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首页 > Products > 1-FXS VoIP Gateway (HT-912) (ATA)
1-FXS VoIP Gateway (HT-912) (ATA)
单价 $26.00 / Piece对比
询价 暂无
浏览 299
发货 Chinaguangdongshenzhen
库存 10Piece
过期 长期有效
更新 2020-08-04 09:19
 
详细信息
Product Name: 1-FXS VoIP Gateway (HT-912) (ATA) Model NO.: HT-912 Application: Enterprise Ports: ?4 Firewall: Firewall Transfer Rate: 10/100Mbps Type: Wired VoIP Protocal: H. 323&Sip Transportation: by Air Trademark: DBL Specification: Package: 10PCS/Carton Origin: China Product Description 1. Support ITU H. 323 V4 and IETF SIP V22. Speech quality ensured by QoS3. VLAN and QoS and VoIP gateway4. NAT Transversal and Route functionsThe DBL 2-FXS Port SIP/H. 323 speech quality ensured VoIP Gateway (HT-922) is designed as a compact, high performance, and low cost VoIP Analog Terminal Adapter (FXS Gateway). It offers two FXS ports to interface with traditional analog phones or PBX trunk lines for VoIP communications. It bundles with lots of features to meet the demand in various network environment. It is an ideal low cost VoIP solution for travelers, SOHO and SME users.Key FeaturesOpen Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)Two 10/100 Ethernet for WAN / LAN connectionsPeer-to-Peer IP CallsSpeech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter bufferLine Echo CancellationVLAN and QoS supportNAT Transversal and Router functionsVoice prompts, HTTP Web, Auto Provision support for configuration and updatesHighly stable embedded Linux operating system in high performance ARM 9 ProcessorBasic FeaturesTwo RJ-11 FXS ports for traditional phone set or PBX's trunk lineLEDs for Power, Ready, Status, WAN, PC, FXSCall Forward, Call Hold, Call TransferDial PlanCaller IDEnhanced FeaturesDynamic selection of codecAdvanced jitter bufferAutomatic traversal of NAT and firewallVLAN / QosRouterEcho cancellation for SpeakerphoneComfort noise generation (CNG)Voice activity detection (VAD)Auto provisioning (requires auto provisioning server)On line firmware upgradeMulti-language support: English and ChineseSupported StandardsITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450RFC 1889 - RTP/RTCPRFC 2327 - SDPRFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony SignalsRFC 2976 - SIP INFO MethodRFC 3261 - SIPRFC 3264 - Offer/Answer model with SDPRFC 3515 - SIP REFER MethodRFC 3842 - A Message Summary and Message Waiting IndicatorRFC 3489 - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)RFC 3891 - SIP "Replaces" HeaderRFC 3892 - SIP Referred-By MechanismDraft-ietf-sipping-CC-transfer-04 - Session Initiation Protocol Call Control - TransferCodec: G. 711 (A/µ Law), G. 729A/B, G. 723.1DTMF: RFC 2833, In-band DTMF, SIP INFO