Item: Voip GSM gateway
Model No.: GOIP-8
Bands: GSM 850/900/1800/1900mhz Supports H. 323 V4 & SIP V2 Supports Sim Bank Supports IMEI change The GoIP-8 is 8 Channels GSM Gateway bridges the GSM and the IP networks by enabling voice communications. It is an ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via a VoIP network. It enable you can save distance charges significantly.
Key Features: -GOIP-8 support Open Standard VoIP Protocols H. 323 V4 and SIP V2 -It has 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 -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 -It support VLAN and QoS -Single or
Multiple Server Registrations Basic Features: -LEDs for Power, Ready, Status, WAN, PC, GSM -Support 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 - 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