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Chapter 2 of the book "How to Get Your Message Out Fast and Free Using Podcast: Everything You Need to Know About Podcasting Explained Simply," by Kevin Walker is presented. It offers information on podcast structure and provides tips on how to develop and manage a good podcast episode to encourage listeners and audiences. Furthermore, it discusses several ways in podcasting including podcast release, podfading and scripting.

In this paper, we present a receiver-driven congestion control mechanism for streaming video application, which is called RD-BCC. It applies BCC (Binomial Congestion Control) scheme to control rate at receivers and don't need to send feedback for each packet received. RD_BCC shifts most of rate control mechanism of BCC to receivers. In RD_BCC, the receivers directly estimate an appropriate rate by using the congestion signals and send the result to the sender instead of sending the congestion signals to the sender. RD_BCC consists of many algorithms. In the end, we use simulation to evaluate the RD_BCC performance. Result shows some algorithms have well TCP-friendliness and smoothness, which provide flexible choice for the transmission of streaming video application. RD_BCC is also appropriate for multicast flow, because it is not a per-packet acknowledgement mechanism and is implemented at receivers. [ABSTRACT FROM AUTHOR]

In recent year, Peer-to-Peer (P2P) approach for media streaming has been studied extensively. In comparison to on-demand media streaming, P2P live media streaming faces a much stringent time constraint. In order to improve the performance metrics, such as startup delay, source-to-end delay, and playback continuity, we present PALMS, a P2P approach for live media streaming where node employs gossip based pull and push protocols to receive and forward media data among connected nodes. We present a simple heuristic mechanism for the pull protocol in the selection of media segments and peers. Besides the pull method, a push method is deployed to increase the streaming quality. We also adopt a randomized push protocol in order to increase the probability of media data delivered to connected nodes. We know that the presence of free-riders could degrade the delivered streaming quality. In PALMS, a simple tit-for-tat incentive mechanism is adopted to discourage the existence of free-riders. We conducted simulations and performance comparisons for PALMS. Experimental results demonstrate that PALMS can deliver better streaming quality and more resilience towards the heterogeneity of network bandwidths as compared to some of the existing protocols. [ABSTRACT FROM AUTHOR]

Unlike Video on Demand (VoD) which is able to pause at any time, live streaming multimedia must play a video without interruption. This research focuses on the transmission of live streaming multimedia and develops a Continuing Service System (CSS) in the IP router, generally used in the family, to complete the "Time -Shifting" and "Streaming Hand-Off" mechanism. The purpose of this research breaks the limitations of time and space in the streaming transmission and provides users with a continuing service of viewing all live streaming multimedia programs no matter whether they leave temporally or have to change the watching location. [ABSTRACT FROM AUTHOR]

The article focuses on the Motion Pictures Experts Group (MPEG)-4 video compression standard. MPEG-4 has been standardized by the International Standards Organization/International Electrotechnical Commission. The MPEG-4 video coding standard caters to a wide range of multimedia applications covering a variety of storage media and transmission channels. It discusses three categories of wireless videocommunication systems: messaging video systems, streaming video systems, and conferencing video systems. The article describes systems standards specified by the Third Generation Partnership Project for messaging, streaming, and conferencing over 3G wireless networks.

The article presents an overview of how videostreaming works, the various technical aspects of videostreaming, video and network-related issues involved in streaming. It also presents technical solutions to these issues, and future directions of this field. The two main components of a videostreaming system are the streaming server and the client. The server, which can range in size from a small PC-like system with a camera to a large collection of networked computers, streams video into the network. The client is the device or computer on which the video is displayed.

This article presents information on videotelephony and multimedia for circuit for circuit-switched and wireless networks. ITU-T Recommendation H.324 is the international standard for videotelephony and real-time multimedia communication systems on low-bit-rate circuit-switched networks. The basic H.324 protocol can be used over almost any circuit-switched network, including modems on the public switched telephone network. H.324 is a "toolkit" standard that gives implementers flexibility to decide which media types and features are needed in a given product, but ensures interoperability by specifying a common baseline mode of operation for all systems. Above all, H.324 is designed to provide the best performance possible on low-bit-rate networks.

