicIP:10000 to 20000 -> private ip (192.168.150.119)
The public ip is set in the config file (kamailio.cfg) and RTP proxy is started with
rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 -d INFO
In the cfg file NATtin is defined. (define with_nat) rtpproxy module is loaded.
This works for most audio situations, except with the carrier Also very important: the carrier i am using for testing requires authentication, so I am using the uac module.
I can see the registration with the provider, i see the inbound call in kamailio, I see the inbound call in asterisk. A connection is set up like it should, but then there is One way audio.
The weird thing is: this route works without a problem: carrier --> FW -->kamailio --> (back through firewall) FW --> user subscribed to kamaiolio. (which should imply the same NAT'ing problems?)
If you need more info, feel free to ask!
PS: What i haven't tried yet is to call from asterisk through kamailio to the carrier number, this is what i'll set up and test next)
Arnout
Op ma 16 sep. 2019 om 13:26 schreef Daniel-Constantin Mierla <miconda@gmail.commailto:miconda@gmail.com>: Hello,
is Kamailio and RTPProxy (or Asterisk) using public IP addresses? Or they listen on a private address and the firewall does port forwarding of a public IP address?
Cheers, Daniel
On Mon, Sep 16, 2019 at 10:51 AM Arnout Van Den Kieboom <arnoutvdkieboom@gmail.commailto:arnoutvdkieboom@gmail.com> wrote: Hi,
First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one.
I started using kamailio in a test environment about a week ago.
for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream
During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio.
I have installed rtpproxy and it works nicely.
There is just one situation where it fails:
Carrier --> FW (NAT) Kamailio --> asterisk
There is only one way audio (from asterisk to carrier) but not from carrier to asterisk.
I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy...
Thanks
Pemp
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - https://www.asipto.com https://twitter.com/miconda - https://www.linkedin.com/in/miconda Kamailio Advanced Training - https://www.asipto.com/u/kat _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - https://www.asipto.com https://twitter.com/miconda - https://www.linkedin.com/in/miconda Kamailio Advanced Training - https://www.asipto.com/u/kat _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/
On Mon, Sep 16, 2019 at 02:06:43PM +0200, Arnout Van Den Kieboom wrote:
If you need more info, feel free to ask!
Start looking at the SDP contents involved. What are the rtp endpoints in the them in the scenarios you are having problems with (the ipadress in the c= line and the ports in the m= lines).
Also is rtpproxy_manage (or whatever you use) being called unconditionally?
-
Arnout Van Den Kieboom -
Daniel Tryba -
Daniel-Constantin Mierla -
Henning Westerholt