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ToolBox™ - RTP Packet Testing & Simulation Tool GL's RTP ToolBox™ testing and simulation tool is designed not only to monitor RTP and RTCP packets, but also to allow users to manually create and terminate RTP sessions, independent of call-signaling protocols such as SIP, H323, MEGACO, or MGCP.
RTP-1 Annunciator The RTP-1 Monitor is a new general purpose annunciator. Designed for 19"rack installations, the RTP-1 is complete with a noise immune filter. the ten point RTP-1 features latching memory.
Advanced RTP / RTCP Toolkit Toolkit provides enhanced media transport functionality, including support for IMS, security and IPv6. The advanced RTP stack support all of the latest standards required to build VoIP solutions.
PacketScan - SIP / H323 / Megaco / MGCP / RTP / RTCP / Video Analysis PacketScan™is a powerful software tool that features real-time capture and monitoring of live IP, VoIP, and IP based Video traffic. It Segregates, captures, and collects statistics on VoIP calls (H323, SIP, MEGACO, MGCP, T.38, and others), monitors QoS (quality of service) on voice and video calls, permits analysis of adherence to protocol standards for the system under test or observation, provides graphical presentation of analysis, including ladder diagrams of protocols.
PacketGen - SIP 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. PacketGen™ is based on a distributed architecture, wherein SIP and RTP software cores can be modularly stacked in one or many PCs to create a scalable high capacity test system.
SIGTRAN - Protocol Analysis SIGTRAN protocol decoder software is a VoIP testing tool that permits real-time analysis, call trace, capture, and filtering of SS7 and ISDN signaling messages over IP. SIGTRAN uses a new connection oriented transport protocol called Stream Control Transport Protocol (SCTP). SCTP is at the same level but more efficient at transport than TCP, UDP, or RTP.
Packet and VoIP Network Monitoring System This system uses GL's Packet probe called "PacketScan™" and/ or Voice band Analyzer, PacketProbe™ to monitor packet flows in real-time within a VoIP network. All major VoIP protocols are supported including SIP, H.323, Megaco, SIGTRAN, RTP, and MGCP. A central database stores the real-time and historic data. The system also facilitates various views using a web interface known as 'PacketScanWeb™'.
RTP - Reference Test Platform RTP Test Platforms provide a Standardized Hardware & Software Test Environment Normalization of Test Platforms allows “Apples-to-Apples” Comparison RTPs minimize Hardware bottlenecks and contributions to test measurements
UDP Network Tester, UDP Flood Generator Tool UDP NetworkTester is a freeware tool which enables you to measure quality of your IP network. It can also be used to generate UDP flood or to simulate UDP DoS attack. SIP call is usually established using a SIP session with a bidirectional RTP stream. SIP and RTP protocols are based on UDP transport protocol.
TTsuite-VoiceQualityRTP The test suite for testing Voice Quality in VoIP Networks with 42 test cases to measure voice quality (PESQ, E-Model), SIP protocol to build up a connection, speech analyzing (MOS) according to E-Model (ITU-T Rec. G. 107) and PESQ (ITU-T Rec. P.862), Gathering RTP session parameters (RTPPackets and RTCPPackets, Bytes, PacketLoss and SSRC, deltaTMean, TMax and deltaTMin), Testing the voice quality of SIP signaled RTP stream (user agents/clients, networks).
DVStorIP-Gen - Evaluation and Analysis Compliance testing and stress verification of networks and IPTV monitoring equipment analysis. The DVStorIP-Gen is a high-performance, cost-effective tool for the evaluation, compliance testing and verification of networks, monitoring and equipment analysis. It supports SPTS and MPTS generation, including UDP and RTP streaming.
NTM Delta - Advanced In Situ Emissivity Independent Non-Contact Temperature Monitor The NTM Delta is the only high accuracy and repeatability pyrometer which incorporates both real-time same-point emissivity and temperature measurement. Its design is based on CI Semi?s well-known NTM500, with thousands of installations worldwide. The Delta?s rapid update rate and small footprint make it ideal for demanding applications such as RTP.
