China GoIP-16 VoIP GSM Gateway GSM Device for Free Call

GoIP-16 VoIP GSM Gateway/GSM Device For Free Call
GoIP GSM Gateway 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 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
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: 1300 g (1 lb) (Including AC/DC Adapter)
Size: 220 mm (W) X 380 mm (L) X 85 mm (H) (Width X Length X Heigth
Power: 12 VDC 3A (AC/DC adapter included)

DBL GSM Gateway GoIP-8 Peer to Peer (P2P) Free Global Calling Free VoIP Calls 1. Can be achieved easily without the traditional limitations on IP PBX or fixed public IPS 2. Easily bridge the local calls of two places without paying any long distance charges
DBL Technology is now offering VoIP devices that can support global callings in peer-to-peer (PTP) configuration. It is ideal for anyone who wants to setup a voice connection between two places without the support of an IP PBX / SIP Server. Fixed public IP addresses are no longer required.
Traditionally, a PTP VoIP call occurs when two VoIP phones communicate directly over IP without IP PBXs between them. When the two VoIP phones are not located in the same network, fixed public IPS are required in order to achieve PTP Calling. DBL is now breaking the barrier by offering VoIP devices that support DDNS Peer-to-Peer (PTP) configuration. In addition, DBL offers models with GSM and PSTN terminations to extend IP callings to the GSM and PSTN networks. Global calling can now be achieved easily without the traditional limitations. DDNS P2P Network Topology
As shown in the network topology below, our DDNS PTP VoIP Configuration covers the most common home/office network configurations where only dynamic public IP address are available. Our VoIP devices are installed behind a router (NAT) on each side and the two routers are connected via internet. DBL's dedicated DDNS server is used to resolve the public IP addresses of the networks where the VoIP devices are installed. The two VoIP devices can then call each other directly. DBL also offers a GSM gateway (GoIP) to extend voice calls to PSTN / GSM networks via GSM termination as shown on the left. Ultimately, you can easily bridge the local calls of two places without paying any long distance charges. Special Features
Hot Line
Encryption / VPN for voice security and VoIP Anti-blocking
Models Support DDNS P2P
All models can now support DDNS P2P.
VoIP Phone: EP-636, EP-8201
VoIP ATAs: HT-812P, HT-912, HT-922, HT-842R, HT-882
PSTN Gateway: HT-812P, HT-822P
GSM Gateway: GoIP, GoIP-4, GoIP-8