Thomson ST2022 DATASHEET

Data Video

Thomson ST2022 - VoIP Business Phone

Voice

CABLE SATELLITE TERRESTRIAL

TELECOM

| The THOMSON ST2022 includes the main

following features:

- 144x32 full graphic display for 3 lines + icons

- Connectivity: Integrated 2 port 10/100 Ethernet switch with
PoE and VLAN tagging

- Phone services:

- 2 lines, Call Forward, Call transfer, Call Hold, Call Waiting

- Handsfree

- Alert (missed calls, Message Waiting Indicator)

- Phone book:
Local (100 entries)
Remote (HTTP XML)

- Caller ID Display

- Automatic Callback

- 3-ways Conferencing

- Downloadable rings

- Multiple power options: power over Ethernet 802.3af and
external power supply adapted (included)

- VoIP Standard: SIP or MGCP

- Voice compression standards: G.711, G.723, G.729ab

- IP addressing: Static or dynamic IP configuration
(integrated (DHCP client)

- Quality of Service: ToS DiffServ, 802.1p/Q, VAD, CNG,
Packet Loss Compensation, Adaptative Jitter Buffer

- Web browser interface for configuration and firmware
(admin & user mode)

- Automatic provisioning system (DHCP, TFTP/HTTP)

- 7 languages

www.thomsontelecompartner.com

| High Quality Solution

The Thomson ST2022 offers a high-quality IP solution at a cost effective price making Voice over IP (VoIP) more accessible for
businesses.
The Thomson ST2022 is a VoIP phone that is compliant with SIP/MGCP and can be used with any SIP compliant PBX, Softswitch
or or IP Centrex solutions of the market. It addresses the need for a cost effective IP solution, helping to make VoIP a viable,
attractive solution for small and large offices environment.

Data Video

CABLE SATELLITE TERRESTRIAL

TELECOM

Thomson
46, quai Alphonse Le Gallo
92100 Boulogne-Billancourt
France
Tel. : [phone]
Fax : [phone]
www.thomsontelecompartner.com

For more information

© 2005 Thomson- Trademark(s) ® Registered\ -Marca(s) Registada(s)\
Photos and features subject to change without notice.
Illustration of product finish may vary from actual color.

| Key Features

• 7 languages (SIP)

• SIP compliant or MGCP

• 144x32 full graphic display

• 2 RJ45 Ethernet switch 10/100

• Handsfree

• Power over Ethernet 802.3af standard

• Navigation keys

• 3 soft keys

| Phone sets

Funtion Support

• Multiple call operation (2 lines)

• Alert (missed calls, message waiting) Indicator

• Mute

• Hold

• Transfer

• Forward

• On-hook dialing

• Handsfree

• Call Log (30 entries; incoming/outgoing calls)

• Local Address Book (100 memories)

• Remote Address Book (HTTP XML)

• Downloadable RTTTL Rings

• Caller ID Display

• Volume control (speaker, handset, headset & ringer)

• Date & time display (idle state)

• Call duration timer

• Diall from Call log or Phonebook

Key Pad

• 3 soft keys

• 2 line keys

• 3 speed dial keys

• 1 MWI/Backlit Voice mail key

• Volume up & down

• Menu, OK and Cancel keys

• Phonebook key

• 2 fixed function keys backlit (mute and handsfree)

LED Indicators

Alert Indicator

Interfaces

2 RJ45 autosensing 10/100Mbps
(one for PC and one for LAN)

| Physical & Environment

• Mounting: Footstand or wallmount

• AC Adapter included

• Power Input: US, EU and UK power supply available

• Operating Temperature: from 0° to 50° C (32° / 104° F)

• Storage Temperature: from -5° to 80° C (-40° / 158° F)

• Humidity: up to 90% non-condensing

| Technical Specifications

SIP Protocol Support

• RFC 3261, SIP: Session Initiation Protocol

• RFC 3264, An Offer/Answer Model with SDP

• RFC 2833, RTP Payload for DMF Digits, Telephony Tones and Telephony Signals

• RFC 3323, A Privacy Mechanism for the Session Initiation Protocol

• RFC 3311, The Session Initiation Protocol UPDATE Method

• RFC 3725, Best Current Practices for Third Party Call Control
in the Session Initiation Protocol

• RFC 3515, The Session Initiation Protocol (SIP) Refer Method

• RFC 3265, Session Initiation Protocol (SIP) - Specific Event Notification

• RFC 2976, The SIP INFO Method

• RFC 3262, Reliability of Provisional Responses in the Session Initiation Protocol (SIP)

• RFC 3725, Best Current Practices for Third Party Call Control (3pcc) in the Session
Initiation Protocol (SIP)

• RFC 3891, The Session Initiation Protocol (SIP) ‘Replaces’ Header

• RFC 2327, SDP: Session Description Protocol

• RFC4235, An INVITE-Initiated Dialog Event Package for SIP.

MGCP Protocol

• RFC 3435, Media Gateway Control Protocol

• RFC 3149, MGCP Business Phone Packages

• RFC 2327, SDP: Session Description Protocol

• RFC 3264, An Offer/Answer Model with SDP

• RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

Audio Code

• G.711 -law/A-law

• G.723.1

• G729ab

Dial Tone Signal Generation

• DTMF (out of band, in Band and INFO in SIP)

• RFC 2833

Internet Support

• IP (RFC 0791), TCP (RFC 0793), UDP (RFC 0768), ARP (RFC 0826) protocols

• DHCP client (RFC 2131)

• SDP

• TFTP

• HTTP

• DNS

• SNTP

Quality of Service

• ToS Diffserv, 802.1p/Q

• VAD, CNG, Packet loss compensation, Adaptative Jitter Buffer

VLAN tagging

• IEEE 802.1Q

Ethernet

• IEEE 802.3 10BASE-T Ethernet

• IEEE 802.3u 100BASE-TX Fast Ethernet

| Configuration

IP Number Assignment

• DHCP client or fixed IP

Configuration Support

• Keypad & LCD

• Web browser management with 2 Levels (User and Admin)

• TFTP/HTTP server download- Local & remote warm reboot

• Password protection for configuration

TECHNICAL SUMMARY

Voice