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VoIP
An application that encodes and digitizes a voice signal, converts the signal to data packets, and transports the packets over a data network running Internet protocol (IP). (www.intel.com)
- InterWorking Labs, Inc.
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Test SIP
VoIP, Video, and IPTV products must withstand and continue to function when facing all the conditions that occur on the Internet. These include attacks from hackers exploiting vulnerabilities in security and protocols as well as the common phenomenon of congestion, delay, jitter, drops, and so on. VoIP, Video, and IPTV products must demonstrate robustness and resiliency in the face of unusual and/or illegal conditions.
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VOIP
Macom Technology Solutions Holdings Inc.
MACOM’s family of VoIP processing devices, in concert with carrier-grade media stream processing software, delivers a complete, integrated VoIP media, data and fax, signaling and control processing solution. Combined with dual-core ARMs, powerful DSP cores, dual Gigabit Ethernet with flexible packet processing engine and a rich set of interfaces, the processors support 8 to 160 channels of packet-TDM and packet-packet operations, with an integrated host controller supporting control, signaling and other application specific functions. Major applications include next-generation network (NGN) POTS/ISDN line cards focused on class 5 replacement, xPON MDUs, IP-DSLAMS, MSAN, and IP-PBX, and Unified Communications.
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VoIP Testing
VQuad™
Testing on the extreme end-point of a VoIP softphone network has become a priority for many service providers. This method of testing represents what an end user is actually experiencing. Voice quality testing within VoIP networks is growing; a major concern is testing VoIP softphones that will be used with VoIP networks. GL’s VQuad™ with VoIP option along with Voice Quality Testing (VQT PESQ) software provides the ability to perform manual or automated Wideband audio tests using HD codecs on the VoIP network.
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VoIP IAD
Is a multi-functional networking device (Integrated Access Device or multiservice access gateway) that provides a WAN-access interface in addition to IP routing and a voice gateway services (see What is a VoIP Gateway?). Installed on the customer premise, the VoIP IAD is classified as customer-premise equipment (CPE). While there is some debate on the topic, a VoIP IAD is generally considered to be more efficient and cost-effective than a multi-unit CPE solution for most business-class IP-telephony applications
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VoIP Adapter
A VoIP adapter is a protocol converter (bridge) between analog phones and the digital (IP) network that carries voice calls.
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VoIP Measurement Technology
Nextragen Solutions has developed a measurement method that analyzes the entire End2End network infrastructure across all data streams. Quality assurance begins with us even before a VoIP telephone system is put into operation.
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VoIP & Network Quality Manager
VoIP monitoring software designed for deep critical call QoS metrics and WAN performance insights
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VoIP Monitoring and Testing Tool
SIP Tester™
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.
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VoIP Infrastructure Monitoring and Maintenance
NetVoice®
NIKSUN NetVoice® is the essential tool to monitor and maintain optimal network operation of your VoIP infrastructure. The NIKSUN solution monitors the network and alerts the administrator when abnormal network behavior or deterioration occurs, providing essential management and trending reports (periodic or on-demand) to plan for future network needs.
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Handheld VoIP Service Tester
925VST
THE 925VST is a powerful handheld voice over packet (VoP)service qualification tool for field deployment and troubleshootingpersonnel who need a rapid and reliable installation tool.
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IP / VoIP Analysis & Simulation
GL offers high density Ethernet/IP tester variant which includes mTOP™ 1U/2U rack mount enclosures within which PacketExpert™ portable USB units are stacked to provide high density GigE ports form factor solution for testing GigE switches, routers and network conditions.
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ADSL / 2 / 2+ Handheld VoIP Tester
QuickTest 500
*Rapid Testing. The QuickTest 500 ADSL2+ Tester's automatic Service Verification Test Suite starts as soon as you switch on (results are typically obtained in less than 3 minutes)*Simple to use. Virtually no product training required with QuickTest 500 ADSL Tester*Configurable Pass / Fail Indicators on the LCD display and clearly presented in multi-colour LEDs on the front panel*No need to get a laptop out*Save and name up to 16 test results at any one time*Line Voltage LED indicates poor, missing or excessive Line Voltage*ADSL/2/2+ Test LEDs to indicate if ADSL Showtime achieved*In Service LED indicates problems with PPP or IP layer*LEDs to indicate problems with the Trained Line Rate, poor Noise Margin or excessive Line Attenuation
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Service Assurance Remote Probe for IPTV, OTT, VoiP & Internet
µNET
The μNET is a palm size probe for testing and monitoring Triple-Play services (IPTV, VoIP, Internet) at the customer set top box or the last mile concentrator (ie.GPON mux). Multiple μNET probes can be used throughout the IPTV network on a temporary or permanent basis. Test results are sent to the external FTP server. All tests are performed in a cycle set up by the user.
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UDP Test Tool, Flood Generator, VoIP Readiness Test Tool
The Multiprotocol Network Tester is a freeware, open source 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. UDP uses a simple transmission model without implicit reliability, ordering and data integrity. Each single UDP packet is transferred independently. The quality of a SIP call depends on delays and loss of IP packets in a network. Long delays lead to large RTP jitter and bad sound quality of a SIP call.
