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VoIP GSM GATEWAY GoIP-1
GoIP 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 (IETF SIP V2)
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 2A (110V-220V) (AC/DC adapter included)
Warranty: One year
GoIP 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 (IETF SIP V2)
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 2A (110V-220V) (AC/DC adapter included)
Warranty: One year
| Hardware characteristics | parameter |
| Model | GoIP-1 |
| Processer | ARM9 133MHz |
| DSP | VP-101-1 196MHz |
| RAM | 16M |
| FLASH | 4M |
| Power Supply | DC12V/2A +-10% |
| Frequency | default 900M/1800M |
| Choice: 850M/1900M | |
| Power Consumption | Max 5W |
| LED | Operating and circuit light |
| Network card | 100/10BASE-T ×2 |
| weight | 0.40KG |
| Operating temperature | 0-45°C |
| Working humidity | 40%-90% non condensation |
| color | grey |
| GSM channel | 1 |
| warranty | 1 year |
| Speech Characteristics | Service Condition | Remark |
| G.168 Echo cancellation | support | 16ms |
| g.723 | Support | |
| g.711A/u | Support | |
| g.729A/B | Support | |
| PLC | Support | |
| CNG | Support | |
| VAD | Support | |
| Jitter buffer | support |
| Network Characteristics | Parameter | Remark |
| LAN Port | ||
| DHCP | support | |
| PPPoE | support | |
| Static IP | support | |
| Network Time | Relay or Port forwarding | Relay need coordinate with DBL Relay Server |
| 10/100BASE-T | 10 or 100 auto adaptation | |
| PC Port | ||
| Static IP | support | |
| DHCP | Max support 200 terminals | |
| 10/100 BASE-T | auto adaptation | |
| Switch mode | support | |