Gsm Terminal Gsm Fixed Wireless Terminal Fixed Wireless Terminal

The GOIP GSM Phoneis a 1 SIM Card Broadband Phone Gateway that had been developed by DBL Ltd. 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 FeaturesOpen 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 FeaturesLEDs 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 FeaturesDynamic 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 StandardsITU: 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