PacketGen™ SIP Bulk Call Generator

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818 West Diamond Avenue - Third Floor, Gaithersburg, MD 20878 Phone: (301) 670-4784 Fax: (301) 670-9187 Email: [email protected]

Website: http://www.gl.com 1

PacketGen™ SIP Bulk Call Generator

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PacketGen™ SIP Bulk Call Generator

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PacketGen™ Application

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.

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Applications

Manual and Bulk Call generation

◦ Supports SIP, RTP, RTCP, with full SIP Functionality - Call Forwarding, Call Hold, Call Transfer, etc

◦ Various traffic generation capabilities – voice, tones, digits, and more

◦ Up to 2000 concurrent calls with full duplex RTP per i7 PC running single SIP/RTP Software Core;

◦ Distributed architecture allows achieving higher call density by interconnecting more number of systems with SIP and RTP software cores

◦ Multiple probes with single GUI at central site

◦ Generate test calls to IP Phone, ATA, PSTN, Wi-Fi, Cellular

◦ Software defined architecture

Stress Testing

◦ Generate 200 Bi-direction RTP streams per SIP-RTP pair (Stackable)

Load router DSPs, Load network pipe

◦ Generate SIP call traffic – 250 INVITES per sec

Find gateway access limitations

◦ Generate proxy registration load

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• Voice Quality

▪ Automatically and manually play/record test voice files in a synchronous manner

▪ Automatically transfer degraded voice files to GL VQT analysis, providing ITU-standard MOS (PESQ, PESQ WB,

PAMS, & PSQM)

▪ Results rated as excellent, good, fair, poor

• Regression and Acceptance Testing –

• CLI allows users to create traffic using their own test software

• Can be used for OEM testing

• Field acceptance test, Traffic generation

• Matrix Testing –

• Distributed network call agent to agent over customer networks

• One-Way transmission – send bi-directional traffic to verify continuity. Enhanced capability with action scripting

• Service level agreement verification

• Others - Protocol Compliance, Codec Compatibility, & Voiceband Testing

Applications…

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Capacity • Distributed architecture for SIP and RTP systems provide high call rates and media streams.

• Provides high density performance; PacketGen™ can generate 2000 simultaneous calls on an

core 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.

Call

Generation

• Full SIP Functionality - Registration, Call Forwarding, Call Hold, Call Transfer, Authentication,

• 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 and

Digit Generation and Detection parameters (power, on/off, pause, and amplitude).

Key Features

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Key Features…

Key features7

Traffic

Handling• Powerful scripting capability for RTP traffic generation.

• 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.

• 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 scenario

http://www.gl.com/testing-interactive-voice-response-systems.html

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Key features

Supported

Codecs

• G.729ab, G.726, G.711 (A-law, µ-law), G.711 Application II (A-law and µ-law with VAD)

• GSM, GSM EFR, GSM HR, EVRC, EVRCB

• iLBC (15.2kbps and 13.33kbps)

• AMR , SPEEX (Narrow Band and Wideband)

• G.722.1, G.722 , H.263

• SMV (licenses required)

Remote

Access

• Remote access capability using GUI or command line interface or through Remote Desktop

Reports • Provides statistics, events and call records

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SIP Setup and Configuration

User Identity Login

User Agent Configuration

Manual and Bulk Call Generation / Reception

Send / Receive Traffic

SIP Call Generation

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SIP Setup and Configuration• Controls the foundation of the desired test environment

• Configures multiple SIP and RTP instances on a local system and/or remote systems

Sip Call Generation

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User Identity Login

• Input the Identity as well as the name / IP address of the SIP Core that forms the call agent

• Identity for each server should be unique

Sip Call Generation

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Protocol Configuration Setup

• Displays all the call agents to which the user has logged in.

• Configures the SIP parameters and users for each call agent.

