Streaming Media

Joost

admin — Fri, 28 Mar 2008 19:49:14 +0000

Joost (pronounced /j oo?st/ “Juiced”) is a system for distributing TV shows and other forms of video over the Web using peer-to-peer TV technology, created by Niklas Zennström and Janus Friis (founders of Skype and Kazaa).Joost began development in 2006. Working under the code name “The Venice Project”, Zennström and Friis assembled teams of some 150 software developers in about six cities around the world, including New York, London, Leiden and Toulouse. According to Zennström at a 25 July 2007 press conference about Skype held in Tallinn, Estonia, Joost has signed up more than a million beta testers and is on track for an end-of-year launch.The teams are currently in negotiations with FOX networks. It has signed up with Warner Music, Indianapolis Motor Speedway Productions (Indianapolis 500, IndyCar Series) and production company Endemol for the betaIn February 2007, Viacom entered into a deal with the company to distribute content from its media properties, including MTV Networks, BET and film studio Paramount Pictures.

P2PTV-TV Streaming over P2P

admin — Fri, 28 Mar 2008 19:44:31 +0000

The term P2PTV refers to peer-to-peer (P2P) software applications designed to redistribute video streams in real time on a P2P network; the distributed video streams are typically TV channels from all over the world but may also come from other sources. The draw to these applications is significant because they have the potential to make any TV channel globally available.In a P2PTV system, each user, while downloading a video stream, is simultaneously also uploading that stream to other users, thus contributing to the overall available bandwidth. The arriving streams are typically a few minutes time-delayed compared to the original sources. The video quality of the channels usually depends on how many users are watching; the video quality is better if there are more users. The architecture of many P2PTV networks can be thought of as real-time versions of BitTorrent: if a user wishes to view a certain channel, the P2PTV software contacts a “tracker server” for that channel in order to obtain addresses of peers who distribute that channel; it then contacts these peers to receive the feed. The tracker records the user’s address, so that it can be given to other users who wish to view the same channel. In effect, this creates an overlay network on top of the regular internet for the distribution of real-time video content.Some applications allow users to broadcast their own streams, whether self-produced, obtained from a video file, or through a TV tuner card or video capture card.Many of the commercial P2PTV applications were developed in China (TVants, TVUPlayer, PPLive, PPStream, SopCast). The majority of available applications broadcast mainly Asian TV stations, with the exception of TVUPlayer, which carries a number of North American stations including CBS, Spike TV, and Fox News. Some applications distribute TV channels without a proper license to do so; this utilization of P2P technology is particularly popular to view channels that are either not available locally, or only available by paid subscription, as is the case for some sports channels.

MSN TV

admin — Fri, 28 Mar 2008 19:41:07 +0000

MSN TV (formerly WebTV) is the name of both a thin client which uses a television for display (rather than a computer monitor), and the online service that supports it
The product and service were developed by WebTV Networks, Inc., a company purchased by Microsoft Corporation and absorbed into MSN (the Microsoft Network). While most thin clients developed in the mid-1990s were positioned as diskless workstations for corporate intranets, WebTV was positioned as a consumer device for web access.The WebTV product is an adapter that allows a television set to be connected to the internet, primarily for web browsing and e-mail. The setup includes a web browser, cord or wireless (i.e. bluetooth or IRDA) keyboard and connection to the Internet (i.e. using modem, ADSL, cable, PLC).While WebTV does not allow as much functionality as a computer-based browser, it is a low-cost alternative to a traditional computer connection to the Internet.It should be noted that the term web TV is also used concerning TV transmissions over the Internet, usually by streaming.

Audio Contribution over IP

admin — Fri, 28 Mar 2008 19:36:58 +0000

* Streaming of high quality audio over IP networks is being increasingly used by broadcasters and others to provide high-quality audio contribution feeds (for example to provide audio links between a sports venue or concert hall and the broadcaster’s studios)
* Audio quality and delay (on duplex transmissions) are key issues for contribution links.
* In the past these links have made widespread use of ISDN services but these are becoming increasingly difficult or expensive to obtain in some parts of Europe and are being phased out in others

Nullsoft Streaming Video

admin — Tue, 18 Mar 2008 03:32:09 +0000

Nullsoft Streaming Video (NSV) is a media container designed for streaming video content over the internet. NSV was developed by Nullsoft, the makers of Winamp.The NSV format is another example of streaming video formats, offered by various companies and media players. Windows Media, QuickTime video, RealAudio and RealVideo streams are just a few examples of these, and each have their benefits and shortcomings.One of the shortfalls of most streaming formats often comes from the difficulty in finding streams, or at least some form of directory service; and in some cases, none are readily available to the public at all.With Nullsoft’s SHOUTcast service, a media streaming and directory system, the NSV format has become very popular, along with SHOUTcast’s enormous listings of Streaming Audio, which reach well into the thousands.With Winamp version 5, came the convenient “Media Library”, which offers integrated listings of both audio and video streams, returned directly from the SHOUTcast service. There was also a Library in Winamp 2.9x and 3, but came under a different name, “Library”.NSV streaming started in early 2003, but the real explosion came with the introduction of Winamp 5. Also, NSV is no longer proprietary, but instead an open format,

SHOUTcast

admin — Tue, 18 Mar 2008 03:30:03 +0000

SHOUTcast is a multiplatform freeware digital audio streaming technology developed by Nullsoft. It allows audio content, primarily in MP3 or HE-AAC format, to be broadcast to and from media player software, enabling hobbyists and professionals to create Internet radio/Web radio networks.SHOUTcast consists of a client-server model, with each component communicating via a network protocol that intermingles audio data with metadata such as song titles and the station name. It uses HTTP as a transport protocol, although multicast is another option.SHOUTcast servers and clients are available for Palm OS, Microsoft Windows, FreeBSD, Linux, Mac OS X, and Solaris. Client-only versions exist on Windows Mobile, Series 60, the PlayStation Portable (PSPradio), and on Nintendo DS (DSOrganize).The output format is supported by multiple clients, including Nullsoft’s own Winamp, VLC media player, Amarok, XMMS, Zinf and Apple iTunes.

