Voip Gateway

Voip gateway

Product characteristics:
1. Standard sip 2. 0 terminal product, powerful applicability, and best for global popularizing; support sip 2. 0, tcp / udp / ip, rtp / rtcp, icmp, arp / rarp, ntp, pppoe, tftp, http, etc; support g. 711 (alaw and u-law) , g. 723. 1 (5. 3k / 6. 3k) , g. 723 (40k / 32k / 24k / 16k) , g. 728, g. 729a / b, ilbc;
2. The powerful dsp digital audio processor to keep a wonderful audio-quality; inside dns, router, nat, dmz, the gateway and dhcp (the client end and the server end) ; jitter control adn packet loss concealment technology;
3. Support incoming call on show, restricting, and holding, disconnection, call transfer, call divert, dialing project, etc. Support conference call;
4. Supports vad, cng, agc, the echo eliminates (g. 168) and the static sound suppression;
5. Support layer 2 (802. 1q vlan, 802. 1p) and layer 3 (qos, , tos diffserv, mpls)
6. Powerful long-distance tftp or http manages; to dispose through the browser or the voice hint;
7. Nat automatic penetration;
8. Support passing through and t. 38 fax;
9. Good for the environment, according to the international electromagnetism radicalizations standard;
10. Select high quality material, original electrocircuit design, and no electric current noise, work steadily;
11. Porcelain and legerity design, let you feel the full-bodied appetency of the product besides expediency schlepping.
Ata (sip2. 0) yes
Router / bridge / dhcp yes
Ethernet port 1wan / 1lan (rj45)
Fxs 1fxs
Fxo pstn pass-through
Exit yes
Weight 0. 38kg
Active status temperature 0~40
Power source adapter ac in: 100v-240v dc out: + 9v / 600ma
Use power: <2. 5w
Authentication fcc / ce

Voip gateway


Ieee 802.3 10 base tsupport dhcp, pppoe dns telnet
Support g.711a/ug.723.1, g.729 /a/b/ab and gsm610 voice codec
Sip, h.323 v4 protocalsip rfc3261support http, https, ftp upgrade
Tcp/ip: internet protocalsupport rtp rtcp protocal
Support vad/cngone wan, one lan, support router
Call forward (always, busy, no answer), call waiting and per call-waiting blocking
Caller number/name display caller id /block outgoing caller-id/reject anonymous call
A voice activity detection (vad) facsimile protocol(option): t.30/t.38
Voip speed dial, voip digital mapper area ring tone supportdnd(do not disturb) mwi
G configurable/adaptive jitter buffer sizesupport vlan protocalsupport web, ivr to configur

VoIP gateway
SIP supported FXS/FXO Gateway

Voip Gateway IAD series to meet the NGN requirement of Telecom carrier, Internet Service Provider, SMB/SME users and home users.It is able to provide safe and high-efficiency voice services, with all kinds of protocols and functions.

The ht-322 is designed as a compact, high performance, and low cost fxo gateway. The fxo detection is optimized to avoid the hold up of the pstn line when the other party is disconnected. This has been one of the key issues in the design of fxo gateway. The incoming pstn caller id is also transmitted to the voip user for more user friendly operation. The ht-322 is a full featured fxo gateway and is designed for easy installation and configuration. It is an ideal solution for voip to pstn termination in both sme and soho environment.

The ht-912 is designed as a compact, high performance, and low cost voip analog terminal adapter (fxs gateway). It comes with one fxs port to interface with a traditional analog phone set or a pbx trunk line for voip communications. It bundles with lots of features to meet the demand in various network environments. It is an ideal low cost voip solution for travelers and soho users.

The ht-812p is designed as a compact, high performance, and low cost voip analog terminal adapter (fxs gateway). It comes with one fxs port to interface with a traditional analog phone set or a pbx trunk line for voip communications. By connecting a pstn line to the bypass port, the phone set connected to the fxs port can also access the pstn line for traditional telephone service. The ht-812p is a full featured fxs gateway and is designed for easy installation and configuration. It is an ideal low cost solution for travelers and soho users.

8 GoIP ,GSM VoIP Gateway:
The GSM Fixed Wireless Terminal 8 ports Gateway GoIP-8 is a 8 SIM Card Broadband Phone Gateway that had been developed by CHINASKYLINE Ltd. GoIP_8 SIM Card Broadband Phone Gateway is a new product that connect the GSM and the VOIP seamlessly. To GoIP_8 what is installed on the Mobile SIM Card, you can register the GSM telephone on the VoIP soft switch. SIP and H.323 agreement are built in the GoIP_8 and configured flexible. Caller ID can be seen by using SIP. Flexible routing can meet the need of all kinds of call forwarding; even more special is that GoIP_8 support multi-device group, it can be easily combined into arbitrary number of channels of Large Gateway Group.

Key Features:
Multiple GoIP8 grouping mode
Provide one cellular channels for IP-PBX
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
Hardware Specifications
Processor: ARM9E 133MHz
DSP: VPDSP101 196MHz
Memory: RAM 16MB/ Flash 4MB
GSM Module: Type: 850MHz, 900MHz, 1800MHz, 1900MHz
Power: Input AC100V ~ 240V, output DC12V/2A +-10%
Power consumption: 32W maximum
Network card: 100/10Base-T x2
LED: Operation and lines light
GSM Passway:eight
Operating temperature: 10C to 40C (32F to 104F)
Storage temperature: 0C to 50C (32F to 122F)
Working Humidity: 40% ~ 90% Not congealed
Weight: 1203 g (1 lb) (Including AC/DC Adapter)
Warranty: one year.


Size: 38*17*7 Weight¡êo1.5Kg

4-channel gsm voip gateway,Goip-4
The 4-channel GSM VoIP Gateway/GoIP-4 bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.

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 -AddressTranslators (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 Law), G. 729A/B, G. 723.1
-DTMF: RFC 2833, In-band DTMF, SIP INFO

Physical and Environmental
-Operating temperature: 10 C to 40 C (50 F to 104 F)
-Storage temperature: 0 C to 50 C (32 F to 122 F)
-Weight: 500 g (1 lb) (Including AC/DC Adapter)
-Size: 105 mm (W) x 100 mm (L) x 28 mm (H) (Width x Length x Heigth
-Power: 12 Vdc 500 mA (AC/DC adapter included).



Siae:36*21*7 Weight:1KG.