- 概述
- 应用场景
- 功能特性
- 产品规格
- 下载
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Cost-effective, High-performance PoE VoIP Phone
To build high-performance VoIP communications at a low cost, PLANET now introduces the latest member of its IP Phone family, the VIP-2020PT enterprise-class 2-Line PoE IP Phone. It complies with IEEE 802.3af PoE interface for flexible deployment. The VIP-2020PT makes it simple for the enterprise featuring voice and data system or expanding voice system to new locations. It helps the company to save money on long distance calls; for example, the remote workers can dial in through a Unified VoIP Communication System just like an extension call but no long distance call charge would occur. The VIP-2020PT also allows call to be transferred to anyone at any location within the voice system, which enables the enterprise to communicate more effectively and is helpful to streamline business processes.
| | | | High Quality HD Voice over IP
The VIP-2020PT delivers HD voice (High-Definition Voice) which is the next generation of voice quality for telephony audio, making the quality of voice better than that (toll quality) of the standard digital telephony and even close to that of a room conversation. HD voice is transmitted in the audio frequency range of 50 Hz to 7 kHz or higher over telephone lines, resulting in higher quality voice and clearer communication. | | Standard Compliance
The VIP-2020PT supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VIP-2020PT is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better multi-media exchange services. | | | | Enhanced, Full-Featured Business IP Phone
The VIP-2020PT is a full-featured enhanced business IP Phone that addresses the communication needs of the enterprises. It provides 2 voice lines and dual 10/100Mbps Ethernet. Furthermore, the VIP-2020PT delivers user-friendly design containing a 12848 Graphic LCD with white backlight, 2 Line keys and 4 soft keys. It supports 5 extension consoles with each consisting of 26 keys.
The VIP-2020PT supports all kinds of SIP based phone features including Call Waiting, Auto Answer, Music on Hold, Caller ID and Call Waiting ID, 3-way Conferencing, Call Hold, Call Forwarding, Black List, DTMF Relay, In-Band, Out-of-Band (RFC 2833) and SIP INFO, among others. Besides office use, the VIP-2020PT is also the ideal solution for VoIP service offered by Internet Telephony Service Provider (ITSP). | | | | Secure, High-Quality VoIP Communication
The VIP-2020PT can effortlessly deliver secured toll voice quality by utilizing cutting-edge 802.1p QoS (Quality of Service), 802.1Q VLAN tagging, and IP TOS (Type of Service) technology. Using voice and data VLAN can easily separate the data and voice, thus maintaining the best quality. | | | | Professional Application
The VIP-2020PT supports Busy Lamp Field (BLF) function that, via the lights on the phone, enables users to easily identify the status of other phones which are connected to the same IP PBX, such as busy, idle, ringing, etc. The connected IP PBX must also support BLF feature. The BLF function is helpful for a receptionist on the front desk to route all incoming calls smoothly. | | | |
| | Enterprise IP PBX Deployment of VIP-2020PT
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Highlights- Supports SIP 2.0 (RFC3261)
- Supports IAX2 and IAX2 line call
- Supports two SIP voice lines
- IEEE 802.3af Power over Ethernet compliance
- Supports multiple road call waiting in line
- Supports HD voice
- Supports SRTP and Busy Lamp Field (BLF)
- Supports 5 extension consoles; max. 130 definable keys
Advanced Features- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer / IP call
- Inband, SIP info, RFC 2833 DTMF Relay
- 9 kinds of ring types and 3 user-defined music rings
- Large dot matrix LCD display and soft keys make user easier to use
- Soft keys and function keys programmable
- Multilanguage realizes localization
- Echo cancellation: Supports G.168, and hands-free can support 96ms
- Support Voice Gain Setting, VAD, CNG
- Full duplex hands-free speaker phone
- Hands-free headset ringing choice
- Voice codec setting for each SIP line
SIP Applications- Call forward / Transfer (blind/attended)
- Call Holding / Waiting
- 3-way conference
- Paging and Intercom
- Call park / Call pickup / Join call
- Redial and click to dial
- Secondary dialing automatically
- Incoming calls / outgoing calls / missing calls (Each supports 100 records)
- SMS and Speed Dial
- Phonebook up to 500 records
- XML phonebook / browser
Call Control Features- Flexible dial map / Hotline / Empty calling no
- Reject service / Black list for reject authenticated call
- White list / Limit cal
- Do not disturb (DND)
- Caller ID / CLIR (reject the anonymous call) / CLIP (make a call with anonymous)
- Dial without register
Network Features- Route and Bridge modes
- PPPoE / DHCP client on WAN / LAN
