4 SIM card GSM VOIP gateway

4 SIM card GSM VOIP gateway

original price: 473,65 USD
Shenzhen, China
Production capacity:
1000 Piece / Week
Shenzhen, China
Beginner
86-755-88290301
Suzy Su
Contact person

Basic Information

Place of Origin Guangdong China (Mainland)
Brand Name HyberTone
Model Number GOIP4
GSM VOIP gatewayGoIP4 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 RegistrationsTwo 10/100 Ethernet circuits connect to the LAN and an additional deviceGSM module for making GSM callsSpeech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter bufferVLAN 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 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 onlyDial PlanPassword protection for both GSM dial in or dial outRetransmit GSM Caller ID to VoIP terminalEnhanced Features Dynamic 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 Standards ITU: 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

Delivery terms and packaging

Packaging Detail: 160 mm (W) x 252 mm (L) x 30 mm (H) (Width x Length x Heigth/piece Delivery Detail: 2 days
Port: China

Payment term

Telegraphic transfer

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