PacketGen™ Software Ver 3.5.7 is Now Available! Download it here
Overview |
SIP Call Generation |
Supported Protocols |
Traffic Generation |
RTP Action Scripting
Statistics and Events |
Command Line Interface |
Voice Quality Testing using PacketGen™
Screenshots |
Specifications |
Other VoIP Products |
Buyer's Guide
Overview
PacketGen™ is a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP
generation) 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.
GL's PacketGen™ breaks ground with high density performance
PacketGen™ on a Duo Quad Core PC can support 1000 simultaneous calls with, both SIP and RTP generation. This
performance number is associated with using the G.711 codec, other codecs may provide higher call densities.
PacketGen™'s distributed architecture allows achieving higher call density by interconnecting more number of systems with SIP
and RTP software cores.
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.
Main Features:
- Distributed architecture for SIP and RTP systems provide high call rates and media streams. Also makes it scalable i.e
easy to add additional load generation capacity
- PacketGen™ breaks ground with high density performance; PacketGen™ can generate 1000 simultaneous calls on a Duo Quad Core PC.
Higher density is also achievable using multiple PC systems
- 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
- Send/Record voice files on any (or all) RTP sessions. Also, provides the necessary voice quality algorithms, thus providing
the ITU standard PAMS, PSQM, PESQ MOS scores
- Perform various actions like send / detect digits / tones (both Inband and Outband), talk and playback actions on any
(or all) RTP sessions to simulate real world traffic
- Supports run-time parameters to control call and traffic behavior – SIP Call Parameters (timers, Reliable Provisional Response,
RTP Source, Packetization time for packets in RTP traffic, Control sending/ detecting of Outband digit codec, Receive Jitter Buffer,
Authentication of incoming calls, and more) and Digit Generation and Detection parameters (power, on/off, pause, and amplitude).
- Supported Codecs
- G.711 (A-law and µ-law - 64 Kbps)
- G.711 App II (A-law and µ-law with VAD)
- G.729A, G.729B (8 kbps)
- G.722
- G.722.1 (24 kbps and 32 kbps Wideband)
- G.726 (40/32/24/16 kbps)
- GSM (13.2 kbps)
- AMR (Narrow band codec - 4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps)
- AMR (Wideband codec - 6.60kbps, 8.85kbps, 12.65kbps, 14.25kbps, 15.85kbps, 18.25kbps, 19.85kbps, 23.05kbps, 23.85 kbps)
- EVRC (Rates - 1/8, 1/2 and 1-)
- EVRCB (Rates - 1/8, ¼, ½ and 1)
- SMV (Modes - 0, 1, 2 and 3- Available if licenses are provided or owned, please call GL)
- iLBC (15.2kbps and 13.33kbps)
- SPEEX (Narrow Band)
- SPEEX (Wideband)
- iSAC (An optional Codec, must be purchased as a separate dongle extension)
- Powerful scripting capability for RTP traffic generation, which allows user to simulate/test IVR kind of systems. Allows for
conditional commands as well as script looping
- Automatic generation of impairments over the RTP for any (or all) established calls. The impairments that can be generated
include
- Latency: Fixed, Uniform, Nominal
- Packet Loss: Periodic, Random, Burst (burst probability and burst size)
- Packet Effects: Out of order, Duplicate Packets
- Remote access capability using GUI or command line interface
- Provides statistics, events and call records
Supported Protocol Standards
PacketGen™ is compliant to the UAC, UAS. Registrant and Redirect Server as per the RFC 3261 (SIP) along with backward compatibility of
SIP RFC 2543 and RFC 3262 (Reliable Provisional Response).
SIP Call Generation
SIP Setup and Configuration
The SIP setup screen controls the foundation of the desired test environment. The user has the flexibility to configure multiple SIP
and RTP instances on a local system and/or remote systems. Each SIP and RTP instance provides additional call density capabilities,
thus allowing a true distributed architecture. The RTP portion is available as either software or hardware. In addition, true RTP Load
sharing is provided within PacketGen™.
Screen Shot of Sip Setup Configuration Window
Manual and Bulk Call Generation
PacketGen™ supports both Manual and Bulk Call Generation, with complete flexibility on each individual call session
- Quick Configuration Utility - Multiple User Agents can be quickly created using any one User Agent as a reference.
- Current Status of each configured session - Status of each configured call such as Dial and HangUp are displayed.
- Traffic Generation and QOS Measurements - Once a call is established, traffic actions such as send files/tones/digits, loop back,
receive voice files, playback, and so on can be performed either manually or automatically
- Call Processing Options including Hold and Call Transfer - The user can place the call on hold and can transfer the call to a third
party by entering the IP address using either UDP or TCP method.
Screen Shot of Manual Call Generation Window
Screen Shot of Bulk Call Generation Window
SIP Registration with auto refresh
PacketGen™ provides facility to register a single or a bunch of User Agents simultaneously. Each Registration gives
flexible configuration options like Registrar server address, Address of Record, Expiry time etc. Also, each Registration can be
configured for automatic registration refresh, after the existing registration expires. A quick configuration utility helps to
configure hundreds of registrations easily.
