Section 18.7. Summary


18.7. Summary

This chapter explored transport mechanisms for application delivery with the highest possible quality. We focuesed on media applications, such as streaming audio and video, one-to-many transmission of real-time audio and video, and real-time interactive audio and video.

We discussed how sampled and digitized voice are treated in networks and investigated two VoIP signaling session protocols: Session Initiation Protocol (SIP) and the H.323 series of protocols , showing how calling and numbering for these protocols can be achieved. SIP identifies user location signals, call setup, call termination, and busy signals. User agents assist in initiating or terminating a phone call in VoIP networks. User agents can be implemented in a standard telephone or in a laptop with a microphone that runs some software. The signaling in H.323 uses both UDP and TCP and is partitioned into call setup , initial communication capability , audio/video communication establishment , communications , and call termination . Various scheduling-policy mechanisms priority queueing , weighted fair queuing , and class-based weighted fair queuing can provide the foundation of a QoS networking architecture.

Senders use various real-time media transport protocols to send a stream of data at a constant rate. One of the protocols for real-time transmission is the Real-Time Transport Protocol (RTP), which provides application-level framing. Real-time applications, such as voice and video, can tolerate a certain amount of packet loss and do not always require retransmission of data. But if the rate of packet loss is very high, the source might use a lower-quality transmission, thereby placing less load on the network.

A video in a single server can be streamed from a video server to a client for each client request. Content distribution networks (CDNs) can be used for streaming data. A video streaming provided to a user in an ISP domain can use Domain Name System (DNS) servers to direct browsers to the correct server. The Stream Control Transmission Protocol (SCTP) is a general-purpose transport protocol for transporting stream traffic. SCTP offers acknowledged and nonduplicated transmission of basic units of data, or datagrams, on connectionless networks.

Finally, a non-Markovian analysis of streaming traffic was presented. A stream of packets generated from a server source can be modeled as a discrete sequence of events and defined as a discrete-time 0-1 valued process called self-similar traffic.

The last two chapters of the book consider two advanced and related subjects: mobile ad hoc networks and wireless sensor networks .



Computer and Communication Networks
Computer and Communication Networks (paperback)
ISBN: 0131389106
EAN: 2147483647
Year: 2007
Pages: 211
Authors: Nader F. Mir

flylib.com © 2008-2017.
If you may any questions please contact us: flylib@qtcs.net