Multimedia communications pdf




















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Doeringer et al. Ferrari and D. Stukovic et al. Kurose, Open issues and challenges in providing quality of service guarantees in high speed networks, ACM Computer Commun. Campbell et al. Lazar, A real-time control management and information transport architecture for broadband networks, Proc. Zurich Sem. Digital Communications, —, May Zilterbart, B. Stiller, and A.

Clark, S. Sheuder, and L. Zhang, Supporting real-time applications in an integrated services packet network: architecture and mechanisms, Proc. Gonndan and D. ACM Symp. Li and W. Liao, Distributed multimedia systems, Proc. Wright, Assessment of alternative transport options for video distribution and retrieval Magazine, 35, 78—87 Hamidi, J.

Roberts, and P. Rolin, Rate control for VBR video coders in broadband Steinmetz and K. Nahrstedt, Multimedia Computing, Communications and Applications, CA, Seitz et al.

Magazine, 32, 56—66 Croworft et al. June Switzerland, November Guillemin, C. Levert, and C. Rosenberg, Cell conformance testing with respect to the Gallego and R. Grunenfelder, Testing and measurement problems in ATM networks, Tarnet, New directions in communications or which way to the information age?

IEEE Commun. Magazine, 24, 8—15 Kupers et al. Magazine, 40, 50—55 Medhi, QoS routing computations with path coding: a framework and network performance, The following figure shows how they. IP provides communication between hosts on different kinds of networks i. It is a connectionless, unreliable packet delivery service. Connectionless means that there is no handshaking, each packet is independent of any other packet.

It is unreliable because there is no guarantee that a packet gets delivered; higher level protocols must deal with that. IP defines an addressing scheme that is independent of the underlying physical address e.

IP specifies a unique bit number for each host on a network. These terms are interchangeable. Each packet sent across the internet contains the IP address of the source of the packet and the IP address of its destination.

For routing efficiency, the IP address is considered in two parts: the prefix which identifies the physical network, and the suffix which identifies a computer on the network. A unique prefix is needed for each network in an internet.

The first four bits of an IP address determine the class of the network. When interacting with mere humans, software uses dotted decimal notation; each 8 bits is treated as an unsigned binary integer separated by periods. IP reserves host address 0 to denote a network. For class A networks the netmask is always All hosts are required to support subnet addressing. While the IP address classes are the convention, IP addresses are typically subnetted to smaller address sets that do not match the class system.

The suffix bits are divided into a subnet ID and a host ID. This makes sense for class A and B networks, since no one attaches as many hosts to these networks as is allowed. Whether to subnet and how many bits to use for the subnet ID is determined by the local network administrator of each network. If subnetting is used, then the netmask will have to reflect this fact. On a class B network with subnetting, the netmask would not be The bits of the Host ID that were used for the subnet would need to be set in the netmask.

IP defines a directed broadcast address for each physical network as all ones in the host ID part of the address. The network ID and the subnet ID must be valid network and subnet values. If the IP address is all ones A router will not forward this type of broadcast to other sub networks.

Each IP datagram travels from its source to its destination by means of routers. All hosts and routers on an internet contain IP protocol software and use a routing table to determine where to send a packet next.

The destination IP address in the IP header contains the ultimate destination of the IP datagram, but it might go through several other IP addresses routers before reaching that destination. The entries can be updated manually by a network administrator or. Protocol RIP. Routing table entries provide needed information to each local host regarding how to communicate with remote networks and hosts.

The most specific to the least specific route is in the following order:. If a matching route is not found, IP discards the datagram. IP provides several other services:. This permits a large packet to travel across a network which only accepts smaller packets. IP fragments and reassembles packets transparent to the higher layers. If TTL reaches zero, the packet is discarded. The Address Resolution Protocol is used to translate virtual addresses to physical ones.

The network hardware does not. ARP employs the third strategy, message exchange. ARP defines a request and a response. A request message is placed in a hardware frame e. Only the computer whose IP address matches the request sends a response.

They provide end-to-end communication services for applications. This is a minimal service over IP, adding only optional checksumming of data and multiplexing by port number. UDP is often used by applications that need multicast or broadcast delivery, services not offered by TCP.

TCP is a connection-oriented transport service; it provides end- to-end reliability, resequencing, and flow control. TCP enables two hosts to establish a connection and exchange streams of data, which are treated in bytes. The delivery of data in the proper order is guaranteed. TCP can detect errors or lost data and can trigger retransmission until the data is received, complete and without errors. A TCP connection is done with a 3-way handshake between a client and a server.

The following is a simplified explanation of this process. The connection is then established and is uniquely identified by a 4-tuple called a socket or socket pair:.