Screen Shot of SIP Registration with auto refresh
Auto Signaling Action
This feature provides a quick and easy method to configure signaling actions, to be performed automatically as
soon as the call session is established. Configuration is based on call sessions, thus each call may be configured for unique activities.
Signaling options include Call Transfer, Call Reject (User-Defined Error), Hold and Re-Direct.
Screen Shot of Auto Signaling Action Configuration Window
RTP Traffic Generation
Once the call is established, PacketGen™ can generate a multitude of traffic, either manually or automatically. This
traffic handling provides the mechanism to test various network conditions and responses. Traffic actions include
- Send Actions – Send GL Proprietary voice files, PCM u-Law or A-Law, GSM, G.729A/B, G.726; Send DTMF or MF Digits (In-band or
Out-Band); Send user-defined single/ dual Frequency tones; Send real time voice from default audio device (microphone).
- Loop Back – Loopback real-time voice traffic from the received RTP/RTCP to the send RTP/RTCP (all received traffic will be
re-generated as send traffic).
- Receive Actions - Record received voice file in GL Propriety file format; Detect incoming single/dual Frequency Tones and DTMF/MF
Digits from in-band received voice; Play received voice to default audio device (speaker).
- Power Measurement – Shows an active receive signal level in dBm.
Auto-Traffic Action
Auto-Action feature provides a quick and easy method to configure signaling as well as traffic actions, once the call session is
established. Configuration is call session based, thus each call may be configured for unique activities.
Traffic options include transmit / record voice, generate / detect tones, digits and noise and send / receive fax.
Advanced traffic options like codec parameters, ptime and Rx jitter buffer control are provided.
Screen Shot of Auto Traffic Action Configuration Window
RTP Action Scripting
PacketGen™ provides a powerful scripting capability to control RTP traffic. Scripting features includes loops, conditional
statements, wait for events, timers etc., Scripting gives the user greater control over the RTP traffic being generated allowing users
to create/test IVR kind of applications. Scripts can be created using the RTP Script Editor, which allows an intuitive, point and click
script setup.
The set of script elements included in the script editor allows user to perform all the traffic generation / reception actions as done
using PacketGen™'s main graphical user interface.
Screen Shot of RTP Action Script Editor
Download Sample RTP Script: Transmission and Reception of DTMF digits
RTP Impairment Generation
PacketGen™ allows user to configure various impairments on outgoing RTP streams. These categories of impairments can be
generated -
- Latency
- Packet Loss
- Packet Effects
Statistics & Events
PacketGen™ provides detailed statistics for each User Agent, SipCore as well as for the entire system. Included in the statistics are
complete/incomplete calls, failed calls (based on user-defined thresholds) and type of generated traffic. Call Statistics window
provide detailed call wise statistics per SipCore.
Screen Shot of Statistics Window
System statistics window provides the overall call statistics such as active calls in progress, completed calls, number of
successful calls, attempted calls, and so on for each SipCore.
Screen Shot of System Statistics Window
All events and statistics can be exported and saved for record or review at a later time.
Command Line Interface
In addition to the GUI, PacketGen™ can also be operated through a Command Line Interface (CLI). All the functionalities of
the PacketGen™ GUI are supported, except the configuration functions. Users can thus operate PacketGen™ from a DOS based
console (instead of the GUI) or easily integrate PacketGen™ into their own applications.
Screen Shot of Command Line Interface
Download Sample PacketGen™'s CLI Script: Registration of SIP setup
Voice Quality Testing using PacketGen™
PacketGen™ also provides the necessary Voice Quality algorithms, thus providing the ITU standard PAMS, PSQM, PESQ
MOS scores. PacketGen™ can be used to establish calls and send/record voice files through the IP network. Voice Quality
testing can be completely automated using PacketGen™ CLI, RTP Action scripting, along with GL's ASR Listener, FCU and
VQT software.
PacketGen™ is one of the platforms for transporting voice in GL's comprehensive Voice Quality Testing (VQT) solutions.
For more details, click here for complete Voice Quality Testing information.
PacketGen™ Screenshots
Specifications
Recommended PC
- Windows 2000/XP/Vista, 2 GHZ, 512 MB RAM, 40 GB Hard Drive, 10/100/1000 Ethernet Port, Parallel or USB port for Dongle,
Headphones and Microphone
SIP Core
- Multiple SIP Cores in a distributed architecture permits capacity scaling
RTP Software Core
- 1000 simultaneous calls per Duo Quad Core PC running single SIP/RTP Software Core,
- Distributed architecture permits 1000 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 - 64 Kbps)
- G.711 App II (ALaw and MuLaw with VAD)
- G.729A, G.729B (8 kbps)
- G.726 (40/32/24/16 kbps)
- G.722
- G.722.1 (24 kbps and 32 kbps Wideband)
- GSM(13.2 kbps)
- AMR (Narrow band codec - 4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps)
- AMR (Wideband codec - 6.60kbps, 8.85kbps, 12.65kbps, 14.25kbps, 15.85kbps, 18.25kbps, 19.85kbps, 23.05kbps, 23.85 kbps)
- EVRC (Rates - 1/8, 1/2 , and 1-)
- EVRCB (Rates - 1/8, ¼, ½ and 1)
- SMV (Modes - 0, 1, 2 and 3- Available if licenses are provided or owned, please call GL)
- iLBC (15.2kbps and 13.33kbps)
- SPEEX (Narrow Band)
- SPEEX (Wideband)
- iSAC (ISAC is an optional Codec. This must be purchased as a separate dongle extension)
Other VoIP Products
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. For more information click
here
Message Automation & Protocol Simulation (MAPS) - SIP Megaco, & MGCP Tester
Message Automation & Protocol Simulation (MAPS) application is designed for SIP and Megaco protocol conformance
testing.