- 802.1 VLAN (voice VLAN / data VLAN)
- VPN (L2TP) and DMZ
- Basic NAT and NAPT
- Main DNS and secondary DNS server
- DNS Relay, SNTP Client, Firewall, openVPN
Maintenance and Management- Integrated web server provides web-based administration and configuration
- Telephone keypad configuration via display menu/navigation
- Automated provisioning and upgrade via HTTPS, HTTP, TFTP
- User Authentication for configuration pages
- Local and Remote Syslog (RFC 3164)
- SNTP Time Synchronization
- TR069
| |
硬件特性 |
2线 |
2-Line enterprise-class IP phone |
显示屏 |
128*48 Graphic LCD with blue backlight |
功能键 |
2 line keys include in 4 DSS keys
4 Soft Keys
12 dialing buttons (0~9, *, #)
12 fixed function buttons
|
协议与标准 |
数据网络技术 |
MAC Address (IEEE 802.3)
IPv4 (RFC 791)
Address Resolution Protocol (ARP)
DNS: A record (RFC 1706), SRV record (RFC 2782)
Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
Internet Control Message Protocol (ICMP) (RFC 792)
TCP (RFC 793)
User Datagram Protocol UDP (RFC 768)
Real Time Protocol RTP (RFC 1889, 1890)
Real Time Control Protocol (RTCP) (RFC 1889)
Differentiated Services (DiffServ) (RFC 2475)
Type of service (ToS) (RFC 791, 1349)
VLAN tagging 802.1p Layer 2 quality of service (QoS)
Simple Network Time Protocol (SNTP) (RFC 2030)
Backward compatible with RFC 2543
Session Timer (RFC 4028)
SDP (RFC 2327)
NAPTR for SIP URI Lookup (RFC 2915)
|
语音网关 |
SIP version 2 (RFC 3261, 3262, 3263, 3264)
SIP supported in NAT networks [including STUN (RFC 3489)]
Message Waiting Indicator (RFC 3842)
Voice algorithms:
- G.711 (A-law and ?-law)
- G.723.1 high/low
- G.729a/b
- G.722.1 (HD Voice)
- G.726
Dual-Tone Multi-Frequency (DTMF), In-Band and Out-of-Band (RFC 2833) (SIP INFO)
Voice Activity Detection (VAD) with Silence Suppression
Adaptive Jitter Buffer Management
Comfort Noise Generation
Echo Cancellation Message |
功能特性 |
高级特性 |
SIP 2.0 (RFC3261) / IAX2
IEEE 802.3af Power over Ethernet (PoE) compliant
Multiple road call waiting in line
Supports HD voice
Supports SRTP and BLF
SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer / IP call
Inband, SIP info, RFC 2833 DTMF Relay
9 kinds of ring types and 3 user-defined music rings
Large dot matrix LCD display and soft keys make user easier to use
Supports 5 extension consoles with each consisting of 26 keys
Soft keys programmable; function keys programmable
Multilanguage realizes localization
Echo cancellation: Support G.168, and Hands-free can support 96ms
Full duplex hands-free speaker phone
Hands-free headset ringing choice
Support Voice Gain Setting, VAD, CNG
Voice codec setting for each SIP line
|
SIP 应用 |
Call forward / Transfer (blind/attended)
Call Holding / Waiting
3-way conference
Paging and Intercom
Call park / Call pickup / Join call
Redial and click to dial
Secondary dialing automatically
Incoming calls / outgoing calls / missing calls (Each supports 100 records)
Phonebook for 500 records
SMS and Speed Dial
XML phonebook/browser
|
呼叫控制功能 |
Flexible dial map / Hotline / Empty calling no.
Reject service / Black list for reject authenticated call
White list / Limit cal
Do not disturb (DND)
Caller ID / CLIR (reject the anonymous call) / CLIP (make a call with anonymous)
Dial without register
|
网络功能 |
Route and Bridge modes.
PPPoE / DHCP client on WAN / LAN
802.1 VLAN (voice VLAN / data VLAN)
VPN (L2TP) and DMZ
Basic NAT and NAPT
Main DNS and secondary DNS server
DNS Relay, SNTP Client, Firewall, openVPN
|
管理 |
Integrated web server provides web-based administration and configuration
Telephone keypad configuration via display menu/navigation
Automated provisioning and upgrade via HTTPS, HTTP, TFTP
User Authentication for configuration pages
Local and Remote Syslog (RFC 3164)
SNTP Time Synchronization
TR069
|
环境 |
电源要求 |
5V DC, 1A
IEEE 802.3af
|
工作温度 |
0 ~ 40 degrees C |
工作湿度 |
10 ~ 65% (non-condensing) |
重量 |
950 g |
尺寸 (W x D x H) |
290 x 260 x 60 mm |
认证 |
CE, FCC, RoHS |
接口 |
Two 10/100BASE-T RJ-45 Ethernet ports
Handset: RJ-9 connector
Headset: RJ-9 connector
RJ-11 EXT connector
DC power jack
Built-in speakerphone and microphone
|
技术参数
日期 |
版本 |
描述 |
下载 |
2013-11-15 |
1.0 |
VIP-2020PT |
|
安装版本
快速安装指南
日期 |
版本 |
描述 |
下载 |
2013-08-29 |
1.0 |
Initial Release |
|
用户手册
日期 |
版本 |
描述 |
下载 |
2013-08-26 |
1.0 |
Initial Release |
| |
图片 |
模块 |
描述 |
|
IPX-2100 |
网络电话PBX系统(100用户) |
|
IPX-330 |
Internet Telephony PBX System |
|
UMG-1000 |
Desktop Unified Office Gateway |
|
UMG-2200 |
Unified Office Gateway |
|
VIP-156 |
SIP Analog Telephone Adapter |
|
VIP-156PE |
802.3af PoE SIP Analog Telephone Adapter |
|
VIP-157 |
1 FXS / 1 FXO SIP Analog Telephone Adapter |
|
VIP-157S |
2 FXS Analog Telephone Adapter |
|
VIP-1680 |
16-Port H.323 / SIP VoIP Gateway |
|
VIP-2480 |
24-Port H.323 / SIP VoIP Gateway |
|
VIP-281 |
2-Port H.323 / SIP VoIP Gateway |
|
VIP-362WT |
802.11n Wireless Desktop IP Phone |
|
VIP-5060PT |
Professional HD PoE IP Phone (6-Line) |
|
VIP-880 |
8-Port H.323 / SIP VoIP Gateway |
|