A selection of terminals support many types of conferences
PJP-50R has three built-in terminals. The LAN port provides high sound quality conferencing between Projectphones. The LINE port (for analog phone lines) facilitates calls with land line phones, cell phones, and analog phone terminals. And to connect to ...
A selection of terminals support many types of conferences
PJP-50R has three built-in terminals. The LAN port provides high sound quality conferencing between Projectphones. The LINE port (for analog phone lines) facilitates calls with land line phones, cell phones, and analog phone terminals. And to connect to a PC as a microphone speaker for Web conferencing or "soft phone" use, you can use the AUDIO (IN/OUT) port. PJP-50R offers a multiinterface voice conference system that can be used in various ways according to the application.
Integrated audio mixer enables conferencing with mixed line types
With its various connectors, PJP-50R can open calls with simultaneous IP, analog phone, and audio connections. Parties away from the office can join IP audio conferences from their cell phones, or join Web conferences from a telephone (voice only).
* Participant can talk with location A by cell phone through PJP-50R at location B
Realism that only an IP connection can offer
IP connections offer 7 kHz of voice bandwidth, or approximately twice that of analog phone lines (3.4 kHz). Because the PJP covers almost all frequencies of the human voice, it creates a realistic conferencing experience where you know who is talking simply by the sound of their voice.
Connect up to 8 locations
The PJP-50R/PJP-100H supports audio conferences of up to 8 locations without an MCU (multi-point control unit). The PJP-50R/PJP-100H support an audio conference of up to 8 locations without an MCU. They are able to connect 4 locations with a mesh connection and 8 locations with a cascade connection.
Smooth duplex performance
Echo can occur in an audio conference system when the microphones pick up the audio reproduced from the speakers. To counter this, the Projectphone incorporates a high-performance adaptive echo canceller that eliminates any of its own sounds that may be picked up. Combining this with a design that minimizes feedback of reproduced sound, it offers smooth full-duplex performance that is easy on both the talkers and listeners.
Adaptive echo canceller for use in any kind of room
The Projectphone's adaptive echo canceller "studies" the operating environment and automatically adjusts the echo-filter factor accordingly. And since it performs well in many different acoustic conditions, you can easily hold conferences in whatever room or space is available.
Zone mode ideal for quiet rooms
Zone mode is recommended for quiet rooms because of its wide audio pickup range. Thanks to the high audio pickup performance of the arrayed microphones, voices are picked up accurately even when distant from the microphones. Because the pickup range can be adjusted, you can ensure that even noises such as from air conditioners and projectors are not picked up.
Spot mode ideal for noisy environments
In this mode, the narrow directivity of the arrayed microphones allows you to pick up sound only from a specific direction. It is the optimal mode for conferences limited to one or two participants, or in noisy environments. Even in open spaces or when surrounded by noise, only the unadulterated voices of the talkers are heard by the other party.
Tracking mode automatically detects the talker
A microphone beam position detection function automatically tracks in the direction of the talker. Only the unadulterated voice of the talker is picked up. Also, by setting the PJP to pick up in a certain direction, you can ensure that it will not track noise sources such as projectors. Because pickup is directed toward the talker, this mode is useful for multiperson conferences where noise sources are present.
Talker position reproduction makes even multiperson conferences "easy on the listeners"
Microphone array's talker detection function and the speaker array's virtual sound source generation function combine to facilitate the "talker position reproduction mode", which modifies audio output according to the seating arrangement of the callers. Sound is output to different positions making it easy to distinguish between multiple voices (up to three talkers per location over a two-location IP conference).
Easily distinguishable voices without crossing, even during multipoint conferences The voices can be reproduced from different speakers according to the locations of the talkers. Outputs without crossing not only make voices easily audible, but also enable easy identification of the location of the current talker.
Web browser-based settings application
This program enables entry of various settings from a Web browser, including communication settings for connection with SIP servers and audio mode settings.
Audio & Video
The PJP-50R is now UNIVERGE certified product.
By linking SIP with the NEC SV8500/SV7000, you can use the PJPs as extensions.
As intuitive to use as an inside phone line
An IP-PBX and SIP server can assign extension numbers to each PJP-50R. You can call other conference participants by dialing their extension numbers instead of their IP addresses.
Use the same extension number even in different locations
No matter what floor, conference room, or desk, as long as you use the same network you can use the same extension number even when the IP address changes.
Enables conferences with multipoint connections
Using the VS-32 (an SV8500 peripheral) which controls multipoint connections, you can hold conferences with up to 32 locations simultaneously.
Audio conferencing with participants outside of the company
An IP-PBX and SIP server connect with external lines to enable participants to join a conference through the telephone, without needing audio conferencing system terminals.
