PacketGen(TM) is a PC-based real-time VoIP bulk call generator for stress testing and precise analysis of the VoIP network equipment. The application 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.

Calls can also be made to IP phones and to IP Analog Telephone Adapters. PacketGen™ can be used to test basic functionality and verify proper protocol implementation in SIP based equipment such as SIP phones and Network servers, as well as Proxy Servers, Registrar servers, and PSTN and Media Gateways

 
Features: 

• Distributed architecture for GUI, SIP and RTP systems (provides high call rates and media streams) 
• Generates both SIP signaling & RTP traffic (voice, fax, digits, tones)
• Full SIP Functionality - Call Forwarding, Call Hold, Call Transfer, etc. 
• Manual and Bulk Calling capabilities 
• Automatic generation of impairments over the RTP for any (or all) established calls. The impairments that can be generated include:  
   o Latency: Fixed, Uniform, Nominal  
   o Packet Loss: Periodic, Random, Burst (burst probability and burst size)  
   o Packet Effects: Out of order, Duplicate Packets
• Fully Remote Controllable
• Automation through easy-to-build scripts
• E-Model based MOS and R-Factor score

SIP Core 

• 500 simultaneous calls per SIP Core per PC (20 to 50 cps/PC) 
• Multiple SIP Cores in a distributed architecture permits capacity scaling

RTP Hardware Core

• RTP Hardware Core supports 120 ports with G.711 
• Audio Codecs supported by RTP Hardware Core are 
  • G.711 
  • G.723.1 
  • G.726 
  • G.729 
  • G.729a 
• Optional 16 or 32 Port Analog Phone Connectivity 
• Optional T.30 Fax Support for RTP Hardware

RTP Software Core 

• 200 Simultaneous calls per PC running single SIP/RTP Software Core, 
• Distributed architecture permits 150 Simultaneous calls per each additional PC 
• RTP Software Core uses simulation of traffic by file transmission and reception 
• Audio Codecs supported by RTP Software Core are G.711 (A-law and u-law), G.729, G.726, GSM

For more details, please visit our website http://www.gl.com/packetgen.html

RTP Toolbox(TM) – RTP Media Emulation

RTP ToolBox(TM) 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.

This tool can be used for 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. The application can be used for:

• 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. 

Features: 

• Supports standard codecs such as G.711 (mu-Law and A-Law), GSM (Full Rate), G.729a, G.729b, G.726 (All Rates), G.722, G.722.1, ISAC, AMR (All Rates), EVRC (All Rates), SMV (All rates), iLBC*,  SPEEX
• Automatic scan option to capture all incoming RTP traffic.
• Monitoring RTP streams using scalable Oscilloscope and Spectrum Analyzer. 
• Generation/Detection of in-band and out-of-band Digits/Tones (DTMF, MF, user-defined, etc.)/Events per RFC-2833 
• User-defined impairments: latency, packet loss, out of sequence and duplicate packets. 
• Detailed statistical information of RTP and RTCP packets. 
• Sending and recording of voice files with a synchronous TX/RX option. 
• G.168 testing for echo cancellation equipment. 
• Talk and Play to Speaker options using PC sound card. 
• Call Generation and Reception ability provides UA simulation 
• Quality Metrics with R-Factor and MOS Factors, Jitter Buffer Statistics, Degradation Factor, Burst Metrics, and Delay Metrics are graphically represented 
• Supports Client-Server functionality (requires additional license)-C++, & TCL clients 

For more details, please visit our website http://www.gl.com/rtptoolbox.html

Message Automation & Protocol Simulation (MAPS)- SIP Conformance Tester

The Message Automation & Protocol Simulation (MAPS) -SIP supports testing SIP proxy servers, Redirect servers, Registrars and user agents such as SIP phones. The MAPS - SIP Conformance Suite tool is designed with 300+ test cases, as per SIP specification of ETSI TS 102 027-2 document.

Test cases include general messaging and call flow scenarios for multimedia call session setup and control over IP networks. Logging and pass/fail results are also reported. Test cases verify conformance of actions such as registration, call control, capability queries and messaging for registrars, proxies and redirect servers.

The application gives the users the unlimited ability to edit SIP messages and control scenarios (message sequences). "Message sequences" are generated through scripts. "Messages" are created using message templates. MAPS can also be used as a Bulk call generator, which generates and receives SIP calls up to 100,000 calls per system. A single MAPS can act as more than one SIP entity at a time and can generate any SIP message on wire in VoIP network and hence most of testing can be automated and also equipments needed to test are reduced.

Main Features

• Simulates UAC, UAS and Registrars
• Supports full customization of all SIP headers and message body
• Provides fault insertion, and erroneous call flows testing capability
• Tests instant messaging and push-to-talk features  
• Handles Retransmissions and remote Retransmissions
• Supports both UDP and TCP
• Scenarios are created using simple commands saved as .txt file
• Ready scripts makes testing procedure simpler, less time consuming and hence time to market SIP products
• Commands in the script supports conditional branching loops and user defined variables 
• Supported on Windows XP/2000 Operating System

For more details, please visit http://www.gl.com/mapssip.html

Message Automation & Protocol Simulation (MAPS)- Megaco Conformance Tester

Message Automation & Protocol Simulation (MAPS) designed for MEGACO testing provides MEGACO test library capable of testing Media Gateways (MG) and Media Gateway Controllers (MGC). The test library includes test functions to generate MEGACO messages, edit messages, simulates a variety of call flows, and control scenario. "Message sequences" are generated through scripts. "Messages" are created using message templates.

The MAPS - MEGACO Conformance Suite tool is designed with 300+ test cases, as per specification of ETSI TS 102 374-2 document. The test suite includes in-built scripts to process MEGACO valid and in-valid behaviors.

Main Features:

• Simulates Media Gateways (MG) and Media Gateways Controller (MGC)
• Generates and processes MEGACO valid and invalid messages
• Supports Add, Subtract, Notify, Modify, Move, ServiceChange, AuditValue, and AuditCapabilities commands
• Supports fully integrated, complete test environment for MEGACO
• Supported on Windows XP/2000 Operating System
• Supports full customization of all MEGACO commands
• Ready scripts makes testing procedure simpler, less time consuming and hence time to market Megaco products
• Provides fault insertion, and erroneous call flows testing capability

For more details, please visit http://www.gl.com/mapsmegaco.html

Gateway Tester – “MAPS SIP – MEGACO” Conformance Test Tool

MAPS - SIP and MEGACO Conformance test tool is an ideal tool to evaluate Gateway / ATA product features such as call connectivity, call signaling, traffic generation, voice quality testing, codec, and hundreds of other features.  Other notable features include –

·         Portable T1E1, Analog, and VoIP Interfaces

·         IP and Megaco protocol conformance testing

·         Echo canceller performance and compliance

·         Multi-protocol call trace for TDM / VoIP

For more details, please visit http://www.gl.com/maps.html