Grandstream GXV-3140

Grandstream GXV-3140

Grandsteam GXV-3140 video phone from Grandstream Networks represents a next generation of interpersonal IP multimedia communications. Extraordinary quality of video, advanced telephone functions and rich user interface are implemented in a smooth industrial design of this stand out device. This model of the phone has crystal clean color 4,3 inch display and 1.3 megapixel CMOS camera, built in dual port switch and wide range of additional interfaces, such as slots for SD cards, USB port, stereo headsets, audio and video output etc, advanced standards of video compression H.263/263+, H.264 and full duplex speaker phone.

This device implements a rich mixture of „all in one”: possibility of real time video conference, support for various WEB and social networks, Internet radio, RSS mail, music streams, games and many more. Grandstream GXV-3140 creates a powerful converged platform for the next generation of IP multimedia communications.
It can work through Wi-Fi with help of USB adapter. Technical Characteristics: 3 SIP lines Color 4.3 inch LDC display with a resolution of 480x272 pixels. 1.3 megapixel CMOS camera Dual port 10/100 Mb/s switch Optional FXS/FXO port SD/MMC/SDHC/, USB, stereo headphones with mic, stereo audio output, video output, 2 default display positions, wall mount High quality full duplex speaker phone with acoustic echo cancellation H.264 base video codec (CABAC) H.263 and H.263+ enhanced video codec with a standard videobitrate from 64kb up to 1Mb and 30 frames per second, QVGA/CIF/QCIF resolutions. Web browser, Yahoo IM, MSN, Google, RSS, Internet Radio, audio/video streams, alarm clock, calendar, games, various ringtones Intuitive user interface, with custom configuration, multi language interface Advanced NAT technology, plug & play High security standards based on TLS/SRTP/AES

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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.

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