Notes on linking with UNIVERGE
- For the SV8500's "terminal type" setting, assign the PJP as a "TYPE 1: Standard SIP terminal."
- When PJPs are linked with the SV8500, the PJPs' multipoint connection functions cannot be used. The SV8500 VS-32 is required for multipoint connections.
- Calls using the IP call function cannot be made between a PJP that is linked with an SV8500 and one that is not.
- Firmware with UNIVERGE SV7000 support
Support for UNIVERGE is included on firmware version 2.02 for the PJP-50R. Firmware versions prior to these will not work-please download the latest firmware.
- Firmware with UNIVERGE SV8500 support
Support for UNIVERGE is included on firmware version 2.10 for the PJP-50R. Firmware version prior to this will not work-please download the latest firmware.
- Shimizu Corporation
Created an active multi-person Web conference system with the PJP-100UH high performance conference microphone speaker.
Adopted Model: PJP-50R PJP-100UH
- RECRUIT MANAGEMENT SOLUTIONS CO., LTD.
Combined Web conference system and electronic whiteboard with the PJP-50R for an optimized remote conferencing experience.
Adopted Model: PJP-50R
- MITSUBA Corporation
Installed an IP audio conference system with over 100 units nationwide to achieve high-fidelity, easy-to-operate remote conferencing.
Adopted Model: PJP-50R PJP-100UH PJP-100H
CA-50R extension cable for the PJP-50R [Discontinued]
* Connect this cable before inserting the Projectphone power cable.
* Be sure to connect the AC adapter to this cable before connecting the power cable to the AC adapter.
Names and functions of parts
|Names and functions of parts|
|1. LINE plug||Connects to the Projectphone's LINE port.|
|2. LAN plug||Connects to the Projectphone's LAN port.|
|3. DC 12 V plug||Connects to the Projectphone's DC IN 12 V terminal.|
|4. LINE port||Connects to the phone line using a modular cable.|
|5. DC IN 12 V terminal||Connects to the AC adapter that comes with the Projectphone.|
|6. LAN port||Connects to the network using a LAN cable.|
|Outer dimensions (W × H × D)|
|Plug||36.5mm × 19mm × 48mm(Case dimensions)|
|Terminal||68mm × 28.2mm × 68mm(Case dimensions)|
|EAN||49 60693 23316 0|
|External interfaces||Ethernet (10/100Base-TX) Analog phone/modular jack, Stereo analog audio in/out × 1 each (mini-jacks) AC adapter connector (DC-12 V IN): for the provided AC power adapter|
|Installation attitude||Horizontal ±5°|
|Power consumption||Approx. 8 W|
|Radio interference standard||FCC part 15 (US), EN 55022 (EU), VCCI-class A|
|Operating environment||Operating temperature: 0-40°C(32-104°F) Operating humidity: 20%-85% (no condensation)|
|Weight||Approx. 1.4 kg (excl. power adapter)|
|Dimensions||283.4(W) × 51.5(H) × 297.5(D)mm|
|Power supply||100 to 240 V AC (50/60 Hz)|
|Accessories||AC adapter, power cable, LAN cable, phone cable, owner's manual|
|Other||Logging facility (SYSLOG) Firmware revision update (HTTP, TFTP) Settings (Web menu and front panel key operations)|
|Arrayed microphones||16mics, Point audio source control, directivity control, diffusion mode (monaural output) Divide mode, talker position output mode , max positions: 3 Output level: 85 dB|
|Frequency range||300-7000 Hz|
|Signal processing||3-ch adaptive echo canceller, noise reduction, microphone/speaker array control|
|Connecting locations||Mesh connection (max. 4 locations), cascade connection (max. 8 locations)|
|Transmission band required for audio codec||G.711 extended original system (fs = 16 kHz)||2-location call: 160 kbps, uplink and downlink / 3-location call: 320 kbps, uplink and downlink / 4-location call: 480 kbps, uplink and downlink|
|μ-law / G.711 A-law||2-location call: 90 kbps, uplink and downlink / 3-location call: 180 kbps, uplink and downlink / 4-location call: 270 kbps, uplink and downlink|
|G.726||2-location call: 60 kbps, uplink and downlink / 3-location call: 120 kbps, uplink and downlink / 4-location call: 180 kbps, uplink and downlink|
|G.729a||2-location call: 24 kbps, uplink and downlink / 3-location call: 48 kbps, uplink and downlink / 4-location call: 720 kbps, uplink and downlink|
|Other functions||Compliance with DHCP, and SIP; internal clock time synchronization with SNTP server|
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