We are planning a new voip system. The trunk is provided to us on a private 10.x.x.x/29 range, and therefore not available direct to our end customers. We plan to have some kind of SIP/RTP proxy in front of the trunk, and a PBX in front of this. The trunk doesn't have any authentication, the end customers should connect and authenticate to the PBX in front, and the RTP streams should terminate to the proxy.
Is this possible to achieve by using Kamailio and RTPProxy?
Trunk | | 10.x.x.x/29 | SIP Proxy | PBX Asterisk etc. | Customer
I have tried many different configurations and browsed every bundled example and web, but still I am not able to do what I want. This have already given me some grey hears, so I guess it's time to check if this is even possible. If not, is there any other approach that is more recommendable? If someone has done this before I would appreciate some example configuration that I could work with.
Regards
Espen, Norway
Hi Espen,
On 03/16/2010 08:43 AM, Espen Berg wrote:
We are planning a new voip system. The trunk is provided to us on a private 10.x.x.x/29 range, and therefore not available direct to our end customers. We plan to have some kind of SIP/RTP proxy in front of the trunk, and a PBX in front of this. The trunk doesn't have any authentication, the end customers should connect and authenticate to the PBX in front, and the RTP streams should terminate to the proxy.
Is this possible to achieve by using Kamailio and RTPProxy?
yes, if you use kamailio 3.0, the default configuration includes the logic for nat traversal using rtpproxy, being a good starting point.
I assume that you don't deal with NAT case, therefore you just have to remove the nat detection conditions (nat_uac_test(...)) to engage rtpproxy everytime.
Do you need to do rtp bridging between trunk and pbx (ie, trunk and pbx are in different networks that are not routable in between)? If yes, check this example:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mod...
Hope it helps, Ramona
Trunk | | 10.x.x.x/29 | SIP Proxy | PBX Asterisk etc. | Customer
I have tried many different configurations and browsed every bundled example and web, but still I am not able to do what I want. This have already given me some grey hears, so I guess it's time to check if this is even possible. If not, is there any other approach that is more recommendable? If someone has done this before I would appreciate some example configuration that I could work with.
Regards
Espen, Norway
I'm still pretty stucked.
This is the goal.
Cisco trunk <-> Kamailio/RTPProxy <-> Asterisk.
The Cisco trunk is located on a private 10.x.x.x/29 range, the SIP-GW and RTP stream are on public IPs, but only reachable via the private 10.x.x.x network. SER should only be the man in the middle and pass traffic from the Asterisk PBX to the trunk and the other way around. Pass as in alter the SIP packages and proxy RTP stream since the end users are not able to talk directly to the 77.x.x.x/10.x.x.x range. Asterisk and the trunk should not register with Kamailio, but pass traffic to each other.
The network:
Customer | Asterisk (Public 217.x.x.100) | SER (nic1: public 217.x.x.99/nic1: 10.x.x.2) | | Trunk (77.x.x.x reachable from the 10.x.x.x network)
I have used the alg and nat example as base, but I still have problems.
Espen
Den 16.03.2010 12:38, skrev Elena-Ramona Modroiu:
Hi Espen,
On 03/16/2010 08:43 AM, Espen Berg wrote:
We are planning a new voip system. The trunk is provided to us on a private 10.x.x.x/29 range, and therefore not available direct to our end customers. We plan to have some kind of SIP/RTP proxy in front of the trunk, and a PBX in front of this. The trunk doesn't have any authentication, the end customers should connect and authenticate to the PBX in front, and the RTP streams should terminate to the proxy.
Is this possible to achieve by using Kamailio and RTPProxy?
yes, if you use kamailio 3.0, the default configuration includes the logic for nat traversal using rtpproxy, being a good starting point.
I assume that you don't deal with NAT case, therefore you just have to remove the nat detection conditions (nat_uac_test(...)) to engage rtpproxy everytime.
Do you need to do rtp bridging between trunk and pbx (ie, trunk and pbx are in different networks that are not routable in between)? If yes, check this example:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mod...
Hope it helps, Ramona
Trunk | | 10.x.x.x/29 | SIP Proxy | PBX Asterisk etc. | Customer
I have tried many different configurations and browsed every bundled example and web, but still I am not able to do what I want. This have already given me some grey hears, so I guess it's time to check if this is even possible. If not, is there any other approach that is more recommendable? If someone has done this before I would appreciate some example configuration that I could work with.
Regards
Espen, Norway
-
Elena-Ramona Modroiu -
Espen Berg