Trunking Gateway / Gateway

GYXS-100 trunking gateway is one of the products in VoIP product series. GYXS-100performs IP to PSTN interconnection function by packetizing and depacketizing voice calls between IP and PSTN. It is widely used for PSTN termination for VoIP calls, or providing PBX access to IP networks. GYXS-100 is specifically designed for service providers and large enterprises to provide VoIP services. With 1U form factor, a GYXS-100 is equipped with a main control module, two line module slots (for one or two 4-port T1/E1 modules), and two power supply slots (for one or two AC or DC power supplies). Voice processing capability is the most important requirement and measurement for all trunking gateways. GYXS-100 configured with one to four T1/E1 interfaces can maximally process 120 simultaneous VoIP calls. GYXS-100 embeds powerful DSP technology which possesses 4800 MIPS processing power, for full load voice processing to support CODEC (G.711, G.729A, G.723), echo cancellation (G.168), DTMF relay (RFC2833), and fax relay (T.30, T.38). GYXS-100 supports many types of features to satisfy various application requirements. Features include calling/called party number translation, routing selection, RADIUS billing interface, second stage dialing and ringback tone, DTMF auto dial, tone detection, RTP voice proxy (for NAT traversal), T.38 fax relay. GYXS-100 supports ISDN PRI and SIP protocols and it is the ideal PSTN termination trunking gateway for IP phones, PC phones and PBX bridge to VoIP service. Like other VoIP products, GYXS-100 has the Web GUI and text configuration interfaces. In addition, a user can remotely upgrade, maintain, download usage logs from and collect statistics from an installed GYXS-100 by using EMS system. Compared to other similar products on the market, GYXS-100 embeds newest VoIP technology and standards, and is an open platform to support many new telecommunication applications to come. Specifications: 1) Flexible configuration of one to four T1/E1 2) Toll voice quality 3) Guarantee 120 DS0 voice processing with any CODEC 4) Ideal PSTN termination for IP Phones, IP PBX and PBX access 5) Embedded with RTP voice proxy, digit translation, auto dial and RADIUS billing interface 6) New technology including SIP protocol, frequency domain echo cancellation and iLBC CODEC