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VoIP Analysis & Simulation

VoIP Call Generation & Analysis - Product Of The Year Award 2004

  Download Whitepaper on "Testing ATAs, Gateways, VoIP PBXs, and other Signal Processing Elements in VoIP Networks"


PacketCheck™ - Software Ethernet Tester  

GL's PacketCheck™ is a PC based Ethernet test tool that is designed to check frame transport ability, and throughput parameters of Ethernet and IP networks.

It can be used as a general purpose Ethernet performance analysis for 10Mbps, 100Mbps and 1Gbps Ethernet local area networks. The PacketCheck™ makes use of the network interface card (NIC) in the PC to transmit and receive raw Ethernet packets over the network. Throughputs up to 500 Mbps can be easily tested.


IPNetSim™ - IP Network Simulator

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)

GL's Message Automation & Protocol Simulation (MAPS) is a protocol simulation and conformance test tool that can be applied to a variety of protocols such as MGCP, SIP, MEGACO, SS7, , ISDN, GSM, and others.

This message automation tool covers solutions for both protocol simulation and protocol analysis. The application includes various test plans and test cases to support the testing of a required real-time scenario.

Along with automation capability, the application gives users the unlimited ability to edit messages and control scenarios (message sequences). "Message sequences" are generated through scripts.

Applications

  • Complete analysis and simulation capability on par with any protocol tester in the market
  • Provides fault insertion, and erroneous call flows testing capability
  • Functional testing, Regression testing and Conformance testing of network elements
  • Ready scripts makes testing procedure simpler, less time consuming and hence time to market products

TDM / VoIP Gateway Tester

MAPS is an ideal tool to evaluate Gateway / ATA with TDM and VoIP Interfaces. The tester supports T1 E1 PRI ISDN, Megaco, and SIP interfaces. Product features include call connectivity, call signaling, traffic generation, voice quality testing, codec, and hundreds of other features. For more details, contact GL Communications.

Other notable features include -

  • Interfaces to Portable T1E1, and VoIP
  • Support for SIP protocol conformance testing
  • Test Echo canceller performance and compliance
  • Multi-protocol call trace for TDM / VoIP

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. 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.

PacketScanWeb™

PacketScanWeb™ remotely works with PacketScan™, a central web server and database to facilitate result display using web based clients. Users can view real-time data, navigate through records, filter the collected VoIP traffic summary, and graphically analyze the call volume, MOS, call completion, failed calls, completed calls, PDD, and more through a simple web browser.


PacketProbe™

GL's PacketProbe™ is an advanced embedded VoIP monitoring software. The PacketProbe™ passively monitors VoIP traffic carried over LAN or WAN by producing real-time per call and per-stream voice quality metrics. Call Detailed Records (CDR) along with voice quality including Mean Opinion Score (MOS) and other vital diagnostic information provide network managers immediate visibility into service quality, call volume, and call details. Service providers should be able to rapidly drill down and diagnose voice quality problems. PacketProbe™ software can be used as an embedded application or as standalone probe appliance. Multiple PacketProbes™ can provide network wide voice quality visibility through GL’s PacketScanWEB™ or other NMS systems.


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. 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

RTP ToolBox™

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, Jitter Implementation and more) within end-user equipment (IP Phones, ATA, MTA and more).
  • 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.

Testing EC on VoIP ATAs and Gateways

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.


G.168 EC Compliance Testing for VoIP Network Elements

GL has developed two powerful tools to characterize echo, noise, and speech characteristics of voice connections. These tools are applicable whether the underlying network technology is PSTN, VoIP, ATM, Wireless, or Satellite. Typical problems that are easily diagnosed are:

  • Echo path delays exceeding the EC's delay limit, click here for a diagram
  • Impulse response dispersion spans across EC's limit
  • NLP is turned off causing residual echo to be heard
  • Impulse response has unusual multipath components that the EC is unable to converge on

Following are the various solutions offered for performing EC testing on VoIP networks. Click on each solution for more details


Voice Quality Testing on VoIP ATAs

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.


SIGTRAN Analysis

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™ - H.323 Single Call Generator

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.


Near Real-time Voice-band Analyzer

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.


Audio Processing Utility

Audio processing utility is a pre-processing application used for manipulating the input audio (tones, and voice) files with impairments such as delay, noise, acoustic echo, line echo, and double talk. An echo file is generated if a corresponding line or acoustic echo filter is specified. It analyzes the given coefficients files and calculates the applied delay & ERL. The degraded audio output file can then be visually analyzed or used for further speech analysis with GL’s Voice Quality Testing, Digital Echo Canceller & other applications.


VoIP VQT

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.


Fax and Modem Analysis – IP

GLInsight™ with FAX for IP is a powerful and feature rich offline software tool that analyzes pre-recorded IP fax calls. GLInsight™ with MODEM for IP analyzes modem traffic recordings within IP packets, by depacketization of the PCM signal, appropriate signal analysis, demodulation, phase analysis, error correction and data decompression.


VoIP Network Testing w/ GL's Media Gateway

The DCOSS MG (Media Gateway) system supports dual redundant Ethernet trunks for traffic and signaling testing of VoIP and IP networks. The DCOSS / MG combination provides user friendly bulk call generation and reception for various voiceband traffic types including voice, fax, and modem. It can be connected to any IP Phone, softphone, VoIP PBX, or VoIP Network / Cloud.


Complete VoIP Lab

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

 
 
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