RTP + SIP Kamailio or FreeSwitch Proxy to solve NAT problem

Problem

We have a SIP server which does not handle multiple interfaces well.

The primary interface is used to serve clients (Network "A")

Another interface is needed to access an upstream provider (Network "B"). At the moment we have a work around where a Asterisk relay is in place on a separate server ( so call flows from A can be passed to B, and calls from B can be passed to A).

Requirements

We need a SIP compliant proxy to run on a dedicated server configured to replace the Asterisk Relay.

We need the system to be able to:

1) Pass through SIP and RTP between Network A and Network B with NO authentication.

2) A true SIP proxy - (eg no separate call-legs)

Important. There is one host on each network that we need to communicate with (a single IPV4 static IP address on each end)

Notes/requests:

1)We will configure the server ourselves

2) We are using Debian Stable, 64 bit.

3) We will not external access, we are after instructions (a detailed step-by-step how-to-guide for this to be completed on a Debian 7 64bit stable system)

4) We require basic comments in the configuration/how-to-guide

5) We require the work, comments, and communication to be done in English

6) We require a detailed listed of the ports and protocols that need to be exposed between the hosts.

7) NAT of any sort can not be used. This needs to be proxy.

About us.

We are confident in:

The server operating system installation

Networking

SIP & RTP

In general IT/server/networking competent

However we do not have any real experience using Kamailio or Freeswitch - so help is sought with the configuration.

Asterisk PBX Linux Arquitectura de software VoIP

Nº del proyecto: #6683172

Sobre el proyecto