Gold Member Since 2010
Audited Supplier
Shenzhen DBL Technology Limited

1channels Gsm Voip Gateway Gsm Gateway, Gsm Gateway Self Contained Voip Gateways, 1 Port Gsm Gateway manufacturer / supplier in China, offering 1 Channel GSM Gateway With 1 SIM Card Goip VoIP, DBL 1/2/4/8-FXS VoIP Gateway (HT-882), 8 Port/Channel GSM VoIP Gateway with SIP&H. 323 (GoIP8) and so on.

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Supplier Homepage Product VoIP GSM Gateway 1 Channel GSM Gateway With 1 SIM Card Goip VoIP

1 Channel GSM Gateway With 1 SIM Card Goip VoIP

FOB Price: US $47 / Piece
Min. Order: 1 Piece
Min. Order FOB Price
1 Piece US $47/ Piece
Get Latest Price
Production Capacity: 5000PCS/Month
Transport Package: 36(L)Cm*21(W)Cm*7(H)Cm
Payment Terms: T/T, Western Union, Paypal, Money Gram
Basic Info
  • Model NO.: GoIP1
  • Protocol: SIP&H.323
  • Number of Channels: 1
  • Color: Gray
  • Power: 12 Vdc 2a (AC/DC Adapter Included)
  • Trademark: DBL
  • Origin: China
  • Type: Gateway
  • Support: IMEI
  • Wire or Not: Wire
  • Weight: 450g (1 Lb) (Including AC/DC Adapter)
  • Warranty: One Year
  • Specification: 26(L)Cm*16(W)Cm*3(H)Cm
  • HS Code: 8517709000
Product Description
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
 
 
 
 
 
Key Features
Open Standard VoIP Protocols (SIP& H. 323)
Single or Multiple Server Registrations
Two 10/100 Ethernet for WAN / LAN connections
Peer-to-Peer IP Calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
Line Echo Cancellation
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, FXS
Dial in mode or dial out mode only
Call forward from GSM to VoIP and VoIP to GSM
Dial Plan
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
Operating temperature: 10° C to 40° C (50° F to 104° F)
 
 
Physical and Environmental
 
Storage temperature: 0° C to 50° C (32° F to 122° F)
Power: 12 VDC 4.5A (110V-220V) (AC/DC adapter included)
Warranty: one year
 
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