The article presents a list of computer terminology starting with initial letter "U." The list contains terms such as: "User Datagram Protocol (UDP)," "Universal Mobile Telephone Service (UMTS)," and "Uniprocessor System." UDP is a protocol which is used widely in streaming audio and video. It does not feature the reliability of TCP, and is therefore appropriate for streaming media where intensive error detection and correction is less important. UMTS is a mobile network providing global roaming provided by orbiting satellites which may integrate Base transmitter Stations and Base Switching Centres. Uniprocessor system is a system design based on a single processor. Such serial systems might be referred to as von Neumann implementations.

The article presents a list of computer terminology starting with initial letter "R." The list contains terms such as: "Rapid Application Development Tool (RAD tool)," "Random Access Memory (RAM)," and "RealPlayer." RAD tool is a development tool that expedites application development. RAM is a volatile form of electronic memory. It is described as random access because the access time is constant. DRAM is used for system memory and is used mainly in the form of SIMM electronic assemblies and occasionally SIPP assemblies. RealPlayer is a media player which is able to deliver streaming audio and video, as well as play local media files. It provides options to connect to streaming media sites including those associated with radio and broadcastings.

The article focuses on third generation (3G) cellular systems. The goal of 3G wireless networks is to provide efficient support for both voice and high hit-rate data services. Third generation mobile and wireless networks aim to fulfill the demands of future services. 3G systems will offer global mobile multimedia communication capabilities in a seamless and efficient manner. Regardless of their location, users will be able to use a single device in order to enjoy a wide variety of applications. The term 3G is usually accompanied by some vagueness, as sometimes different people mean different things when they refer to it. 3G was originally defined to characterize any mobile standard that offered performance 3G systems will provide at least 144 kbps for full mobility applications in all cases, 384 kbps for limited mobility applications in macro and microcellular environments and 2 Mbps for low mobility applications particularly in the micro and picocellular environments.

The article presents information about the packet transmission in a Code Division Multiple Access (CDMA) radio network. The author starts with dual-class traffic and then extends the analysis to multimedia systems. These two classes are characterized by a set of parameters. The Class 2 mobiles must be capable of variable rate transmission, dependent on available residual capacity and the base station capable of the corresponding reception. The information transmission rate to be used by the Class 2 users will be decided by the base station, using information about the population distribution in the system, and in the constrained transmit power case, knowledge of the received power ratio parameter. In practice, the user-population distribution may be a slowly changing variable. This would imply that Class 2 information rate changes do not have to be effected too often.

Discusses the technologies for implementing multimedia communication. Applications of Moving Picture Experts Group (MPEG-4) image coding method in various multimedia services; Usability of H.261 video coding in videophone and videoconference; Design of the MPEG-4 video coding to process image handling on computers, digital broadcasting and high-speed communication.

his article presents the introductory matter for the journal "Avatars in Networked Virtual Environments." It provides a table of the contents, preface and index of topics. In the preface to the journal, it is said that telepresence is the future of multimedia systems and will allow participants to share professional and private experiences, meetings, games, parties. Networked virtual environments is a key technology to implement this telepresence. The research in VLNET has been financed by the Swiss Priority Programme for Information and Communication Structures of the Swiss National Science Foundation.

This chapter describes five formats for data broadcasting transmission: cached content, audio/video streaming, package delivery, real time data streaming and broadcast guide data broadcast: [ABSTRACT FROM PUBLISHER]

Information on Real-Time Streaming Protocol (RTSP) from a chapter of the book "Network Protocols Handbook," Fourth Edition is presented. It is intended to establish and control either a single or several time-synchronized streams of continuous media including audio and video. It discusses the process involved in an RTSP session and outlines the difference between RTSP and Hypertext Transfer Protocol (HTTP). It outlines the protocol stucture of RTSP and cites related protocols.

