Test & Measurement
Portable Platforms for GL's VQT, IP, and VoIP Products
GL Communications Inc. announced today the release of portable platform solutions for Voice Quality Testing (VQT), IP, and VoIP Products. In a statement released to the press Mr. Rob Bichefsky, Senior Manager, at the company said, VQuad™ with the Dual UTA can be purchased with a NetBook PC for a truly lightweight portable solution. The Dual UTA is fully compatible with the miniature NetBooks and allows the VQuad™ software to be installed for full testing without any performance issues. He added, NetBook PCs are smaller and less expensive than traditional notebook computers, and they have a smaller screen, are lighter, and normally have a longer battery life.
Mr. Bichefsky further added, GL's VQuad™ with Dual UTA hardware purchased along with the NetBook PC can be conveniently packaged as a complete voice quality testing unit and shipped to intended location. The Dual UTA is connected to the NetBook PC using an available USB connection. The Dual UTA supports up to two mobile phones simultaneously (mobile phone, Bluetooth phone, Analog FXO, Phone Handset, PTT Mobile Radio) and up to two Dual UTAs can be supported on a single NetBook (thus allowing a maximum of 4 simultaneous devices). Both the GL VQuad™ and GL VQT applications can be remotely accessed via the Internet.
Other IP and VoIP Portable Products are:
PacketCheck™ - Software Ethernet Tester
PacletCheck™ – software ethernet tester and is designed to check Ethernet and packet transport integrity and throughput.
PacketGen™ - SIP Bulk Call Generator
PacketGen™ is a PC-based real-time VoIP bulk call generator for stress testing and precise analysis of the VoIP network equipment.
PacketScan™ - SIP / H323 / Megaco / MGCP / RTP / RTCP / Video Analysis
PacketScan™ - is a real-time VoIP analyzer that captures live IP traffic, and segregates them into SIP/H323 calls and collects statistics about the calls.
RTP Toolbox™
RTP ToolBox™ is designed not only to monitor RTP and RTCP packets, but also allows users to manually create and terminate RTP sessions, independent of call-signaling protocols such as SIP, H323, MEGACO, or MGCP.