Gold Member Since 2010
Audited Supplier
Shenzhen DBL Technology Limited

Voip Gsm Gateway, Avaya Voip Goip, Voip Gsm Terminal manufacturer / supplier in China, offering 16 Ports GSM Gateway GoIP16, 1 Channel GSM VoIP Gateway/GoIP 1/GoIP GSM, 8 Channels GSM VoIP Gateway/GoIP 8 and so on.

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Supplier Homepage Product VoIP GSM Gateway 16 Ports GSM Gateway GoIP16

16 Ports GSM Gateway GoIP16

FOB Price: US $405 / Piece
Min. Order: 1 Piece
Min. Order FOB Price
1 Piece US $405/ Piece
Get Latest Price
Production Capacity: 5, 000PCS/Month
Payment Terms: T/T, Western Union, Paypal
Basic Info
  • Model NO.: GoIP-16
  • Type: Wired
  • Information content: Voice
  • Condition: New
  • Operating Temperature: 10° C to 40° C (50° F to 104° F)
  • Size: 105 Mm (W) X 100 Mm (L) X 28 Mm (H) (Width X Lengt
  • Trademark: DBL
  • Usage: Telephone, Workstation, Server, Mobile phone
  • Support Network: GSM
  • Certification: CE, ISO, RoHS, GS
  • Support: H. 323&SIP Vlan & Qos
  • Storage Temperature: 0° C to 50° C (32° F to 122° F)
  • Warranty: One Year
  • Origin: Guangdong, China
Product Description
Single port VoIP 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 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
GSM module for making GSM 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 Processor

Basic 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 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
1. H. 323& SIP 2.0
2. Support IMEI change
3. Have encryption VPN and Relay
4. frequency: 850/900/1800/1900M
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