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Voice Monitoring
VOIP Echo
*The Voice Band Analyzer (VBA) is an analysis tool for monitoring the quality of voice band traffic (including hybrid echo) over VoIP, TDM and wireless networks*VBA works in conjunction with GL's TDM, Packet, and Wireless non-intrusive capture products, such as T1 and E1 Call Capture and Analysis, VoIP PacketScan&trade, and GSM, CDMA, and 3G Call Capture Products. VBA instantly detects voice file captures and with appropriately installed analysis modules, monitors speech levels, noise levels, and hybrid echo. Other analysis modules such as ITU-T P.561, P.562, P.563, fax and modem analysis, and many others can be hosted as plug-ins.*The VBA application can also operate on pre-captured files. It supports A-Law, µ-Law, 16-bit PCM (Intel), 16-bit PCM (Motorola) and MS Wave file formats.*Required software PKV100 and VBA032 at each probe location and PKV170 PacketScanWEB - Network Surveillance Software*Required hardware - 1U rack PC with sufficient hard disk for capturing over 10,000 calls (80 GB HD) for each VBA colocated with a probe*Additional hardware - PC for PKV100 PacketScan™ Software
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Conformance Testing Software
Clarinet-TTCN VOIP ETS
Protocol: the ITU-T H323 and H225 define VOIP call signaling protocol for "Packet based multimedia communication systems".IUT: H323/H225 Terminal, gatekeeper and gatewayATS: Based on first ETSI Drafts TS 101 804-2 and 101 804-3 rewritten by ACACIA
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NetIQ Vivinet Diagnostics
NetIQ Vivinet Diagnostics quickly pinpoints call quality problems in Voice over IP (VoIP) networks and explains why you are experiencing reduced call quality. Vivinet Diagnostics reduces the time needed to resolve voice quality issues and lessens the skills required for VoIP troubleshooting, both before and after deployment of VoIP.
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IP Telephony
PLANET integrated HD VoIP solution consisting of IP PBX system, VoIP gateway, VoIP phone, IP intercom and IP paging, and analog telephone adapter features wideband-compliant, crisp and clear sound quality. PLANET IP telephony products support enterprises of all kinds to expand offices and branches in new locations.
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WAN Emulation
Net.Storm is able to simulate any network condition to check how tolerant your services, protocols and devices are when quality / capacity degradations occur. It replicates accurately combined effects such as packet delay, errors, loss, bandwidth variations, traffic shaping and traffic policing.VoIP.Master is an advanced turn-up solution for VoIP allowing technicians to deploy VoIP thanks to its emulation capabilities that permit users to connect VoIP.Master to SIP trunks and networks emulating an IPBX, making multiple VoIP calls ensuring the trunk is operational and performing to pre-agreed SLAs.
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Enterprise Session Border Controller
eSBC
Session Initiation Protocol (SIP) is the communication technology employed to set up and tear down VoIP phone calls
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GL Products Online & Interactive Demos
*NetSurveyorWeb™ is a centralized VoIP network monitoring system. A number of GL's PacketScan™ probes and/or,PacketProbes™ are deployed in remote locations across the network to passively monitor VoIP traffic. Whenever a monitored call completes, the scan/probe calculates a variety of quality metrics (MOS, loss, delay, jitter, etc) and sends the metrics to the central Oracle database for storage.**The PacketScan™ is a feature-robust Windows® based software tool that captures and monitors live IP traffic. In the VoIP world, it can monitor and measure SIP, H323, Megaco, MGCP, T.38 and video calls. In addition to quality metricsPacketScan™ also captures signaling information and sends that to the NetSurveyorWeb™ database as well.**GL's PacketProbe™ is an advanced CPE(Customer Premises Equipment) based VoIP monitoring reporting and diagnostic appliance, stemming from GL's suite of market leading voice analysis tools.
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Advanced Communication Test System
ACTS
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.
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RJ11, RJ14, RJ22 & RJ25 Connectors
Our high-quality RJ11, RJ14, RJ22 & RJ25 connector products are the ideal components for telecom applications like standard telephone systems, modems, fax machines and VoIP systems.
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4 Port T1/E1/PRI Bulk Call Generator
BCG-4PRI
This unit has four PRI ports allowing to generate real calls to a T1/E1/PRI connected device for the stress-test. It can be used to generate call volume to VoIP geteways, PBXs, Fax servers, RAS serves.
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8 Port T1/E1/PRI Bulk Call Generator
BCG-8PRI
This unit has eight PRI ports allowing to generate real calls to a T1/E1/PRI connected device for the stress-test. It can be used to generate call volume to VoIP geteways, PBXs, Fax servers, RAS serves.
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Bulk Call Generator and network simulator
Emutel Harmony
The Emutel Harmony Bulk Call Generator and network simulator is a highly efficient, cost-effective and flexible tool, invaluable for use in the development, performance verification and pre-deployment testing of many types of VoIP, ISDN and analog equipment.
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FAX Adapters
The VoIP ATA for FAX is a powerful, cost-effective solution for customers who need to securely send and receive faxes using a traditional, analog fax machine often needed for legal and privacy compliance such as HIPPA or FERPA.
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Utel Insight
Utel Insight unlocks real-time network analysis and insights for telecoms operators, enabling consistent, superior network quality and first-class customer experience by presenting information about network traffic, across mobile, fixed and VoIP domains.
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Media Gateway
A Media Gateway is a networking device that provides valuable and essential functions for organizations, service providers, and network operators. When interconnecting traditional time-division-multiplexed (TDM) telephony systems with modern voice-over-internet protocol (VoIP) systems, a VoIP media gateway works by providing protocol conversion (translation) and interoperability between the two network technologies. Also known as an IP media gateway or SIP media gateway, the device converts a traditional analog plain-old-telephone-system (POTS) voice signal or related digital telephony signals such as T1, DS3, STM1 or OC3 into packets for transmission over a data network such as the Internet, private IP Network or a corporate LAN. When the legacy system is based on digital integrated services digital network (ISDN) technology, the device may be called a digital VoIP gateway or ISDN media gateway. Some manufacturers combine gateway functionality with WAN-access (internet-access) capability into a single device or network appliance. Such a dual-purpose device may be known as an access media gateway.