RTP - 1300 - Rapid Thermal Processing system Ecopia’s RTP (Rapid Thermal Processor) is able to do heat treatment on maximum 4inch wafer in very high speed. And, we have installed 12 pcs of hallogen lamps and it can make it possible to do heat treatment on the sample uniformly. Furthermore, it is very convenient to use and we are providing at reasonable price. Therefore, it is good to use in university and laboratory.
CT505-30 - LANforge-FIRE VoIP Call Generator The CT505-30 supports SIP VOIP calls, including RTP data-stream generation. It can make 30 calls to itself (60 VoIP soft phones) per chassis. The CT505-30 is excellent for testing whether or not an existing infrastructure can handle increased VoIP call load, and can also be used to test SIP gateways and other infrastructure.
VQmon/SA-VM - Video Quality Analysis for Videoconferencing and IPTV Specifically designed for integration in network analyzers, IPTV probes, residential gateways, IP set top boxes, videoconferencing and test equipment, VQmon/SA-VM efficiently analyzes multiple parallel packet (either RTP or MPEG Transport) video streams producing Telchemy Video Quality Metrics (TVQM).
SIP Tester™ - VoIP monitoring and testing tool StarTrinity SIP tester™ is a VoIP load testing tool which enables you to test and monitor VoIP network, SIP software or hardware. It is able to simulate and passively monitor thousands of simultaneous incoming and outgoing SIP calls with RTP media, analyze call quality and build real time reports. (call generator, call simulator)
SAM-MT 450 - ATSC & ATSC-M/H Monitoring Probe via SNMP SAM-MT 450 is an SNMP enabled ATSC and ATSC M/H monitoring probe, which covers SNMP based monitoring of all important parameters on all technical layers. It covers: RF layer, MPEG2 layer, M/H layer, IP/UDP/RTP/SDP layer, FLUTE layer, ESG layer, Audio/Video layer, Application layer (EPG, Closed Captioning), Conditional Access layer.
VQmon/HD - Video Quality Analysis for Videoconferencing and IPTV Specifically designed for integration in network analyzers, IPTV probes, residential gateways, IP set top boxes, videoconferencing and test equipment, VQmon/HD (formerly VQmon/SA-VM) efficiently analyzes multiple parallel packet (either RTP or MPEG Transport) video streams producing Telchemy Video Quality Metrics (TVQM). Video stream subjective quality estimates and detailed diagnostics for each stream can be produced either at the end of each call or at any point during a call providing much greater insight than simpler loss/jitter approaches such as MDI.
ACTS - Advanced Communication Test System As the name suggests this new server performs a variety of tests related to cable system advanced communication services, including VoIP and high speed data. The server enables a VoIP RTP test from any field test point with or without Unsolicited Grant Service (UGS), as well as high speed throughput (UDP up to 40 Mbps max). The VoIP test enables efficient service pre-installation verification with a simple, yet comprehensive set of measurement data including upstream and downstream latency, jitter, packet loss and MOS.
TTplugins – Empower Your TTworkbench With TTplugins you can easily extend the features of TTworkbench without any implementation effort. They enable your test environment to speak the language of the system you are testing. That way you can immediately start developing your test specification with full access to your system under test. Choose from a variety of existing plugins or create your own ones.
Available plugins: ASN.1 | CAN | Ethernet | FTP | GPIB | HP Quality Center | HTTP | IDL | IP | LTE_NAS | LTE RRC/S1/X2 | MOST | Protobuf | Release Management | RS232 | RTP | SCP | SNMP | SQL | SSH | TCP | Telnet | TLS | TTtwo2three | TTmex | UDP | WSDL | XSD
H.323 - Single Call Generator PacketH323™ is a PC-based real-time VoIP call generator for providing voice band testing over the H.323 VoIP Protocol. PacketH323™ can generate single calls, either manually or in an automated fashion, and generate an array of traffic once the call session is established. Call Records, Call Statistics, Protocol Events are all captured and displayed to the user in real time. PacketH323™ is fully Client/Server enabled, thus, the user can control PacketH323™ from a remote location either within a Client GUI or a Client Command Line (supports scripting applications). An optional hardware RTP can support real-time voice calls from analog phones, or fax calls from fax machines.
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RTP - Real-time Transport Protocol
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