VOIP телефон VP-102 Один канал

VOIP phone VP-102Product OverviewOne Line Phone. Open Standard VoIP Protocols (ITU H. 323 V4 and/or IETF SIP V2). All standard PBX functions . Buttons and keys for all commonly used functions Message waiting indicator. Full duplex speaker phone Highkights Low cost IP Phone Built-in H.323 V4 and SIP V2 Built-in 3-way conference Full duplex speakerphone PC head phone jack Encryption transversal Wall mount and desktop Key Features Open Standard VoIP Protocols (ITU H.323 V4 and/or IETF SIP V2) All standard PBX functions Four call appearances support two simultaneous calls Two 10/100 Ethernet circuits connect to the LAN and an additional device 3-Line LCD (Icons, Alphabets/Numbers, Numbers) Buttons and keys for all commonly used functions Headset port Message waiting indicator Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer Full duplex speaker phone VLAN and QoS support NAT Transversal and router functions Menu, HTTP Web, Auto Provision support for configuration and updates Highly stable embedded Linux operating system in high performance ARM 9 Processor Basic Feature Call forward Call transfer Call hold Mute Redial Display caller ID Display call duration Display date and time Access voice mail Send DTMF tones Message waiting indication (MWI) 100 phone book entries 30 most recent call records for dialled, incoming, and missed calls Adjustment of LCD contrast (4 levels) Adjustment of handset volume (6 levels) Adjustment of speaker phone volume (6 levels) 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), GSM, G.729A/B, G.723.1 DTMF: RFC 2833, In-band DTMF, SIP INFO