PacketGen™ Software Ver 3.1.4 is Now Available! Download it here
Overview | SIP Call Generation | Supported Protocols | Traffic Generation | RTP Action Scripting
RTP Options | 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. 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.
Main Features:
Supported Protocol Standards
PacketGen™ is compliant with the latest SIP specifications including RFC 2327: SDP: Session Description Protocol, RFC 3261
SIP Session Initiation Protocol as well as backwards compliant with RFC 2543.
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 the 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 - 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
Auto Signaling Action
Auto-Signaling Action 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 Action Configuration Window
Traffic Generation
Once the call is established, PacketGen™ can generate a multitude of traffic, either manually or automatically. This
traffic generation provides the mechanism to test various network conditions and responses. The traffic actions include
- Send Actions – Send GL Propriety 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
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 Options
PacketGen™ provides the user control over the jitter buffer used while recording incoming RTP streams as well as ability to
generate impairments on the outgoing RTP streams.
Receive Jitter Buffer Control
PacketGen™ uses a jitter buffer in the receive direction to handle jitter caused by transmission over a VoIP network that
can be controlled by the user. Users can set static or dynamic jitter buffer as well as configure dynamic jitter buffer minimum and
maximum lengths. Users also have the option of turning off the jitter buffer entirely.
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 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. Using PacketGen's
Command Line Interface, RTP Action scripting, along with GL's ASR Listener, FCU and VQT software, Voice Quality testing can be
completely automated.
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, 2 GHZ, 512 MB RAM, 40 GB Hard Drive, 10/100/1000 Ethernet Port, Parallel or USB port for License, Headphones
and Microphone
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
Other VoIP Products
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.
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™ UA Registration Window](javascript:myOpen("images/packetgen_img9.gif")){kind=link}
![PacketGen™ Phone Book Dialog](javascript:myOpen("images/packetgen_img10.gif")){kind=link}
![PacketGen™ Call Statistics Window](javascript:myOpen("images/packetgen_img11.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}
