Specifications
1)four channels
2)SIP and H.323
3)NAT transversal and route function
4)Vlan and QoS
5)high quality,good price,
Key Features- Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
- Single or Multiple Server Registrations
- Two 10/100 Ethernet circuits connect to the LAN and an additional device
- GSM module for making GSM calls
- Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
- VLAN and QoS support
- NAT Transversal and Router functions
- Voice prompts, HTTP Web, Auto Provision support for configuration and updates
- Highly stable embedded Linux operating system in high performance ARM 9 Processor
|
Basic Features- LEDs for Power, Ready, Status, WAN, PC, GSM
- Call forward from GSM to VoIP and VoIP to GSM
- Dial in mode or dial out mode only
- Dial Plan
- Password protection for both GSM dial in or dial out
- Retransmit GSM Caller ID to VoIP terminal
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 internal 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)
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.