Kamilio 5.5 or opensips 3.2 TCP to UDP Proxy
Budget: $250 – $750 USD
I have a linux server running asterisk running Chan_sip which only support UDP, and can not be upgraded.
The asterisk Server communicates to Phones and SIP TRUNK Providers on UDP port 5060 using the LAN Public IP X.X.X.X interface.
I need to have some new phones connect to the Server over TCP Port 5062
I need to have some new phones connect to the Server over TLS Port 5089
We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following:
1) The Proxy should communicate with the asterisk server on 127.0.0.1 UDP SIP port 5060.
2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the 127.0.0.1 interface.
3) ALL TCP SIP traffic on 5062 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC)
4) Listen for incoming SIP TLS traffic on 5089 on the LAN IP X.X.X.X. and proxy this SIP TLS Traffic to and from the Asterisk server on UDP Port 5060 using the 127.0.0.1 interface.
5) ALL TCP SIP traffic on 5089 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC)
6) The other phones and the sip trunk providers still use UDP 5060 and should not be proxied via Kamailio , and instead should communicate with the Asterisk server directly.
Deliverables will provide the complete kamailio.cfg or opensips.cfg to allow the remote phone to register on the asterisk server using either TCP or TLS. These phones should be able to make and receive calls just like the current phones connected on UDP Port 5060.
Candidate should have complete understanding of the Proxy solution being provided as well as Asterisk.
this project should not involve any additions to the core asterisk system or any addons. The solution should be a simple SIP Proxy.
The asterisk Server communicates to Phones and SIP TRUNK Providers on UDP port 5060 using the LAN Public IP X.X.X.X interface.
I need to have some new phones connect to the Server over TCP Port 5062
I need to have some new phones connect to the Server over TLS Port 5089
We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following:
1) The Proxy should communicate with the asterisk server on 127.0.0.1 UDP SIP port 5060.
2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the 127.0.0.1 interface.
3) ALL TCP SIP traffic on 5062 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC)
4) Listen for incoming SIP TLS traffic on 5089 on the LAN IP X.X.X.X. and proxy this SIP TLS Traffic to and from the Asterisk server on UDP Port 5060 using the 127.0.0.1 interface.
5) ALL TCP SIP traffic on 5089 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC)
6) The other phones and the sip trunk providers still use UDP 5060 and should not be proxied via Kamailio , and instead should communicate with the Asterisk server directly.
Deliverables will provide the complete kamailio.cfg or opensips.cfg to allow the remote phone to register on the asterisk server using either TCP or TLS. These phones should be able to make and receive calls just like the current phones connected on UDP Port 5060.
Candidate should have complete understanding of the Proxy solution being provided as well as Asterisk.
this project should not involve any additions to the core asterisk system or any addons. The solution should be a simple SIP Proxy.