VoIP SIP ATA ( Analog Telephony Adapter) Series

VA01 VoIP SIP ATA with 1 FXS port
VA02 VoIP SIP ATA with 2 FXS ports
VA11 VoIP SIP ATA with 1 FXO, 1 FXS ports

 VoiP_ATA 

Features

  • Compliant with IEEE 802.3, 802.3u 10/100M Ethernet standard

  • Supports SIP 3261 (Radvision SIP stack), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP, DNS, and DHCP

  • Interoperable with various third party SIP end user devices

  • Supports popular voice-coders including G.729, and G.711 (A-law and μ-law)

  • Supports standard voice features such as Caller ID, Consultation Hold, Call Waiting, Blind Call Transfer, Attended Call Transfer, Call Forwarding, Mid-Call DTMF (only using RTP) and 3-Way Conferencing

  • Supports Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), and Line Echo Cancellation

  • Supports DIGEST authentication (MD5 only)

  • Provides easy configuration through a web interface and limited configuration through the handset dial-pad

  • Voice over data prioritization ensure high-quality telephone service

  • Supports Local PSTN call transfer to VoIP call or VoIP call transfer to local PSTN call

  • Selective PSTN / VoIP voice communications

  • PSTN Backup support through Hardware Relay—applicable when power is down or when WAN interface is down (Optional)

  • STUN support for NAT traversal

Specifications

Standard

  • IEEE 802.3 / 802.3u 10/100M Ethernet Standard

  • SIP v1(RFC2543), v2(RFC 3261)

  • Voice codec : G.711(A-law, μ-law), G.729A/B

Physical Interface

  • 10/100Mbps Ethernet, RJ45

  • RJ11 Port for Phone

  • RJ11 Port for PSTN (Optional)

  • Power DC 9V, 1A

Voice Feature

  • Voice codec
    G.711 : 64k bit/s (PCM)
    G.723 : 6.3k / 5.3k bit/s
    G.726 : 16k / 24k / 32k / 40k bit/s (ADPCM)
    G.729A : 8k bit/s (CS-ACELP)
    G.729B : adds AD & CNG to G.729

  • Supports silence Suppression, VAD( Voice activity detection), CNG ( Comfortable noise generator) and LEC (Line echo canceller)

  • Supports Packet Loss Compensation

Call Function

  • Call ID

  • Call Waiting Caller ID

  • Consultation Hold

  • Blink Transfer

  • Attended Transfer

  • 3-Way Conferencing

  • Call Waiting

  • Call Forwarding

  • Call Return

  • IP Dialing

  • Speed Dialing

Network Function

  • PPPoE

  • DHCP Client

  • Friendly web-based configuration

PSTN Backup (Optional)

  • Supports PSTN Backup

Management Feature

  • Web Browser

  • Telnet

  • Voice configuration

  • HTTP

Security

  • HTTP 1.1 basic/digest authentication for Web setup

  • MD5 for SIP authentication (RFC2069/RFC2617)

Led indicator

  • Phone, Line, Ethernet and power indicator

Temperature

  • 32~122°F (0 ~50°C)

Humidity

  • Max. 95% (Non-Condensing)

Regulatory Approvals

  • FCC, CE

Application

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