GoIP-16 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