GL conveys the availability of its Product PacketGen™ - SIP Bulk Call Generator

Share Article

GL Communications Inc. conveyed today the availability of its product PacketGen™ - SIP Call Generator Software.

PacketGen™ is a PC-based real-time VoIP bulk call generator

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.

Communications Inc. a leader in providing PC-based test, analysis and simulation products and consulting services to the worldwide telecommunications industry, conveyed today the availability of its product PacketGen™ - SIP Call Generator Software.

Speaking to media persons, Mr. Jagdish Vadalia Senior Manager said, ”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 1000 simultaneous calls.”

He added, “GL's PacketGen™ breaks ground with high density performance on a Duo Quad Core PC, it 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.”

Mr.Vadalia further added, “PacketGen™'s distributed architecture allows achieving higher call density by interconnecting more number of systems with SIP and RTP software cores. And it 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.”

Some of the Important Features:

·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).
·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 systems
·Up to 20 SipCores can be run on the same PC or a multiple PC system. All 20 SipCores can be remotely controlled from a single system
·RFC 3261 compliant, RFC 2833 digit generation/detection
·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).
·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 or through Remote Desktop
·Provides statistics, events and call records
·Supported on Windows® XP (32 bit and 64 bit) / Vista (32 bit) / 7 (32 bit and 64 bit) OS


·Stress Testing
·Manual and Bulk Call Generation
·Voice Quality Analysis
·Regression and Acceptance Testing
·Matrix Testing
·Protocol Compliance, Codec Compatibility Testing

About GL Communications Inc.,

Founded in 1986, GL Communications Inc. is a leading supplier of test, monitoring, and analysis equipment for TDM, Wireless, IP and VoIP networks. Unlike conventional test equipment, GL's test platforms provide visualization, capture, storage, and convenient features like portability, remotability, and scripting.

GL’s TDM Analysis & Emulation line of products includes T1, E1, T3, E3, OC-3, OC-12, STM-1, STM-4, analog four-wire, and analog two-wire interface cards, external portable pods, and complete system solutions. Capabilities include voiceband traffic analysis and emulation across all traffic types (voice, digits, tones, fax, modem), all protocols (ISDN, SS7, GR-303, Frame Relay, HDLC, V5.X, ATM, GSM, GPRS, LTE, etc.), and with capacities up to thousands of channels. Our newest products provide astonishing capacity and capture capability up to and including gigabit speeds.

GL’s VoIP and IP products generate / analyze thousands of calls and traffic simultaneously with traffic types such as frames, packets, voice files, digits, video, tones, noise, and fax. Almost all codecs are supported including G.711, G.729, AMR, EVRC-A,B,C, GSM, iSAC, and many more. Additional features include visual analysis, real-time listening, and recording. The product line also includes Ethernet / IP Testing capability that simulates and checks frame transport and throughput parameters of Ethernet and IP networks, including delay, errors and other impairments.

GL's Voice Quality Testing (VQT) product line complements all of GL's products. Using ITU-standard algorithms (PAMS, PSQM, and PESQ), GL's VQT provides a widely accepted solution for assessing voice quality in the telecom industry. Voice Quality Testing across multiple networks (T1, E1, T3, E3, OC-3, OC-12, VoIP, Wireless, and Landline) is available.

GL’s Wireless Products perform protocol analysis and voice quality assessment on GSM, CDMA, UMTS, and LTE networks. Connections can be made to any wireless phone with automated call control, GPS mapping and real-time signal measurements.

GL’s Echo Canceller testing solutions provide the broadest range of simulation and analysis, including line and acoustic echo. GL’s compliance testing per G.168. G.167 and P.340 across TDM, IP, VoIP and Wireless networks is widely accepted in the industry.

GL’s wireless VQT solutions help assessing impairments to voice quality such as poor mobile phone quality, voice compression and decompression algorithms, delay, loss and gain in speech levels, noise, acoustic and landline echo, and other distortions are easily assessed and accurately measured.

GL’s Handheld data testers can test a wide variety of communications facilities and equipment including T1, fractional T1, E1, fractional E1, T3 and E3 modems, multiplexers, CSU, DSUs, T1 CSUs, DTUs, NTUs and TIUs and more. The testers provide convenience, economy, and portability for almost any interface, including RS232, RS-422, RS-530, X.21, T1, E1, T3, E3, and many others.

GL’s Network Surveillance and Monitoring products include Probes for TDM, IP, VoIP, ATM, and Wireless networks. An open standards based approach provides a scalable, feature rich, real-time access to network characteristics. Centralized or distributed access, efficient transport and database loading allow compatibility with 3rd party and standards based monitoring systems.

Share article on social media or email:

View article via:

Pdf Print

Contact Author

Mr.Shelley Sharma

Gururaj S D