Grandstream BT 201 IP Phone

Grandstream BT 201 IP Phone

The Budgetone Series from Grandstream brings to the world highly cost-competitive, latest generation SIP phones. In designing Budgetone series a goal was to employ its groundbreaking innovative technology in order to delver to the market IP phones that will provide their users with unparalleled sound quality, extensive list of features, and simplicity - all at an incredibly low price.

The Grandstream BT 201 is without a doubt ne of the most desirable, feature rich phone in the BudgeTone series. The BT 201 model supports an number of highly technologically advanced telephony features incorporating a full-duplex hands-free loudspeaker with advanced acoustic echo cancellation technology, 10M/100M Ethernet port, a number of popular audio codecs, voicemail with visual indicator and custom ringtones, all in all putting up unbelievable value to both business and consumer market. But if you already have a set up of analogue phones, maybe you might be interested interconnecting your setup with Grandstream ATA HandyTone-502 Analog Telephone Adapter the latest product in the HandyTone series product line.

The Grandstream BT 201 specifications include:
• 1x 100 Mbps auto sensing Ethernet RJ45 ports • 1x LAN RJ45 10/100Mbps • 1x 2.5mm Headset Port • 1x LED (red color) • 1x Full-duplex loudspeaker • 25-Keys Keypad and 12-Digit CallID LCD display • SRTP, TLS • Voicemail light indicator • FCC / CE / C-Tick Compliance • All of the standard features and functionality of voice • Fully compatible and Interoperable with SIP platforms • Power Adapter Input: UL certified, 100-240VAC 50-60 Hz, Output: +5VDC, 1200mA • Dimensions: 18cm x 22cm x 6.5cm W/D/H

Providers in database: 6433
Register VoIP Provider

RTP Delivers Multimedia Data In Real Time

RTP Delivers Multimedia Data In Real TimeA Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.

View more WIKI
Colombia HablaporInternet

Build or grow your VoIP company in hours. Hablaporinternet will provide you all the hardware, software and professional services to put in production a high voice quality service.

Read more