RoIP 102,with sip server for voice communicatio between voip,radio and gsm network,Cross network roip gateway

Specifications

Overview
The fundamental of RoIP(Radio over IP)technology is to convert the audio and PTT signals in 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
Support a 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
 

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)

 FXS+PSTN series:HT-812P(one fxs port with a PSTN live line),HT-822P(two fxs ports with a PSTN live line),HT-522(two fxs ports with two PSTN live lines),HT-544(four fxs ports with four PSTN live lines)

 FXO series:HT-322(two fxo ports),HT-342(four fxo ports)

 FXO+FXS series:HT-112(one fxo port and one fxs port),HT-222(two fxo ports and two fxs ports),HT-442(four fxo ports and four fxs ports)

 GSM&CDMA series:GoIP 1 (one channel), GoIP 4 (four channels), GoIP 8 (eight channels), CoIP 4 (four channels)

 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)

If you have any doubt or want to know further information,please feel free to contact me.