PacketGen™ SIP Bulk Call Generator 818 West Diamond Avenue - Third Floor, Gaithersburg, MD 20878 Phone: (301) 670-4784 Fax: (301) 670-9187 Email: info@gl.com Website: http://www.gl.com 1 1
PacketGen™ SIP Bulk Call Generator 2
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. 3
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 4
Applications… • 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 • 5
Key Features • 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 Full SIP Functionality - Registration, Call Forwarding, Call Hold, Call Transfer, Authentication, Generation • 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). 6
7 Key features Traffic • Powerful scripting capability for RTP traffic generation. Handling • 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. Key Features… • 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 7
Key features • Supported G.729ab, G.726, G.711 (A-law, µ-law), G.711 Application II (A-law and µ-law with VAD) Codecs • 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 Remote access capability using GUI or command line interface or through Remote Desktop Access • Reports Provides statistics, events and call records 8
SIP Call Generation SIP Setup and Configuration User Identity Login User Agent Configuration Manual and Bulk Call Generation / Reception Send / Receive Traffic 9
Sip Call Generation 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 10
Sip Call Generation 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 11
SIP Call Generation… 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. 12
Sip Call Generation 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 13
Sip Call Generation User Agent Configuration – Media Parameters • Indicate the media capabilities of the User Agent • Used to negotiate media characteristics of the call during call establishment 14
Sil Call Generation 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 15
SIP Call Generation… 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 16
SIP Call Generation… 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 17
SIP Call Generation… 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 18
Sip Call Generation Manual Generation 19
Sip Call Generation Bulk Call Generation 20
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 • 21
Auto Traffic & Signaling Actions… 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 22
Auto Traffic & Signaling Actions… Auto Traffic Actions 23
Auto Traffic & Signaling Actions… Traffic Handling - can generate a multitude of traffic, either manually or automatically 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. • 24
Auto Traffic & Signaling Actions… Auto Traffic Actions 25
Auto Traffic & Signaling Actions RTP Impairment Generation • Various impairments can be configured on outgoing RTP streams Categories of impairments can be generated – latency, packet loss, and packet effects • 26
Auto Traffic & Signaling Actions Other Run-Time Parameters SIP Options • 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 27
Auto Traffic & Signaling Actions Other Run- Time Parameters… RTP Options • Call Agent configuration with the RTP Source Information, Handling Packetization time, Send Outband info, and Rx Jitter Buffer 28
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