PacketGen™ - SIP Bulk Call Generator
GL offers the following SIP/RTP bulk call generators and packet analyzer:
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.
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 cores.
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.
GL provides PacketScan™ HD (PKV120) for High Density IP Traffic Analysis w/ 4x1GigE Network Monitoring and PacketScan™ FB (PKV121), a File Based IP Traffic Analyzer Server for near real-time processing of traces.
GL also offers MAPS™ SIP & MAPS™ SIP HD for advanced SIP/RTP bulk call generation with traffic. It is available as special purpose 2U network appliance that is capable of high call intensity (hundreds of calls/sec) and high volume of sustained calls (tens of thousands of simultaneous calls).
MAPS™ HD network appliance is designed to easily achieve 4 to 20,000 endpoints per server. Using a stack of multiple servers, a larger test system with 100K-200K calls (all controlled from a single Master Controller) is achievable for enterprise to carrier grade testing.
The network appliance performs signalling and traffic generation for a vast array of communication protocols covering IP and Wireless networks. It supports simulation of SIP UA, IMS SIP UE, SIP I+T, SIGTRAN, IuCS, MGCP, NCS, H.248/Megaco, Cisco SCCP (Skinny), Clear Channel (No Call Control) and provides non-reference-based voice quality using E-model (R-factor) and MOS with five mapping scales.