- 概述
- 应用场景
- 功能特性
- 产品规格
- 下载
- 相关产品
Cost-effective, High-performance PoE VoIP Phone
To build high-performance VoIP communications at a low cost, PLANET has integrated high-definition voice into a cost-effective SIP phone. It complies with IEEE 802.3af PoE interface for flexible deployment. The VIP-1010PT 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-1010PT 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 VoIP Voice
The VIP-1010PT 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-1010PT supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VIP-1010PT 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-1010PT is a full-featured, enhanced business IP Phone that addresses the communication needs of the enterprises. It provides 1 voice line and dual 10/100Mbps Ethernet. Furthermore, the VIP-1010PT delivers user-friendly design containing a 132x64 graphic LCD with white backlight.
The VIP-1010PT supports all kinds of SIP-based phone features including LDAP, Call Waiting, Auto Answer, Music on Hold, Caller ID 3-way Conferencing, Call on Hold, Call Forwarding, Black List, DTMF Relay, In-Band, Out-of-Band (RFC 2833) and SIP Info, among others. Besides office use, the VIP-1010PT is also the ideal solution for VoIP service offered by Internet Telephony Service Provider (ITSP).
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Enterprise IP Telephony Deployment of VIP-1010PT
The VIP-1010PT is much easier to install and configure than the traditional phone system. Its low cost and high-definition voice quality give you value for money. Base on standard SIP 2.0, it is compatible with all the standard SIP-based servers.
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Highlights- Supports SIP 2.0 (RFC3261)
- Supports 1 SIP voice line
- IEEE 802.3af Power over Ethernet compliant
- Supports HD voice
- LDAP/ TR-069 / SNMP
Phone Features- 1 line (supporting 1 SIP account)
- Supports call waiting, call forwarding, call transfer
- 3-way conferencing
- Call on hold, mute, auto-answer, redial
- Phonebook (500 groups), blacklist (100 groups), call logs (100 entries)
- 5 remote phone book URL supported
- Keypad Lock
- DND (Do Not Disturb)
- Volume adjustable, ring tones selectable
- Call Pickup/Group Call Pickup
- Speed Dial
- Intercom
- Daylight Saving
- Network Packet Capture
- Country Ringtone Signal
- Direct IP Call
- Auto Redial / Hot Desking
- Hotline / XML Browser / Action URL
- Multi-Languages: Default: English and Simple Chinese
IP PPX Feature - HD Voice
- Dial Plan
- SMS, Voicemail, MWI Message Notification
- Wideband Codec: G.722
- Narrowband Codec: PCMA, PCMU, G.729, G.722, G723_53, G23_63,G726_32
- VAD, CNG , Echo Canceller
- Full-Duplex Speakerphone
Security Features- Supports HTTPS (SSL)
- Supports SRTP for Voice Data Encryption
- Supports Login for Administration
- SIP Over TLS
Network Features- SIP V1 (RFC2543), V2 (RFC3261)
- Static IP/DHCP for IP configuration
- 3 DTMF modes: In-Band, RFC2833, SIP INFO
- HTTP/HTTPS Web Server for Management
- NTP for Auto Time Setting
Administration Features- Auto provisioning using FTP/TFTP/HTTP/HTTPS/PnP
- Dial through IP PBX using Phone Number
- Dial through IP PBX using URL Address
- Configuration Managements with Web, Keypad on the phone and Auto Provisioning
- SNMP
- TR069
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硬件特性 |
1线 |
1-line cost-effective IP phone |
显示屏 |
132 x 64 graphic LCD with blue backlight |
功能键 |
4 Soft Keys
10 Programmable Keys
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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)
TCP (RFC 793)
User Datagram Protocol UDP (RFC 768)
Real-time Protocol RTP (RFC 1889, 1890)