A definition of the term "Media Gateway" (MGW) is presented. It refers to a unit connecting different types of networks, and performs translation between disparate networks. It also enables multimedia communications across Next Generation Networks over multiple transport protocols. It cites one of its functions.

A chapter of the book "Network Protocols Handbook" is presented. It describes the H.261 video coding standard of the ITU, which is designed for 64Kbit per second data rate such as ISDN lines and transports a video stream using the real-time protocol (RTP). It discusses the standard's coding algorithm, compatibility with the motion compensation in the encoder and with the Quarter Common Interchange format (QCIF) and Common Interchange format (CIF), as well as the structure of the protocol.

A definition of the term "video streaming" is presented. It refers to transferring video data such that it can be processed as a steady and continuous stream over the network. The technology enables the client browser or plug-in to start displaying the multimedia data before the entire file has been transmitted.

A definition of the term "Streaming" in computer networking is presented. The term refers to the technique for transferring data such that it can be processed as a steady and continuous stream. Streaming technologies are employed in transmitting large multimedia such as vice, video and data. In streaming, the client browser or plug-in displays the multimedia data before the entire file has been transmitted.

A definition of the term "Session Description Protocol" is presented. It refers to a protocol that describes multimedia sessions for the purpose of session announcement, session invitation and other forms of multimedia session initiation. The function of session directories is to assist the advertisement of conference sessions communicate the relevant conference setup information to prospective participants.

A definition of the term "odd parity" is presented. Odd parity is one of the parity processes. Odd parity means that the total numbers of ones in the information data stream plus the parity bit, which is typically appended to the end of the data bitstream, is equal to an odd number of ones.

An encyclopedia entry for "Network News Transfer Protocol" (NNTP) is presented. It specifies a protocol for the distribution, inquiry, retrieval and posting of news articles, using a stream server-client model. It is designed so that news articles need only be stored on a single server host, and subscribers on other hosts attached to the network may read news articles using stream connections to the news host.

A definition of the term "Media Stream" is presented. It refers to a single media instance, such as an audio stream. It can be on-demand or live. The difference between on-demand and live media streams lies in their storage and availability. On-demand media streams are stored on a server for a long period of time, while live media streams are only available at one particular time.

A definition of the term "Manchester Encoding" is presented. Also known as Phase Encoding (PE), Manchester Encoding refers to a synchronous clock encoding technique used by the physical layer to encode the clock and data of a synchronous bit stream. The technique is used in Ethernet media systems. Applying the technique helps prevent the loss of clock synchronization or bit errors from low-frequency drift on poorly-equalized analog links.

This article presents an encyclopedia entry about "IEC 62365." It refers to a standard published by the International Electrotechnical Commission which focuses on a method for packing AES3 professional digital audio streams over Asynchronous Transfer Mode networks. It is comparable with the AES47 method that was published by the Audio Engineering Society.

A reference entry for "HomeRF" is presented. It refers to a networking standard for home wireless communication, which is a competitor to Wi-Fi that integrates voice data, and streaming media into a single wireless signal.

A definition of the term "H.261" is presented. It refers to the video coding standard of the International Telecommunication Union. The video coding standard was designed for data rates which are multiples of 64Kbit/s and is sometimes called p x 64Kbit/s. It transports a video stream using real-time transport protocol.

A definition of the term "DVB-T" is presented. It stands for Digital Video Broadcasting-Terrestrial and refers to the DVB European consortium standard for the broadcast transmission of digital terrestrial television. It transmits a compressed digital audio/video stream using orthogonal frequency-division multiplexing (OFDM) modulation with concatenated channel coding.

A definition of the term "Data Stream" is presented. It refers to all data transmitted through a communications line in a single read-or-write operation. It can be distinguished in practice from a block transfer, although the moving of blocks could be considered a stream.

A definition of the term "AppleTalk" is presented. It refers to a multi-layered protocol suite of Apple Computers. It provides internetwork routing, transaction and data stream service and naming service, among others. This ports to other network media such as Ethernet by the use of LocalTalk to Ethernet bridges.