MAPS – MGCP Tester – Designed for MGCP testing, that can simulate MGC to test Media Gateways with various types of calls. This test tool can
also be used to perform protocol conformance testing (MGCP protocol implementations) as per IETF Standard according to RFC 3435.
The MAPS - MGCP Protocol Conformance Test Suite is designed with 70+ test cases, as per RFC 3435. Test suite includes in-built scripts, which
tests the functionality of the Media Gateway for MGCP protocol valid and in-valid behavior
MAPS – Megaco Tester - 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.
The MAPS - MEGACO Conformance Suite tool is designed with 250+ test cases, as per specification of ETSI TS 102
374-2 (2004-11) standard. The test suite includes in-built scripts to process MEGACO valid and in-valid behaviors.
MAPS – SIP Tester - supports testing SIP proxy servers, Redirect servers,
Registrars, Registrants, and user agents such as SIP phones. Test cases include general messaging and call flow
scenarios for multimedia call session setup and control over IP network. Logging and pass/fail results are also reported.
The MAPS - SIP Conformance Suite tool is designed with 300+ test cases, as per SIP specification of ETSI TS
102-027-2 v4.1.1 (2006-07)) standard. Test cases verify conformance of actions such as registration, call control,
capability queries and messaging for registrars, proxies and redirect servers.
PacketScan™ - VoIP Testing Tool
GL's PacketScan™ a real-time VoIP analyzer that captures live IP traffic,
and segregates them into SIP/H323 calls and collects statistics about the calls. Hundreds of calls can be monitored in real-time
including detailed analysis of selected voice band streams. 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.
SIGTRAN Analyzer
GL's SIGTRAN protocol decode permits real-time analysis, call trace, capture, and
filter of SS7and ISDN signaling messages over IP protocol. This software is a companion to GL's award winning SS7 and ISDN protocol
analyzers.
RTP ToolBox™ - RTP Testing and Simulation Tool
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. 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.
IPNetSim™
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.
Buyer's Guide:
* Specifications are subject to change without notice.
Back to VoIP Analysis and Simulation Index Page
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![PacketGen™ Main Screen](javascript:myOpen("images/packetgen_img1.gif")){kind=link}
![PacketGen™ Status](javascript:myOpen("images/packetgen_img2.gif")){kind=link}
![PacketGen™ User Login Window](javascript:myOpen("images/packetgen_img3.gif")){kind=link}
![PacketGen™ Protocol Configuration Setup Window](javascript:myOpen("images/packetgen_img4.gif")){kind=link}
![PacketGen™ User Agent Configuration Window](javascript:myOpen("images/packetgen_img5.gif")){kind=link}
![PacketGen™ Sip Call Parameters Window](javascript:myOpen("images/packetgen_img6.gif")){kind=link}
![PacketGen™ User Accounts Window](javascript:myOpen("images/packetgen_img7.gif")){kind=link}
![PacketGen™ UAS Authentication Parameter Tab](javascript:myOpen("images/packetgen_img8.gif")){kind=link}
![PacketGen™ Phone Book Dialog](javascript:myOpen("images/packetgen_img10.gif")){kind=link}
![PacketGen™ Manual Call Reception Window](javascript:myOpen("images/packetgen_img12.gif")){kind=link}
![PacketGen™ Bulk Call Reception Window](javascript:myOpen("images/packetgen_img14.gif")){kind=link}
![PacketGen™ Bulk Call Events Window](javascript:myOpen("images/packetgen_img15.gif")){kind=link}
![PacketGen™ Bulk Call Status Window](javascript:myOpen("images/packetgen_img16.gif")){kind=link}
![PacketGen™ Buffer Management Window](javascript:myOpen("images/packetgen_img17.gif")){kind=link}
![PacketGen™ Call Records Window](javascript:myOpen("images/packetgen_img18.gif")){kind=link}
![PacketGen™ Captured Events Window](javascript:myOpen("images/packetgen_img19.gif")){kind=link}
![PacketGen™ Captured Error Events Window](javascript:myOpen("images/packetgen_img20.gif")){kind=link}
![PacketGen™ Captured Events Search Window](javascript:myOpen("images/packetgen_img21.gif")){kind=link}
![PacketGen™ Error Log Window](javascript:myOpen("images/packetgen_img22.gif")){kind=link}
