The 8-Channel GSM VoIP Gateway (GoIP_8)is a 8 SIM Card Broadband Phone Gateway that had been developed by DBL 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 Softswitch.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 - 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
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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
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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 Chines
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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
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