Multi-Network VoIP Routing and SIP Media Interworking Platform

The challenge

The customer, THE GOAL, operated an Internet-facing VoIP platform but needed to establish reliable connectivity with a landline SIP-trunk provider accessible through a separate network.

The existing network arrangement could not simply be extended to the provider without addressing several technical requirements. SIP signalling and RTP media needed to reach the correct provider endpoints, while the Asterisk gateway also had to communicate with the internal VoIP system.

A further requirement was to restrict general Internet connectivity from the Asterisk gateway without preventing the specific connections required for voice services.

The challenge was to bring these separate network environments together, maintain reliable call routing and ensure that the gateway used the designated network paths.

This required more than configuring an ordinary SIP trunk. It involved designing the virtualised infrastructure, separating network interfaces, establishing selective routing and integrating the different components into a working telecommunications platform.

The solution

I designed and implemented a multi-virtual-machine architecture using Debian, Proxmox virtualisation, CentOS and Asterisk 16.

The solution separated the Internet-facing VoIP environment from the dedicated SIP-trunk network. One virtual machine operated the softswitch, while a second virtual machine provided the Asterisk gateway responsible for interworking with the landline provider.

I configured the Asterisk gateway with two network interfaces. The first connected it to the internal VoIP network, and the second provided access to the dedicated provider network.

Rather than allowing unrestricted routing, I configured specific static routes towards the provider's Session Border Controller and designated RTP media endpoint. This allowed the gateway to communicate with the required infrastructure through the appropriate interface.

I also configured the SIP peers and dialplan routing, integrated the softswitch with the Asterisk gateway and tested connectivity between the different network environments.

The completed design provided a controlled communication path between the customer's Internet-facing VoIP platform and the landline SIP-trunk provider.

My responsibility covered the architecture, configuration, network integration, troubleshooting and delivery of the technical implementation on behalf of CRTLHOST.

The outcome

The project was completed in January 2025 and successfully established end-to-end voice communication between THE GOAL's Internet-facing VoIP environment and its landline SIP-trunk provider.

According to the customer's signed confirmation, SIP signalling and RTP media operated through the designated network paths, allowing the two telecommunications environments to interconnect as required.

The implementation also met the customer's requirement for controlled SIP-trunk connectivity with restricted general Internet access for the Asterisk gateway.

The project demonstrated how virtualisation, network segmentation and selective IP routing could be combined to solve a practical telecommunications interworking problem.

THE GOAL subsequently confirmed my responsibility for the architecture, configuration, integration and implementation, as well as the successful delivery of the completed solution.