GL provides a complete "VoIP Lab" for test, simulation, measurement, and analysis of VoIP networks and other networks that connect to VoIP.

IPNetSim™ can simulate an entire IP network in a single box. All the conditions encountered in a real-time IP network are simulated such as network latency, network delay variation (jitter), bandwidth, congestion, packet errors, bit errors and other link impairments independently in both directions at speeds of up to 100 Mbps (or1Gbps per link). It can be used to test IP end point devices such as (Gateways, IADs, IP phones, Soft phones, and so on) for voice, fax, data, or multimedia transmission over IP.

Message Automation & Protocol Simulation (MAPS™) is a SIP test tool/traffic generator, which is used to simulate any interface in a SIP network and perform protocol conformance testing (SIP protocol implementations, Megaco). MAPS as a testing tool can simulate User Agents (User Agent Server- UAS, User Agent Client - UAC), Proxy, Registrar, and Redirect Servers. The tool is designed with 300+ test cases, as per specification of ETSI TS 102 027-2 document.

Some of the prominent features of MAPS are listed below,

• Simulates UAC, UAS, proxy, Registrars, Redirect Servers, and so on
• Generates single call and bulk traffic
• Generates upto 100, 000 calls per system
• Tests instant messaging and push-to-talk features
• Performs comprehensive RTP media testing, IMS testing (CSCF, Application Server)

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.

Some of the prominent features of PacketScan™ are listed below,

• Users can perform a host of activities on the captured calls, allowing you to get an exact picture of QOS (quality of the service) and the technical adherence (adherence to the protocols specified by the standardizing authority) of the system under test.
• It allows users to listen/record VoIP calls in real-time; perform power, frequency, spectral, tone and digit analysis with ease and precision.
• It's ability to monitor / record audio and video data of a session to files (in QuickTime *.qt format), allows users to perform powerful video analysis.
• The captured VoIP calls with video can be played back using 3rd party VLC Viewer application.
• Detailed call statistics such as packet loss, gap, jitter, delay, RTP performance statistics, and unparalleled voice band statistics can be monitored simultaneously.

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.

Some of the prominent features of PacketGen™ are listed below,

• Distributed architecture for GUI, SIP and RTP systems provides high call rates and media streams. This also makes it scalable & easy to add additional load generation capacity.
• Generates both SIP signaling & RTP traffic (voice, fax, digits, tones)
• Full SIP Functionality - Registration, Call Forwarding, Call Hold, Call Transfer, Authentication, etc.
• Manual and Bulk Calling capabilities with complete flexibility on each call session

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.

Some of the prominent features of RTP ToolBox™ are listed below,

• Testing and developing enhanced voice features (VAD, Echo Cancellation, Codec, Digit Regeneration, Digit Generation, Fax over IP, Jitter Implementation etc) within end-user equipment (IP Phones, ATA, MTA etc).
• Testing media gateway telephony interfaces.
• End-to-End network testing before and during VoIP deployment.
• Automated testing of Digital Signal Processing embedded into network elements.

GL provides several tools to thoroughly test VoiP ATAs - PacketGen, PacketScan, RTP Toolbox are some of the tools. The analog telephone adapter (ATA) permits consumers to easily utilize the broadband services for VoIP with a conventional landline. It has lot of advantages as it preserves the existing infrastructure (phones, wiring, jacks etc). These ATA's are totaly inexpensive and portable and also they provide very easy number mobility.

The VoIP phones and ATA's must be free of noise and other hindrances that may cause a degraded voice quality when using this equipment. GL's VQuad™ w/ FXO Analog option and/or VoIP option, along with Voice Quality Testing (VQT), provides the complete automated network testing solution including four analog 2-wire interfaces (RJ-11), up to eight SIP User Agents and the voice quality measurement and analysis tools.

GL's SIGTRAN protocol decoder software permits real-time analysis, call trace, capture, and filter of SS7 and ISDN signaling messages over IP protocol. This software is available as optional software under GL's PacketScan software. It is also complementary to GL's award winning SS7 and ISDN protocol analyzers.

PacketH323™ is a PC-based real-time VoIP call generator for providing voice band testing over the H.323 VoIP Protocol. PacketH323™ can generate single calls, either manually or in an automated fashion, and generate an array of traffic once the call session is established. Call Records, Call Statistics, Protocol Events are all captured and displayed to the user in real time.

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.

Providing clear, uninterrupted voice is critical in Network and Echo Cancellation development. GL's Voice Quality Testing (VQT), accessed through an easy to use GUI interface, along with the many supported platforms for transporting voice, provides the voice quality measurement and analysis tools for all types of networks carrying voice traffic.