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GSM Gateway
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  • Overview
  • Specification
  • Inquire
  • GSM VoIP 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 cell phone roaming via a VoIP network. Significant savings on long distance charges can be realized.


    Physical and Environmental

    Weight: 500 g (1 lb) (Including AC/DC Adapter)
    Size: 105 mm (W)×100 mm (L)×28 mm (H)

    Power: 12V DC 500 mA (AC/DC adapter included) 


    Basic Features

    1. LEDs for Power, Ready, Status, WAN, PC, GSM

    2. Call forward from GSM to VoIP and VoIP to GSM

    3. Dial in mode or dial out mode only;

    4. Dial Plan

    5. Password protection for both GSM dial in or dial out

    6. Retransmit GSM Caller ID to VoIP terminal


    Key Features

    1. Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
    2. Single or Multiple Server Registrations
    3. Two 10/100 Ethernet circuits connect to the LAN and an additional device
    4. GSM module for making GSM calls
    5. Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
    6. VLAN and QoS support
    7. NAT Transversal and Router functions
    8. Voice prompts, HTTP Web, Auto Provision support for configuration and updates
    9. Highly stable embedded Linux operating system in high performance ARM 9 Processor
  • Supported Standards

    1. ITU: H.323 V4, H.225, H.235, H.245, H.450
    2. RFC 1889 - RTP/RTCP;
    3. RFC 2327 SDP
    4. RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
    5. RFC 2976 SIP INFO Method;
    6. RFC 3261 SIP
    7. RFC 3264 Offer/Answer model with SDP
    8. RFC 3515 SIP REFER Method
    9. RFC 3842 A Message Summary and Message Waiting Indicator
    10. RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
    11. RFC 3891 SIP Replaces Header
    12. RFC 3892 SIP Referred-By Mechanism
    13. Draft-ietf-sipping-cc-transfer-04 Session Initiation Protocol Call Control - Transfer
    14. Codec: G.711 (A/µ law), G.729A/B, G.723.1
    15. DTMF: RFC 2833, In-band DTMF, SIP INFO


    Enhanced Features

    1. Dynamic selection of codec; 2. Advanced jitter buffer

    3. Automatic traversal of NAT and firewall

    4. VLAN / Qos;

    5. Router

    6. Echo cancellation for Speakerphone

    7. Comfort noise generation (CNG)

    8. Voice activity detection (VAD)

    9. Auto provisioning (requires auto provisioning server)

    10. Online firmware upgrade

    11. Multi-language support: English and Chinese

  • Please fill in the form below and we shall contact you as soon as possible
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    Inquiry for  GSM Gateway PEG901
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