SIP Bulk Call Generator
PacketGen™ Software Ver 14.8.21 is Now Available! Download it here
SIP Call Generation |
Supported Protocols |
Traffic Generation |
RTP Action Scripting
Statistics and Events |
Command Line Interface |
Voice Quality Testing using PacketGen™
Other VoIP Products |
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, with SIP and RTP software cores modularly stacked
in one or many PCs to create a scalable high capacity test system capable of generating more than 2000 simultaneous calls.
GL's PacketGen™ breaks ground with high density performance (2000 Simultaneous Calls)
PacketGen™ on an i7 PC can support 2000 simultaneous calls with, both SIP and RTP generation. This performance number is associated with using the G.711 codec, while 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
PacketGen™ can be used to test basic functionality and verify proper protocol implementation in SIP based equipment such as SIP phones and Network Servers, Proxy Servers, Registrar Servers, and PSTN and Media Gateways.
- 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 2000 simultaneous calls on an i7 PC. Higher density is also achievable using multiple systems
- Up to 20 SipCores can be run on the same PC or on multiple PC systems. All 20 SipCores can be remotely controlled from a single system.
- Full SIP Functionality - Registration, Call Forwarding, Call Hold, Call Transfer, Authentication, etc
- Manual and Bulk Calling capabilities with complete flexibility on each call session
- RFC 3261 compliant, RFC 2833 digit generation/detection
- Generates both SIP signaling & RTP traffic (voice, fax, digits, tones)
- 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).
- 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
- Automate the IVR testing process - call establishment and traffic generation / detection process through scripts
- Monitoring IVR System for voice and data quality
- Send/Record voice files on any (or all) RTP sessions. Recorded voice files can be sent to VQT software for voice quality analysis (requires additional license)
- 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
- Allows user to create early media (media can be transferred or received even before call is established) scenario
- Remote access capability using GUI or command line interface or through Remote Desktop
- Provides statistics, events and call records
- GL’s Audio File Conversion Utility (Audio FCU) is used in conjunction with GL’s packet products to automatically convert any voice file with different codec, into *.glw file format and vice versa, during send and record voice functions.
- GL’s Audio Stream Utility is used to playback the selected calls from a remote computer to the speaker on the local client PC.
- Testing for Stress
- Manual and Bulk Call Generation
- Voice Quality Analysis, Digits, Tones, and Voice Files
- Regression and Acceptance Testing
- Matrix Testing
- Protocol Compliance, Codec Compatibility Testing
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
User Agent (UA) Configuration Parameters
Various parameters can be configured. They are grouped into 4 tabs:
- SIP Parameters – Allows user to set certain SIP/SDP headers for outgoing messages. User can set UA Name, Host Name, Port, Sip Server Address, NAT option, and multiple contact entries for each UA.
- Media Parameters – Allows user to set the RTP media parameters (RTP) for the User Agent. These parameters indicate the media capabilities of the User Agent. These are used to negotiate media characteristics of the call during call establishment.
- Extra Headers – Configures extra SIP/SDP headers to be used for each User Agent. These headers are non-critical headers, and will be included as is, in the appropriate messages sent for this User Agent.
- UAC Authentication – Configures the user authentication information required for UAC simulation.
Screen Shot of User Agent Configuration
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
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
Reliable Provisional Responses (RPR)
The ability to send "reliable provisional responses" and start early media actions. We have two options in Reliable Provisional Responses Viz: Required and Supported. Below diagram shows a SIP call flow with RPR's and early media.
Screen Shot of Reliable Provisional Responses (RPR)
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-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
Audio Stream Utility
The existing "Playback" feature is used to play the selected call to speaker on the local computer (SIP/RTP core). To allow these calls to be heard from remote
systems, GL has introduced Audio Stream Utility with PacketGen™. This utility automatically streams the voice of a selected call to a speaker on a remote
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.
Script Editor has been enhanced with a Monitor Tone option that helps the user to detect single/dual frequency tones. The NAT Address option has also been
modified so that the PacketGen™ is configured to send only NAT Address in a contact header while sending any request.
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 -
- Packet Loss
- Packet Effects
Voice Quality Testing using PacketGen™
PacketGen™ can be used to establish calls and send/record voice files through the IP network. These voice files are then analyzed using GL’s Voice Quality
Testing (VQT) application as per ITU algorithms. 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.
Audio File Converter Utility (AFCU)
PacketGen™ now transmits and records voice files using a GL proprietary file format (.glw). The accompanying GL Audio File Converter Utility (AFC) will
automatically convert any voice file, encoded as G.711, G.729ab, G.726, or GSM, into *.glw file format and vice versa. This allows the ability to send/receive
voice files at a higher density with multiple codecs (the file is predefined with the desired codec). It also allows for Discontinuous Transmission/Reception.
The Auto FCU (part of AFCU) is generally used in conjunction with GL's VQT application and converts degraded voice files from their native codec format to
a standard format used by VQT. The Command line interface (CLI) in AFCU allows the users to load, start, and stop Auto FCU configurations, convert glw to
pcm and vice-versa using the commands
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
PacketGen™'s CLI Script: Registration of SIP setup
Configuring parameters using .INI files
PacketGen™ provides user customizable .ini files to suit the testing requirements. The .INI file is read once by the RtpCore on startup and will be applicable as long as RtpCore runs.
RTCP_XRConfig.ini - PacketGen™ handles signaling negotiation (as per RFC 3611) for RTCP-XR attributes through SDP in "RTCP_XRConfig.ini" file.
- Allows to configure EVRC packing format
- Provides options to disbale/enable RTCP packet transmission, and Digit detection qualification time and power.
Screen Shot of RTCP_XRConfig.ini
Screen Shot of Command Line Interface
- Windows® XP (32 bit and 64 bit)/ Vista (32 bit)/ 7 (32 bit and 64 bit) OS,
- 2 GHZ, 512 MB RAM, 40 GB Hard Drive,
- 10/100/1000 Ethernet Port,
Parallel or USB port for Dongle, Headphones and Microphone
RTP Software Core
- Multiple SIP Cores in a distributed architecture permits capacity scaling
- 2000 simultaneous calls on an i7 PC running single SIP/RTP Software Core
- Distributed architecture permits 2000 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 (5 bit 40 kbps/4 bit 32 kbps/3 bit 24 kbps/2 bit 16 kbps)
- G726 (40/32/24/16 kbps with VAD)
- 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)
(requires additional license)
- 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-), EVRC0
(requires additional license)
- EVRCB (Rates - 1/8, ¼, ½ and 1), EVRCB0
(requires additional license)
- 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)
- GSM HR (rate – 5.6kbps, packet time multiples of 20msec.)
- GSM EFR (rate - 12.2kbps, packet time fixed at 20msec.)
* Specifications are subject to change without notice.
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