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         Trunking gateway is one of the models of NGN series developed by GaoKe Communication By IP interface, E1 interface conversion between IP packets and packets of PCM, Trunking Gateway can connect the new generation IP-based voice network to a conventional trunk equipment, such as public switched telephone network (PSTN) end Bureau or tandem office, or enterprise PBX.
         As a carrier-grade device, 1/2/4/8 E1 Trunking SIP Gateway TG6000 is designed for telecom operators, value-added service providers and medium-sized enterprises according to their requirements and design. Efficient software and hardware design and powerful DSP processing power to ensure that the Trunking Gateway at full load traffic conditions can still achieve the conversion PCM voice signals and IP packets, package, complete speech signal codecs such as G.711 and G.729A, and echo cancellation and other major functions. 1/2/4/8 E1 Trunking SIP Gateway TG6000 supports ISDN PRI signaling so as to achieve the call control between the PSTN or PBX.

VoIP channel number: 120/4*E1
Ethernet Interface: 3* 10/100/1000M RJ45(1WAN+2LAN)
Digital Interface: 4*E1 for optional
Digital Interface Type: E1 (75Ω and 120Ω optional)
Call Control Protocol: SIP
PSTN signaling: PRI , 1 signaling , 7 signaling
Characteristics of echo cancellation: ITU-T G.168 2000,128ms(MAX)
Speech coding: ITU-T G.711(A-law/μ-law)、G.723.1、G.729
Fax technology: T.30、VBD、T.38
QoS technology: IPv4 ToS
Environmental Operating Temperature: 0 ℃ ~ 50 ℃; Humidity: 10% to 90%,non condensing
Power supply: +220V AC 0.5A/-48VDC 1.5A 40W
Dimensions(mm)(Width × depth × height): standard 19-inch 1U high rack 367 × 224 × 44mm
Voice Functions: 1*WAN port for active; Ring Groups; Advice of Charge; TCP, UDP & TLS protocols; SIP based NATing; SIP Transparent Proxy; Adaptive and Static Jitter Buffers; Least-Cost Call Routing; Static/Dynamic IP Addressing; Automatic Number, Multi VLAN Trunking; Identification; Call control using SIP Protocol; Packet Filtering; SIP adaptation; Application Level Gateways; SIP header manipulation; Voice Quality Monitoring

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