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Broadcast Fusion Media Broadcasting System

The integration of traditional broadcasting and new media

Embedded digital telephone broadcasting system

The TH806Di networked intelligent hotline telephone guidance system consists of a host, live broadcast room terminals, guidance room terminals, telephone gateways, etc., and is a distributed system based on IP architecture through a local area network. It can also be connected to the AoIP platform for AES67 supervision.
·Enhanced SIP digital telephone coupler
Adopting a loose coupling mode for networking, different access methods can be customized and configured, and different numbers of intelligent terminals can be equipped, providing different work scenarios such as single guidance, multi guidance, no guidance, and only guidance.
·Broadcast level strategy design
Introducing an automatic mixing algorithm for conference systems, the channel gain of both the caller and the host can be dynamically adjusted through gain sharing during multi-party calls, achieving the goal of balanced loudness and reducing total noise.
·Embedded networked hotline telephone guidance function
Embedded intelligent terminal with touch screen and local physical buttons
Backend database support, including the establishment of user profiles and automatic association of incoming calls, black and white list management, entry of relevant textual information, automatic recording, and retrospective of historical records, etc
Open third-party control protocols to support integrated multimedia broadcasting.

Features


SIP (VoIP)+AES67 (AoIP) dual IP network architecture for remote network management
Flexible configuration of SIP gateway (PSTN \ E1) and SIP intelligent terminal with embedded host as the core
Simultaneously equipped with AES/EBU, analog lines, and AES67 interface to connect to the broadcasting system
The system can be set to operate in coupler mode, seat mode, or traditional multi phone mode
Support single guide mode, multi guide mode, no guide mode, and only guide mode
Broadcast grade dedicated DSP algorithms, including Echo Cancellation (LEC), AutoMix, etc
Built in N-1 digital mixing matrix, supports multi-party communication (no need to separately connect PGM signal)
SIP intelligent terminal adopts touch capacitive screen+physical buttons, PoE or local power supply, flexible and convenient operation
Built in user database and log database, automatic association of user information with incoming calls, black and white list management
Automatic recording, supporting local terminal tracing and playback of historical call records
Support internal communication between intelligent terminals, including voice, text, and video (requires expansion of camera module)
Open control protocol, able to collaborate with third-party multimedia systems (customizable intelligent guidance module)

Interface indicators


host
Standard channel: 6 SIP external lines simultaneously connected, processed, and broadcasted
SIP access: 8-port 10M/100M/1000Mbps (RJ45) (including 6 ports with PoE power supply)
Analog circuit: XLR balanced, reference level+4dBu
Digital line: XLR balanced, compliant with AES3 and GY/T 158-2000
AoIP network: 4 100M/1000Mbps (RJ45 interface)
AoIP protocol: compliant with GY/T 304-2016 and GY/T 322-2019 standards
Digital processing: adaptive filtering (LEC), automatic mixing (Auto Mix), automatic gain (AGC)
Auxiliary interface: RS485 CANBUS (DB9)
System Management: Web based
Host size: 19 inches 1U height depth 350mm
Power consumption: Dual power supply (AC 85-245V/45-65Hz), overall power consumption of 30 watts

smart terminal
网络端口: 10/100/1000Mbps, 支持PoE 供电
人机操作:7 寸1024x600 彩色触摸屏,26键本地键盘
产品尺寸:(角度1): 268x185x189mm
(角度2): 268x201x168mm
功耗(PoE): 1.596~11.86W
SIP gateway (PSTN) end
Number of external lines: 8 FXOs (RJ11)
SIP network interface: 1 WAN, 1 PC (10M/100Mbps, RJ45)
Management interface: 1 RS485, 1 local debugging port
Busy function: 8 independent settings, panel display
Product size: 19 inches 1U height depth 350mm
Lightning protection: Level 1
Power consumption: Dual power supply (AC 85-245V/45-65Hz), overall power consumption of 12 watts

VoIP features
Call control: SIP2.0 over UDP/TCP/TLS, MGCP
Network protocols: SSH, HTTP, HTTPS, DHCP, PPPoE, Radius, DNS, STUN
Speech coding: Broadband ADC/DAC 16KHz sampling, G.711a/u, G.729AB, G.722
Caller ID: Bellcore/Telcordia, ETSI, BT, DTMF
QoS:DiffServ、TOS、802.1P/Q VLAN

Service Hotline:
0512-68079850/51/52/53
Suzhou Headquarters
Telephone:0512-68258269
fax:0512-68079850-8005
Email:sales@szfortune.com
address:Building B, Research Complex, No. 18 Kechuang Road, High tech Zone, Suzhou City, Jiangsu Province
Copyright: Suzhou Fuchuan Technology Co., Ltd Record number:苏ICP备16046229号-2 技术支持:拾久科技