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The VQuad™ provides the one box solution for sending/recording voice over a variety of network interfaces in an automated and synchronized manner. In addition to sending/recording voice, the VQuad™ also supports automated call control for establishing calls and verifying the call is up prior to sending/recording voice. A GPS option permits drive testing with VQT MOS scores plotted to local street maps (most all countries supported). Network interfaces supported with the VQuad™ include:

• Wireless phone, Wireless radio, WiFi Phone
• VoIP Simulation (supports both SIP and H.323 protocols)
• 2-wire Analog Simulation (FXO port)
• T1/E1 Simulation (supports both CAS and PRI ISDN protocols)

The GL VQT software provides the voice quality measurement and analysis tools for all types of networks carrying voice traffic. The VQT provides the analysis of the recorded voice files using ITU specifications P.862.1 (PESQ), P.861 (PSQM), and P.800 (PAMS). The VQT is an intrusive quality of service algorithm that can be executed in an automated fashion along with the VQuad™. For more call density, the software can be used with DCOSS, APS, PacketGen™, or T1/E1 Analysis products. Results are saved to a database that may be queried using the remote NetViewer application.

The VQuad™ provides the one box active QoS solution for sending/recording voice over a variety of network interfaces in an automated and synchronized manner. Using the GL VQT, analysis of the recorded voice files can automatically be executed from the VQuad™ in an intrusive manner. The VQT utilizes the ITU algorithms PESQ, PSQM, and PAMS. In addition to sending/recording voice, the VQuad™ also supports automated call control for establishing calls and verifying the call is up prior to sending/recording voice.

The GL Passive Quality of Service testing allows one to monitor voice anywhere within a network (using the desired VoIP/TDM/Analog Network probe) and gather statistics and quality of service measurements. The various network probes utilize GL hardware and software applications to support T1/E1, Analog, and VoIP networks. All network probes are non-intrusive and require no active testing on either side of the call. Rather, the non-intrusive Passive Monitoring utilizes the metrics from the voice or VoIP RTP and provides statistics, analytical measurements, and in certain cases MOS results.

The Global Network VQT system can be achieved using multiple VQuad™ systems (nodes) at various locations, accessing the network as wireless, analog, VoIP, and TDM. A Central Command Center will provide full control of all nodes by running user-defined scripts for all desired tests. A central VQT will provide the voice analysis of all nodes along with additional analytical metrics, round trip delay measurements, jitter, clipping, and call control statistics. Finally, a remote NetViewer application can remotely control the entire system, display real time statistics and status of the entire system, and query the VQT results and display in both tabular and graphical formats.

MobNetMon™ can monitor "off-air" GSM network signals as one drives through a GSM service area. Any of the world's GSM bands can be monitored with the included "quad-band" wireless phone and accompanying software. MobNetMon™ can determine weak points in a wireless network by generating calls, identifing primary and surrounding GSM base stations, decoding GSM protocol messages, including call traces and protocol analysis.

The Universal Telephony Adapter (UTA) is a hardware adapter that interfaces a PC sound device to telephony equipment. It features all the circuitry necessary to universally adapt to any telephony instrumentation, including standard Plain Old Telephone Systems (POTS), digital hybrid interfaces, Voice over Internet Protocol (VoIP) networks, wireless cellular phones and mobile radios. Used in conjunction with GL's VQuad™ application, the UTA can be directed to automatically send and record sample voice files between nodes of a telephony network. These files can then be submitted to GL's VQT software application for further analysis.

The DCOSS APS (Analog Phone Simulator) converts a GL DCOSS into an analog phone simulator that simulates a bank of up to 384 analog telephones. Using a basic DCOSS with T1 trunks along with the APSCB24/48 external boxes, the DCOSS APS may be used to test a Central Office (CO), PBX, Gateway, or other telecommunications equipment, which provide local loop interfaces. The DCOSS APS can generate/receive analog calls and send/receive fax, modem, voice files, DTMF/MF digits and Single/Dual Frequency Tones. The DCOSS APS can be executed in manual mode or automated Bulk Calling with scripting support.

The BRI Phone Simulator (BPS) converts GL's DCOSS into a BRI ISDN Terminal Phone Simulator. The BPS can simulate a bank of up to 128 BRI telephones (terminal side), each supporting one D-channel and two B-channels. Using a basic DCOSS with E1 PRI ISDN trunks along with a BPS external box, the DCOSS BPS may be used to test a BRI Network.

The DCOSS converts any windows based PC (Windows 2000/XP) into a central office complete with T1 and/or E1 trunks as well as Analog and/or BRI ISDN phone interfaces. The DCOSS supports a multitude of protocols including CAS, PRI ISDN, SS7 and GR-303. The DCOSS supports manual and automated Bulk Calling and can simulate a wide variety of traffic.

The T1/E1 Analyzer is a powerful voice band, data, signaling, bit error rate tester, and protocol analyzer. It can perform analysis and emulation of various signal types including voice, digits, and tones; various protocols including HDLC, ISDN, SS7, CAS, Frame Relay, GSM, GPRS, CDMA, MLPPP, ATM, and UMTS. It is capable of T1 or E1 PCM signal visualization, capture, storage, analysis, and emulation.

PacketGen™ is a PC-based real-time VoIP bulk call generator for stress testing and precise analysis of the VoIP network equipment. PacketGen™ is based on a distributed architecture, wherein SIP and RTP software cores can be modularly stacked in one or many PCs to create a scalable high capacity test system. An optional hardware RTP can support 120 real-time voice calls from real phones, or fax calls from fax machines.

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. Applications include testing of IP phones, Gateways, IP Routers and Switches, and Proxies. Hundreds of calls can be monitored in real-time including detailed analysis of selected voice band streams.

GL's RTP ToolBox™ testing and simulation tool is designed not only to monitor RTP and RTCP packets, but also to allow users to manually create and terminate RTP sessions, independent of call-signaling protocols such as SIP, H323, MEGACO, or MGCP.

The Near Real-time Voice-band Analyzer (VBA) is an analysis tool for monitoring voice band network traffic. The VBA can host different analysis modules for monitoring speech and noise levels, line echo, and acoustic echo. The standard modules included in the application are ITU-T P.56 Active Voice Level analysis, Line Echo (Hybrid) analysis, and Acoustic Echo analysis. Other analysis modules such as ITU-T P.561, P.562, and P.563 can be hosted as plug-ins.

GL’s UTA features a built in Round Trip Delay (RTD) measurement capability. Two UTAs can automatically determine the RTD, regardless of what network they may be analyzing. A one-way MANUAL delay measurement is also achievable using the UTAs and Adobe Audition.

GL Communications has the broadest range of testing solutions for echo cancellers, that provides accurate measurements of echo paths, impulse responses, delay, loss, echo return loss (ERL), and echo path delay (EPD). Tools are provided for thorough G.168 compliance testing including hybrid simulation, application of stimulus, capture of response, and graphical analysis of response.

This system provides real-time voice quality measurement across a diverse set of networks. Voice calls are automatically placed between end points; quality is measured and provided for display at an NMS. Voice measurements include MOS (Mean Opinion Score), round trip delay (RTD), jitter, clipping, voice levels, etc. The essential elements of this system are Wireless / Wireline / VoIP Nodes to establish calls and send / receive voice files in real-time, and the Regional Command Center (RCC) to control the nodes. It also includes Remote Client NetViewer to remotely control the RCC, access, sort, and display all data collected by the RCC.