Gateway VoIP Unify Mediatrix 4102

The Unify Mediatrix 4102 VoIP Gateway connects analog phones, faxes and modems to IP communication systems while preserving existing analog infrastructure. Available in the 4100 Series family, the 4102 supports multiple voice codecs and common VoIP protocols to deliver high-quality voice transmission and flexible deployment options for branch offices, multi-tenant sites and medium-size enterprises.

Advantages

The Mediatrix 4102 provides interoperable operation with equipment from leading vendors, supports simultaneous high-compression codecs on each analog port to save bandwidth, and includes security features such as TLS/SRTP, certificate management and HTTPS for protected management, signaling and media. It supports intelligent PSTN bypass for emergency calls during power outages or network failures, offers web-based setup and encrypted auto-provisioning via TFTP/HTTP, and enables reuse of existing analog phones and faxes to reduce installation and move/change costs.

Suitable for

• Branch office connectivity to an IP network
• Medium-size enterprise sites maintaining analog endpoints
• Multi-tenant buildings with mixed analog/IP equipment
• Service providers deploying VoIP gateways for customers

Key specifications

• Model: Unify Mediatrix 4102 (part of Mediatrix 4100 Series)
• MPN: L30220-D600-A214
• EAN: 4050026002065
• Interfaces: FXS analog ports (series available in 2/4/8/16/24 ports; 4102 specific port count per product data)
• Voice codecs: G.729A, G.729B, G.711 (and support for high compression codecs)
• Protocols: SIP, MGCP
• Management: SNMP, TR-069, web interface, encrypted auto-provisioning (TFTP/HTTP)
• Security: TLS, SRTP, HTTPS, certificate management, AES encryption
• PSTN bypass: Intelligent PSTN bypass for emergency calls during power loss or network failure

How to use

Install the Mediatrix 4102 in the network rack or on a flat surface and connect analog phones, faxes or modems to the device’s FXS ports. Connect the unit to the IP network using the Ethernet interface and configure network and VoIP settings via the web interface or a provisioning server. Load credentials and certificates if required for TLS/SRTP. Verify codec and SIP/MGCP settings, and test inbound and outbound calls including emergency PSTN fallback to confirm proper operation.

Recommendations

Place the gateway on a reliable power source and configure encrypted provisioning and TLS/SRTP to secure signaling and media. Verify and enable PSTN bypass where local regulations or business continuity require emergency call support. Use high-compression codecs if bandwidth is constrained and ensure firmware and certificate updates are applied regularly.