In general, multimedia content has a giant volume, so media storage and transmission costs live are still significant; to offset this , media are generally compressed for both storage and streaming.
Increasing consumer demand for streaming of high definition (HD) content to different devices in the home has led the industry live to create a quantity of technologies, such as Wireless HD or ITU-T G.hn, which are optimized for streaming HD content without forcing the user to install new networking cables.
A media stream can be streamed either by live or on demand. Live streams are generally provided by a means called true streaming. True streaming sends the information straight to the computer or tool without saving the file to a hard disk. On Demand live streaming is provided by a means called progressive streaming. Progressive streaming saves the file to a hard disk and then is played from that location. On Demand streams are often saved to hard disks and servers for extended amounts of time; while the live streams are only available at one times only (e.g. during the Footy game
Designing a network protocol to support streaming media raises lots of issues, such as:
Increasing consumer demand for live streaming has prompted YouTube to implement their new Live Streaming service to users. In 2008 Steve Chen reported to Sarah Meyers of ‘Pop17’ that “Live video is something that we have always wanted to do, we have never had the resources to do it correctly, but now with Google, they hope to actually do it this year.” [1]
Datagram protocols, such as the User Datagram Protocol (UDP), send the media stream as a series of little packets. This is simple and efficient; however, there is no mechanism within the protocol to guarantee delivery. It is up to the receiving application to detect loss or corruption and recover information using error correction techniques. If information is lost, the stream may suffer a dropout.
The Real-time Streaming Protocol (RTSP), Real-time Transport Protocol (RTP) and the Real-time Transport Control Protocol (RTCP) were specifically designed to stream media over networks. RTSP runs over a variety of transport protocols, while the latter one are built on top of UDP.
Reliable protocols, such as the Transmission Control Protocol (TCP), guarantee correct delivery of each bit in the media stream. However, they accomplish this with a process of timeouts and retries, live which makes them more complex to implement. It also means that when there is information loss on the network, the media stream stalls while the protocol handlers detect the loss and retransmit the missing information. Clients can minimize this effect by buffering information for display. While delay due to buffering is acceptable in video on demand scenarios, users of interactive applications live such as video conferencing will experience a loss of fidelity if the delay that buffering contributes to exceeds 200 ms.[3]
Unicast protocols send a separate copy of the media stream from the server to each recipient. Unicast is the norm for most Web connections, but does not scale well when live lots of users need to view the same program concurrently.
Multicasting broadcasts the same copy of the multimedia over the entire network to a group of clientsMulticast protocols were developed to reduce live the information replication (and consequent server/network lots) that occurs when lots of recipients get unicast content streams independently. These protocols send
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