SIP Call Generation…

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User Agent Configuration – SIP Parameters• Each call agent can be configured with ‘1 to infinite’ number of users, with additional information such as registrar, proxy, codec,

NAT (Network Address Translation), SIP header, SDP headers, and authentication

Sip Call Generation

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User Agent Configuration – Media Parameters• Indicate the media capabilities of the User Agent

• Used to negotiate media characteristics of the call during call establishment

Sip Call Generation

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User Agent Configuration – Extra Headers

• Allows user to configure many non-critical headers

• User can add both SIP headers as well as SDP headers per User Agent

Sil Call Generation

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User Agent Quick Configuration• Facilitates to register a single or a bunch of user agents simultaneously

• Provides flexible configuration options like Registrar server address, Address of Record, Expiry time etc

• A quick configuration utility helps to configure hundreds of registrations easily


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User Agent Registration• Facilitates to register a single or a bunch of user agents simultaneously

• Provides flexible configuration options like Registrar server address, Address of Record, Expiry time etc

• A quick configuration utility helps to configure hundreds of registrations easily


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Manual and Bulk Call Generation

• Supports both manual and bulk call generation, with complete flexibility on each individual call session

such as –

• Quick configuration utility

• Status of each configured session

• Traffic generation for QOS measurements

• Call processing options including hold and call transfer


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Manual Generation

Sip Call Generation

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Bulk Call Generation

Sip Call Generation

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Auto Traffic & Signaling Actions

• Auto-Action feature provides a quick and easy method to configure signaling as well as traffic actions, once the call

session is established

• RTP Traffic options include transmit / record voice, generate / detect tones, digits and noise and send / receive fax

• Supports generation of impairments on outgoing RTP streams – Latency, Packet Loss, Packet Effects

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Auto Signaling Actions

• Configure signaling actions to be performed automatically as soon as the call session is established

• Signaling options include call transfer, call reject (user-defined error), hold and re-direct

Auto Traffic & Signaling Actions…

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Auto Traffic Actions


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• Send Actions – Send GL Propriety voice files, DTMF or MF Digits (In-band or Out-Band), user-defined single/ dual frequency

tones, 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.

Traffic Handling - can generate a multitude of traffic, either manually or automatically


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Auto Traffic Actions


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RTP Impairment Generation

• Various impairments can be configured on outgoing RTP streams

• Categories of impairments can be generated – latency, packet loss, and packet effects


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Other Run-Time Parameters

• Call Agent (SipCore) can be configured with the SIP protocol timers T1 and T2, Reliable Provisional Responses, Early Media

Actions, Call Setup Behavior, and Progress timer

SIP Options


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Other Run-Time Parameters…

• Call Agent configuration with the RTP Source Information, Handling Packetization time, Send Outband info, and Rx Jitter Buffer

RTP Options


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Other Run-Time Parameters…

• Authentication of incoming calls, Digit Generation and Detection parameters (power, on/off, pause, and amplitude), putting traffic

actions put under hold, Terminate original call, and handling automatic Re-Registration

Advanced Options


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Scripting Traffic Actions

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• Powerful automated scripting capability to control RTP traffic

• Simple user interface to create scripts

• Conditional statements , stack multiple actions

• Create/test IVR kind of applications

Scripting Traffic Actions…

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• Provides detailed statistics for each user agent, sip core as well as for the entire system

• Call Statistics window provide detailed call wise statistics per sip core

• 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 sip core

• Provides various events screens such as call records, captured events, captured error events, tone / digit detection,

bulk call events, events search, and error log

Statistics and Events

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Call Statistics

Statistics and Events…

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Captured Events


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Bulk Call Generation Status• Graphical representation of the call status of each bulk call

• Displays the call status in various colors


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Command Line Interface

Operate PacketGen™ from a DOS

based console

Allows easy integration of

PacketGen™ into other

applications for customization

Supports all the functionalities of

the GUI, except the configuration

functions

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Audio File Converter Utility (Audio FCU)

• GL Audio File Converter Utility (AFCU) 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)

PacketGen™ Utilities

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Audio Streaming Utility• Play the selected calls audio to the local speaker

• Client-Server application used to stream and playback 16 bit raw linear files

PacketGen™ Utilities

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RTCP-XR Ini File

• Capable to handle signaling negotiation (as per RFC 3611)

for RTCP-XR attributes through SDP.

• PacketGen™ handles signaling negotiation according to the

settings done in “RTCP_XRConfig.ini” file.

• User customizable .ini file depending upon requirements

PacketGen™

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RTPConfig Ini File

• Configures EVRC packing format

• Provides options to disable/enable RTCP packet transmission, and Digit detection qualification time and power.

• INI file is read once by the RtpCore on startup and will be applicable as long as RtpCore runs.

PacketGen™

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THANK YOU

Please visit http://www.gl.com/packetgen.html for more details

http://www.gl.com/packetgen.html

PacketGen™ SIP Bulk Call Generator

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