User Datagram Protocol

admin — Tue, 18 Mar 2008 03:22:24 +0000

User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages sometimes known as datagrams (using Datagram Sockets) to one another. UDP is sometimes called the Universal Datagram Protocol. It was designed by David P. Reed in 1980.UDP does not guarantee reliability or ordering in the way that TCP does. Datagrams may arrive out of order, appear duplicated, or go missing without notice. Avoiding the overhead of checking whether every packet actually arrived makes UDP faster and more efficient, at least for applications that do not need guaranteed delivery. Time-sensitive applications often use UDP because dropped packets are preferable to delayed packets. UDP’s stateless nature is also useful for servers that answer small queries from huge numbers of clients. Unlike TCP, UDP is compatible with packet broadcast (sending to all on local network) and multicasting (send to all subscribers).

Common network applications that use UDP include: the Domain Name System (DNS), streaming media applications such as IPTV, Voice over IP (VoIP), Trivial File Transfer Protocol (TFTP) and online games.

IPTV

admin — Tue, 18 Mar 2008 03:21:29 +0000

IPTV (IP Television) is a system where a digital television service is delivered by using Internet Protocol over a network infrastructure, which may include delivery by a broadband connection. A general definition of IPTV is television content that, instead of being delivered through traditional broadcast and cable formats, is received by the viewer through the technologies used for computer networks.For residential users, IPTV is often provided in conjunction with Video on Demand and may be bundled with Internet services such as Web access and VoIP. The commercial bundling of IPTV, VoIP and Internet access is referred to as “Triple Play” service (adding mobility is called “Quadruple Play”). IPTV is typically supplied by a service provider using a closed network infrastructure. This closed network approach is in competition with the delivery of TV content over the public Internet, called Internet Television. In businesses, IPTV may be used to deliver television content over corporate LANs.

Real-time Transport Protocol

admin — Fri, 07 Mar 2008 13:30:45 +0000

The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 which was made obsolete in 2003 by RFC 3550. Real time transport protocol can also be used in conjunction with RSVP protocol which enhances the field of multimedia applications.

RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports 16384-32767. RTP can carry any data with real-time characteristics, such as interactive audio and video. Call setup and tear-down for VoIP applications is usually performed by either SIP or H.323 protocols. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server.

It was originally designed as a multicast protocol, but has since been applied in many unicast applications. It is frequently used in streaming media systems (in conjunction with RTSP) as well as videoconferencing and push to talk systems (in conjunction with H.323 or SIP), making it the technical foundation of the Voice over IP industry. It goes along with the RTCP and is built on top of the User Datagram Protocol (UDP). Applications using RTP are less sensitive to packet loss, but typically very sensitive to delays, so UDP is a better choice than TCP for such applications.

According to RFC 1889, the services provided by RTP include:

* Payload-type identification - Indication of what kind of content is being carried
* Sequence numbering - PDU sequence number
* Time stamping - allow synchronization and jitter calculations
* Delivery monitoring

The protocols themselves do not provide mechanisms to ensure timely delivery. They also do not give any Quality of Service (QoS) guarantees. These things have to be provided by some other mechanism.

Also, out of order delivery is still possible, and flow and congestion control are not supported directly. However, the protocols do deliver the necessary data to the application to make sure it can put the received packets in the correct order. Also, RTCP provides information about reception quality which the application can use to make local adjustments. For example if a congestion is forming, the application could decide to lower the data rate.

RTP was also published by the ITU-T as H.225.0, but later removed once the IETF had a stable standards-track RFC published. It exists as an Internet Standard (STD 64) defined in RFC 3550 (which obsoletes RFC 1889). RFC 3551 (STD 65) (which obsoletes RFC 1890) defines a specific profile for Audio and Video Conferences with Minimal Control. RFC 3711 defines the Secure Real-time Transport Protocol (SRTP) profile (actually an extension to RTP Profile for Audio and Video Conferences) which can be used (optionally) to provide confidentiality, message authentication, and replay protection for audio and video streams being delivered.

Real Time Streaming Protocol

admin — Fri, 07 Mar 2008 13:29:28 +0000

The Real Time Streaming Protocol (RTSP), developed by the IETF and created in 1998 as RFC 2326, is a protocol for use in streaming media systems which allows a client to remotely control a streaming media server, issuing VCR-like commands such as “play” and “pause”, and allowing time-based access to files on a server.The sending of streaming data itself is not part of the RTSP protocol. Most RTSP servers use the standards-based RTP as the transport protocol for the actual audio/video data, acting somewhat as a metadata channel. The RTSP server from RealNetworks also features Real’s proprietary RDT as the transport protocol.The protocol is similar in syntax and operation to HTTP but RTSP adds new requests. While HTTP is stateless, RTSP is a stateful protocol. A session ID is used to keep track of sessions when needed. This way, no permanent TCP connection is needed. RTSP messages are sent from client to server, although some exceptions exist where the server will send to the client. Below are the basic RTSP requests. A number of typical HTTP requests, like an OPTION request, are also frequently used.