CDMA- VoIP-Gateway COIP

CDMA VOIP GATEWAYKey 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 CDMA module for making CDMA 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 ProcessorBasic 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 TelephonySignals 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 AddressTranslators (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