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VoIP Gateway Yeastar TB400 4x BRI ports, SIP, G.711/G.722/G.726/G.729A, 10/100 Mbps, TLS/SRTP
Yeastar TB400 is a compact VoIP gateway with four BRI ports designed to connect ISDN BRI lines to SIP-based phone systems and SIP trunks. The device supports SIP (RFC3261) signalling and a range of voice codecs including G.711, G.722, G.726 and G.729A, and provides secure transport options via TLS and SRTP. Network connectivity is via a 10/100 Mbps Ethernet interface, and the unit accepts 100–240 V input with a 12 V/2 A output. The TB400 is intended for small office telephony, PSTN migration projects and hybrid SIP deployments where reliable ISDN-to-IP conversion is required.
The Yeastar TB400 delivers flexible ISDN BRI to SIP conversion with support for both media and signaling security, enabling integration into modern IP PBX environments while preserving legacy ISDN service. It supports multiple industry-standard codecs to balance bandwidth usage and voice quality, and its 10/100 Mbps Ethernet and standards-based networking ensure straightforward connectivity in existing networks. The unit’s compact dimensions make it suitable for rack or shelf placement in small equipment rooms.
The TB400 is specifically designed for use with ISDN BRI lines and SIP systems; it is not suitable for environments that require a large number of concurrent PSTN channels beyond the capacity provided by four BRI ports. It also does not provide gigabit Ethernet, so networks that require 1 Gbps LAN connectivity or higher throughput for additional services may need a different gateway.
Install the TB400 in a location with reliable network and power access and connect the ISDN BRI lines to the BRI ports. Connect the Ethernet port to your LAN and configure network settings via the device’s web GUI. Configure SIP trunk or IP PBX settings in accordance with your SIP provider or PBX documentation, select appropriate codecs for your deployment, and enable TLS/SRTP where required for secure transport. Use the GUI to set TE/NT mode and any necessary channel mapping to ensure correct call routing between ISDN and SIP endpoints.
For reliable operation, keep firmware updated, enable TLS and SRTP when transporting calls over untrusted networks, select appropriate codecs (for example G.722 for higher audio quality where bandwidth allows), and verify TE/NT configuration during initial setup to ensure proper ISDN signalling behavior.
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