IDM RG3200 Unified Communication Gateway | VoIP Gateway

IDM RG3200  Unified Communication Gateway

I       System Overview

I.1       Overview

The IDM RG-3200 series is a modular integrated dispatch network gateway developed by Raytrans for specialized applications. It offers powerful signaling and media processing capabilities and includes various communication interfaces. It supports E1, FXO/FXS, 4G LTE, trunking interfaces/base stations/handheld radios, audio/shortwave/ultrahigh frequency, E&M interfaces, magnetic interfaces, and SIP. With dual power supplies, it is mainly used in specific industries such as IPCC, SIP dispatch, and unified communications.

I.2       Performance Features

(1).     Modular design with a 19-inch 3U chassis that can accommodate up to 17 service slots;

(2).    The main unit provides 7 Gigabit Ethernet ports, with 2 ports supporting link aggregation for signaling and media processing. It supports SIP protocol and a range of voice codecs including G.711 U-Law and A-Law, G.711 Appendix 1, G.723.1 and G.723.1 Annex A, G.729 Annex A and Annex B. It supports IP, NAT, ICMP, ARP, HTTP, BOOTP, FTP, TFTP, DHCP, SNMP, Diff-Serv, and other network protocols, as well as the G.168 protocol with up to 128ms delay;

(3).     System clock supports local clock, line recovery clock, and other clock modes;

(4).  Magnetic interface board supports 4 magnetic telephone connections, enabling conversion between magnetic telephone signaling and SIP signaling, with options for standard or 2100Hz carrier magnetic interfaces;

(5).     Analog telephone interface board provides 10 analog telephone (FXO/FXS) interfaces;

(6).    The device supports an SNMP-based network management system;

(7).    E1 card provides 8 E1 interfaces, including 6 G.703 E1 interfaces, 1 passthrough E1 interface, and 1 transmission E1 interface;

(8).  The device offers 17 service slots for inserting various expansion function boards to implement communication transmission/access, signaling conversion, and dispatch functions. Supported expansion boards include:

u       E1 Business Access Board: Supports 8 G.703 E1 interfaces;

u       Audio Business Access Board: Provides 4 E/M 2/4-wire interfaces;

u       LTE Access Board: Supports 1 4G LTE interface;

u       Radio Access Board: Supports 4 radio business interfaces for integrating shortwave/ultrahigh frequency radios into the SIP softswitch system;

u  Trunking Access Board: Supports 4 trunking business interfaces for integrating trunking radios into the SIP softswitch system;

u  Echo Cancellation Board: Supports up to 600ms delay for applications in high-delay scenarios like satellite transmission;

u   Universal Business Board: Provides 4 universal 64K service slots, supporting mixed insertion of various service cards like FXO/FXS, magnetic, E/M 2/4-wire, radio, etc., to achieve various voice services and SIP conversion;

u       Supports co-line voice services with E1 line cascading transmission.

II       Device Structure

The IDM RG-3200 features a standard 3U, 19-inch chassis, designed to be compact and to fully utilize the front panel space for user convenience. The front panel includes Ethernet ports, Ethernet management ports, CPU console interface, power terminals, power switch, and various indicator lights, with button switches and DIP switches.

2.1       Front Panel Diagram

2.2       Rear Panel Diagram

The rear panel provides 17 general-purpose slots from left to right.

III       Networking Applications

Supports up to 170 64K time-slot services, consolidating voice, audio, video, and LAN communications into a single high-speed data stream for bidirectional transmission via DDN leased lines, microwave, satellite, and other communication links. Supports audio and data business passthrough, as well as Ethernet channel physical isolation passthrough. Interfaces for telephone exchange (FXO), phone (FXS), radio/PTT, LTE mobile, magnetic telephone, public address, and mixing console can all be centrally managed at the dispatch station.

IV       Technical Specifications

4.1       VoIP Protocol

u    VoIP Protocol: SIP;

u    Voice Codecs: G.711 U-Law and A-Law, G.711 Appendix 1, G.729 Annex A and Annex B;

u     Network Protocols: IP, NAT, ICMP, ARP, HTTP, BOOTP, FTP, TFTP, DHCP, SNMP, Diff-Serv;

u Voice Processing Technologies: Voice Activity Detection (VAD), Comfort Noise Generation (CNG), Echo Cancellation G.168 (≤128ms delay), etc.;

u     Supports automatic routing;

u     Supports caller ID and call display;

u     Supports generation of dial tone, ringing tone, etc.