Real-time Control Protocol (RTCP) (RFC 1889)
Simple Network Time Protocol (SNTP) (RFC 2030)
Backward compatible with RFC 2543
Session Timer (RFC 4028)
SDP (RFC 2327)
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语音网关 |
SIP version 2 (RFC 3261, 3262, 3263, 3264)
Message Waiting Indicator (RFC 3842)
Voice algorithms:
- PCMA
- PCMU
- G.729
- G.722
- G723_53
- G23_63
- G726_32
Dual-tone Multi-frequency (DTMF), In-Band and Out-of-Band (RFC 2833) (SIP INFO)
Voice Activity Detection (VAD) with Silence Suppression
Comfort Noise Generation
Echo Cancellation Message
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功能特性 |
电话特性 |
1 line (supporting 1 SIP account)
Supports call waiting, call forwarding, call transfer
3-way conferencing
Call on hold, mute, auto-answer, redial
Phonebook (500 groups), blacklist (100 groups), call logs (100 entries)
5 Remote Phone Book URL supported
LDAP
DND (Do Not Disturb)
Volume adjustable, ring tones selectable
Call Pickup/Group Call Pickup
Speed Dial
Intercom
Daylight Saving
Network Packet Capture
Country Ringtone Signal
Direct IP Call
Auto Redial
Hotline
XML Browser
Hot Desking
Keypad Lock
Action URL
Multi-Languages: Default: English and Simple Chinese
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IP PBX 特性 |
HD Voice
Dial Plan
SMS, Voicemail, MWI Message Notification
Wideband Codec: G.722
Narrowband Codec: PCMA, PCMU, G.729, G.722, G723_53, G23_63, G726_32
VAD, CNG, Echo Canceller
Full-Duplex Speakerphone
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安全特性 |
Supports HTTPS (SSL)
Supports SRTP for Voice Data Encryption
Supports Login for Administration
SIP Over TLS
Dial without Register
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网络特性 |
SIP V1 (RFC2543), V2 (RFC3261)
Static IP/DHCP for IP configuration
3 DTMF modes: In-Band, RFC2833, SIP INFO
HTTP/HTTPS Web Server for Management
NTP for Auto Time Setting
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管理功能 |
Auto provisioning using FTP/TFTP/HTTP/HTTPS/PnP
Dial through IP PBX using Phone Number
Dial through IP PBX using URL Address
Configuration Managements with Web, Keypad on the phone and Auto Provisioning
SNMP
TR069
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环境 |
Power Requirements |
5V DC, 1.2A
IEEE 802.3af
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Operating Temperature |
0 ~ 40 degrees C |
Operating Humidity |
10 ~ 65% (non-condensing) |
Weight |
651g (without box) / 920g (with box) |
Dimensions (W x D x H) |
193 x 190 x 35 mm |
Emission |
CE, FCC |
Connectors |
Two 10/100BASE-TX RJ-45 Ethernet ports
Handset: RJ-9 connector
Headphone: RJ-9 connector
DC power jack
Built-in speakerphone and microphone
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技术参数
日期 |
版本 |
描述 |
下载 |
2014-10-31 |
1.0 |
VIP-1010PT |
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安装版本
日期 |
版本 |
描述 |
下载 |
2014-07-29 |
50.141.2.27 |
Initial Release |
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快速安装指南
日期 |
版本 |
描述 |
下载 |
2014-06-09 |
1.0 |
Initial Release |
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用户手册
日期 |
版本 |
描述 |
下载 |
2014-07-09 |
1.0 |
Initial Release |
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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 |
|
VGW-400FO |
4-Port SIP VoIP Gateway (4 FXO) |
|
VGW-400FS |
4-Port SIP VoIP Gateway (4 FXS) |
|
VGW-402 |
4-Port SIP VoIP Gateway (2 FXS + 2 FXO) |
|
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-2020PT |
Enterprise HD PoE IP Phone (2-Line) |
|
VIP-362WT |
802.11n Wireless Desktop IP Phone |
|
VIP-5060PT |
Professional HD PoE IP Phone (6-Line) |
|
VIP-6040PT |
Gigabit Color LCD HD PoE IP Phone (4-Line) |
|