Specifications

Roip302, Cross-Network Gateway
1.SIP/TCP protocol
2.3 PTT ports
3.Radio,Voip,GSM,Public Announce
4.high quality, good price

  The Roip302, Cross-Network is designed to convert the audio and PTT signals into a radio terminal into IP packets and then transmit the data via the IP networks. The challenge in this technology is to insure that the audio is transmitted in real time and the PPT control signal is transmitted immediately and reliably. The radio range is general limited by the restricted transmitting power, the antenna sensitivity, and other environmental factors.
 

Key Features
Open Standard VoIP Protocols (IETF SIP V2)
Two 10/100 Ethernet for WAN / LAN connections
Supply dynamic DNS service
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
Line Echo Cancellation
VLAN and QoS support
NAT Transversal and Router functions
Support IP voice stream to via a recording server
Highly stable embedded Linux operating system in high performance ARM 9 Processor
 
Basic Features
Two RJ-11 FXS port for traditional phone set or PBX's trunk line
Support three PTT port
LEDs for Power, Ready, Status, WAN, PC, FXS
Call Forward, Call Hold, Call Transfer
Dial Plan
Caller ID
 

Enhanced Features

Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese
 

 

Supported Standards

ITU: H.323 V4, H.225, H.235, H.245, H.450
RFC 1889 - RTP/RTCP
RFC 2327 – SDP
RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 – SIP INFO Method
RFC 3261 – SIP
RFC 3264 – Offer/Answer model with SDP
RFC 3515 – SIP REFER Method
RFC 3842 – A Message Summary and Message Waiting Indicator
RFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 – SIP “Replaces” Header
RFC 3892 – SIP Referred-By Mechanism
draft-ietf-sipping-cc-transfer-04 –  Session Initiation Protocol Call Control - Transfer
Codec: G.711 (A/µ law), G.729A/B, G.723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO 

 

Application:
Saving Cost on International and Long Distance Calls,when you are in the other countries& you have this gateway at your Local office,you could dial & receive phone call from your customers as you are there without Long Distance fee.
Very excellent solution for international company with branch offices oversea.

 

SKYLINE International Communication Co., ltd is a high-tech enterprise integrated with scientific research, development, production and sales. The company has been engaged in the research, development and production of communication products since its establishment. Our products are listed as follows.

FXS series:HT-912(one fxs port),HT-922(two fxs ports),HT-842R(four fxs ports).HT-882(eight fxs ports)

GoIP series:Goip-1,Goip-4,Goip-8

GSM Modem:Q2303-8/16/32/64, Q2403-8/16/32/64,Q2406-8/16/32/64,Q24plus-8/16/32/64(8/16/32/64 is the sim card slots number.)

 RoIP series: RoIP302, RoIP302M, RoIP102

 IP Phone: EP-636 (one or two lines), EP-8201 (four lines)

 Recorder Box:TYH8200 (one line), TYH8201 (two lines), TYH636 (eight lines)