4.2       E1 Interface

u       Interface Standard: G.703 / G.704;

u       Connector Type: DB37, with DB37 to micro-coaxial adapters;

u       Data Rate: 2.048Mbps;

4.3       Serial Interface

u     Interface Standard: EIA-422/EIA-232;

u     Connector Type: 4 RJ45 physical interfaces, with DB25 conversion cables and 4 DB25 interfaces;

u   Asynchronous Interface Rates: Configurable to 300bps, 600bps, 1200bps, 2400bps, 4800bps, 9600bps, 19200bps, 38400bps, 57600bps, 115200bps;

u Synchronous Interface Rates: Configurable to 1200bps, 2400bps, 4800bps, 9600bps, 19200bps, 38400bps, 57600bps, 115200bps;

4.4       Analog Voice Interface

u       Connector Type: RJ45;

u       Interface Types: FXS, FXO, E&M VOC, MSG;

u       Codec: G.711 (64k);

u       Compression: ITU G.711 A-Law;

u       Audio Range: 300Hz – 3400Hz;

u       Impedance: Balanced 600Ω;

4.5       Ethernet Interface

u       Network Connection: End-to-end bridging;

u       Standard: IEEE802.3;

u       Connector Type: RJ45;

u       Data Rate: 10/100Mbps;

4.6       HDMI Interface

u       Sampling Rates: 16k, 32k, 44.1k, 48k;

u       Bitrate Range: Configurable, recommended 64k-256kbps;

u       Encoding Formats: AAC, PCMA, MP3, MP2;

u       Gain Adjustment: -24dB to +24dB, default 0 (no gain adjustment);

u       Bitrate Range: 256K-4Mbps;

u       Bitrate Control: CBR, VBR, AVBR, FIXQP;

u       Encoding Formats: H.264, H.265;

u       Encoding Protocols: H.265 Main Profile Level 5.0, H.264 Baseline/Main/High Profile Level 5.1;

u       Intelligent Encoding: SmartP, DualP, ROI;

u       HDMI Input Capability: 4K30;

u       Encoding Capability: 4K30x3 or 1080P30x12;

u       Frame Types: I/P;

u       Decoding Capability: 1080P30x8;

4.7       High-Fidelity Audio Interface

u       Input Interfaces: Supports HDMI, 3.5mm Line-In;

u       Output Interfaces: Supports HDMI, 3.5mm Line-Out;

u       Sampling Rates: Supports 48K, 44.1K, 32K, 16K, 8K;

u       Encoding Formats: Supports AAC, MP3, MPEG2, PMCA, PMCU;

4.8       Magnetic Interface

The magnetic interface board supports 4 magnetic connections, enabling conversion between magnetic telephone signaling and SIP signaling, supporting both standard and 2100Hz carrier magnetic interfaces.

(1).Standard Magnetic

u Conversion between user voice signals and motherboard signals; conversion between digital signaling and line status, providing identification and status information for network management;

u     Can be used as a connection interface between magnetic telephones, magnetic switches, and between them.

(2). 2100Hz Carrier Magnetic

u  Conversion between 2100Hz carrier signaling and ringing or 16-time-slot digital signaling, providing identification and status information for network management;

u  Converts 2100Hz in-band signaling from a remote PCM device’s audio relay card into a ringing signal output and converts received ringing signals into 2100Hz in-band signaling for the remote end;

u   Converts local 2100Hz in-band signaling into 16-time-slot digital signaling for the remote end and converts remote PCM’s 16-time-slot digital signaling into 2100Hz output.

4.9       Trunking Interface

u Supports SIP protocol and can join SIP softswitch server conferences;

u Supports PTT intercom signaling, with definable E and M signaling effectiveness, and supports PTT handsets and trunking radio access.

4.10       Radio Interface

u  Provides functionality to disable listening, mute, record, delete, add, etc., for any radio at the operation control end;

u  Control console can apply PTT control mode, VAD voice activation mode, or PAD console operation mode for voice transmission to the radio. Enables sending and receiving permissions through the console;

u  The SIP integrated dispatch system can send instructions to the radio gateway via the network, with the gateway forwarding (pass-through) to the radio and supporting querying, modifying, and deleting radio parameters;

u    Supports interfacing with 146 radios

u via two-wire/four-wire (FSK modulation) using full-duplex asynchronous serial communication at a rate of 1.2kbps;

u Provides 2 audio and PTT interfaces for interfacing with shortwave radios, ultrahigh frequency radios, and data links, supporting voice communication and control with satellite and trunking systems;

u  Realizes voice interoperability and data control between radios and the SIP integrated dispatch system through the radio gateway;

u    Supports network control of radio PTT function;

u    Supports network control of duty functions, with the ability to query the current channel, squelch status, etc.;

u   Supports network channel switching and squelch control;

u   Features WEB interface configuration management for SIP data configuration, volume control, noise control, and radio control.

4.11       E&M 2/4-Wire Audio

u       Supports 2-wire and 4-wire modes, providing identification and status information for network management;

u       Completes conversion between E&M signaling and SIP signaling, and IP transmission of audio information;

u       Supports transmission via E1 interface;

u       Supports co-line voice.

V       Functional Parameters

5.1       Operating Environment

u       Temperature: 0ºC–40ºC;

u       Humidity: 90% (non-condensing);

5.2       Dimensions

IDM RG-3200: 19-inch 3U standard chassis; Width x Depth x Height: 485 x 310 x 135mm.

5.3       Power Consumption

Maximum Power Consumption: 230W ±10%.