High Quality Content by WIKIPEDIA articles! Asterisk is a software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. Its name comes from the asterisk symbol. Asterisk is released under a dual license model, using the GNU General Public License (GPL) as a free software license and a proprietary software license to permit licensees to distribute proprietary, unpublished system components.
High Quality Content by WIKIPEDIA articles! TNZ Group Ltd is a New Zealand based telecommunications supplier founded in 2001. It was formed as a new corporate entity to feed the growing messaging requirements of today's leading businesses, specializing in building solutions to suit customer requests. Working with Fax, SMS, Email, PSTN and VoIP, TNZ Group Ltd works to take these technologies and integrate them with the Internet to create Unified messaging.It largely operates as a telecommunications wholesaler and supplier, offering its messaging technologies to resellers. TNZ Group Ltd is privately owned and operated from Auckland, New Zealand.
Existing wireless networks enable different important applications over IP based network such as public internet and Voice over Internet protocol is one of the important applications which have become a possible alternative to public switched telephone network (PSTN). This project investigates one of the time sensitive data applications on WiMAX i.e. VoIP. I carried on WiMAX because WiMAX networks provide advance features and protocols to support the Quality of service (QoS). This project also indicates about the impact of load and mobility on the voice calls. This book provides details about the Voice Over Internet Protocol (VoIP), Which includes types of VoIP calls, VoIP System and Components, VoIP Protocols, VoIP codecs and VoIP Quality of Service (QoS). It also provides detailed study of WiMAX including WiMAX Protocol Layer and WiMAX Quality Of Service(QoS)
High Quality Content by WIKIPEDIA articles! H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK and NetMeeting and is widely deployed worldwide by service providers and enterprises for both voice and video services over Internet Protocol (IP) networks. It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over Integrated Services Digital Network (ISDN), Public Switched Telephone Network (PSTN) or Signaling System 7 (SS7), and 3G mobile networks. H.323 Call Signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN.
High Quality Content by WIKIPEDIA articles! VoIP User is a community driven and financed SIP based VoIP network. The projects aim is to introduce people to the concept of VoIP by allowing members to experiment with SIP and IAX2 devices.The VoIP User network was designed to operate within a community environment and therefore differs substantially from most other VoIP networks. The main highlighted difference being that users can call PSTN phone numbers through VoIP User's PSTN gateway without incurring any call charges. The way that VoIP User is funded is noteworthy: Calls into VoIP User?s numbers generate a small amount of per minute revenue (the ?termination charge?), and this money goes into a community account or ?pot?.
Telecommunications technologies are evolving at a rapid pace. The old Public Switched Telephone Network (PSTN) is being replaced with wireless and voice over IP (VoIP) systems. This requires the service providers to offer their services on competitive prices, on one hand, and to ensure the interoperability of their services over hetero- geneous networks on the other. Added to this is the challenge of keeping up with the expectations of the clientele as regards quality of service (QoS). Thus to enable the successful deployment and functioning of a telecommunications network, it is equally important to estimate the speech quality as it may be perceived by the humans. The goal of this research was to derive superior non-intrusive speech quality esti- mation models. Model superiority was sought in a multi-objective sense: 1) enhance- ment of prediction accuracy of the derived models as compared to the previous ones. 2) model simplicity or parsimony was desired as it may enhance the computational efficiency. In this research this is achieved by employing a novel approach based on Genetic Programming (GP).
High Quality Content by WIKIPEDIA articles! Teletraffic engineering is the application of traffic engineering theory to telecommunications. Teletraffic engineers use their basic knowledge of statistics including Queueing theory, the nature of traffic, their practical models, their measurements and simulations to make predictions and to plan telecommunication networks at minimum total cost. These tools and basic knowledge help provide reliable service at lower cost. Because the approach is so different to different networks, the networks are handled separately here: the PSTN, broadband networks, mobile networks, and networks where the possibility of traffic being heavy is more frequent than anticipated.
High Quality Content by WIKIPEDIA articles! A voice browser is a web browser that presents an interactive voice user interface to the user. In addition, it typically provides an interface to the PSTN or a PBX. Just as a visual web browser works with HTML pages, a voice browser operates on pages that specify voice dialogues. Typically these pages are written in VoiceXML, the W3C's standard voice dialog markup language, but other proprietary voice dialogue languages remain in use. A voice browser presents information aurally, using pre-recorded audio file playback or using text-to-speech software to render textual information as audio. A voice browser obtains information using speech recognition and keypad entry (e.g., DTMF detection).
- 16, 24 or 32 FXS or FXO—Simultaneous voice or fax calls on all ports. Advanced local call switching.- Full SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem bypass, DTMF relay. Codecs G.729, G.723 etc.- Outstanding Interoperability—Interoperable for voice and T.38 fax with leading SIP service providers, soft-switch vendors and Asterisk IP-PBXThe SmartNode 4300 VoIP Gateway provides 12 to 32 analog FXS or FXO interfaces to connect phones, fax or PSTN trunk lines to your IP-based communications (IP PBX, UC systems and SIP Trunks).Like every SmartNode, the SN4300 supports every industry-standard CODEC to deliver toll-quality voice on every call. The Unified Communications Agent™ (UCA) provides any-to-any multi-path switching (simultaneous SIP, H.323, ISDN, and POTS calls with routing and conversion between TDM/PSTN and IP/Ethernet networks—plus T.38 and SuperG3 FAX). VoIP-over-VPN with voice encryption provides secure voice and data via IPsec with AES/DES strong encryption and automated keying via Internet Key Exchange (IKE). In addition, advanced call-router functionality includes least-cost call routing with flexible dialed-number plan support.The SmartNode survivability suite provides PSTN fallback to ensure business continuity in case the IP network fails. In addition, SmartNode delivers a smooth transition to VoIP with strong number portability support accepting incoming calls from the PSTN throughout the VoIP service provider's number porting process.Preserve investments in legacy phone equipment while taking the next steps toward unified communications with the SN4300 VoIP Gateway. Providing 12 to 32 FXS or FXO interfaces and one 10/100/1000 Ethernet, the SN4300 delivers a reliable, cost-effective solution for the Enterprise.