During the connection setup phase, these values are entered in a table and saved for the duration of the connection. Every TCP segment has a header. The header comprises all necessary information for reliable, complete delivery of data.

Among other things, such as IP addresses, the header contains the following fields:. Sequence Number - This bit number contains either the sequence number of the first byte of data in this particular segment or the Initial Sequence Number ISN that identifies the first byte of data that will be sent for this particular connection. An ISN is chosen by both client and server. The following figure illustrates the three-way handshake. The sequence number is used to ensure the data is reassembled in the proper order before being passed to an application protocol.

It is the sequence number of the next expected byte of data. This field is only valid when the ACK control bit is set. Since sending an ACK costs nothing, because it and the Acknowledgement Number field are part of the header the ACK control bit is always set after a connection has been. Control Bits - This 6-bit field comprises the following 1-bit flags left to right :. Window Size - This bit number states how much data the receiving end of the TCP connection will allow.

The sending end of the TCP connection must stop and wait for an acknowledgement after it has sent the amount of data allowed. A checksum can only detect some errors, not all, and cannot correct any. Internet Control Message Protocol is a set of messages that communicate errors and other conditions that require attention.

Some ICMP messages are returned to application protocols. It sends an ICMP echo request and waits for a reply. Ping can be used to transmit a series of. Domain names are significant because they guide users to where they want to go on the Internet.

A domain name is simply an alphanumeric character string separated into segments by periods. Name servers contain information on some segment of the DNS and make that information available to clients who are called resolvers. If resolve is passed a domain name, a series of queries take place between the computer that called resolve and computers running name server.

For example, to resolve the domain name www. C can be used. Chances are that your local name server does not have the requested information, so it queries the root server. The root server will not know the IP address either, but it will know where to find the name server that contains authoritative information for the. This information is returned to your local name server, which then sends a query to the name server for the.

Again, this name server does not know the requested IP address, but does know the local name server that handles rabbitsemiconductor. This information is sent back to your local name server, who sends a final query to the local name server of rabbitsemiconductor. This local name server returns the requested IP address of. However, the very nature of the Internet means that this network is not suited for data transmission in real time and as a result the quality of audio sent over the Internet is usually of mediocre quality.

This theory specifically addresses the analysis and solution of these problems to enable an audio conferencing or telephone application over the Internet to change its behaviour to maintain an acceptable auditory quality even in cases where the network is quite congested. These solutions, in the form of control mechanisms, have been implemented and tested on the audio conferencing and telephone software on Internet Free Phone which we developed. A study on the effect that these machines would have on an Internet which developed to integrate the Fair Queuing service discipline has shown that while these mechanisms would still be necessary, they would perform even better on this network type.

The principal role of RTP is to implement the sequence numbers of IP packets to reform voice or video information even if the underlying network changes the order of the packets.

In addition, RTP may be conveyed by multicast packets in order to route conversations to multiple recipients. It is a control protocol for RTP flow, making it possible to convey basic information on the participants of a session and the quality of service.

The RTP header carries synchronisation and numbering information. The data coding will depend on the compression type. One RTP channel is used per type of flow: one for audio, one for video.

The field xxx is used for synchronisation. RTP offers an end to end service. It adds a header which provides timing information necessary for the synchronisation of sound and video type real time flow. Each of them uses a separate port from a pair of ports. This instant must be taken from a clock which increases in a monotonous and linear way in time to enable synchronisation and the calculation of the jitter at the destination.

The SSRC identifies the synchronisation source simply called "the source". This identifier is chosen randomly with the intent that it is unique among all the sources of the same session.

The number of identifiers is given in the CC field. This is an estimation of the time interval for an RTP data packet which is measured with the timestamp and which is in the form of a whole number. This is in fact the relative transit time between two data packets.

To do so, it must use a reservation protocol such as RSVP or make sure that the communication links used are correctly proportioned in relation to the use which is made of it. RTP is a session protocol, but it is placed in the application. It is for the developer to integrate it. How is the type of flow transported?

The type of flow is theoretically used in IP. Encode input sequence a1a2a3. Explain Non Uniform Quantization. Explain the two algorithms to achieve non uniform quantization. Explain with a neat diagram the six hierarchical layers for the bitstream of an MPEG- 1 video.

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Original Title: multimedia communication Notes. Related titles. Carousel Previous Carousel Next. Jump to Page. Search inside document. In this chapter, we are interested in multimedia communications; that is, we are interested in the transmission ECE Dept. Fig 1. For example, conferences involving more than ECE Dept. For example, one might think that lossy text ECE Dept.

The result of this observation is that jitter of only a few milliseconds during a ECE Dept. Dozens of programs exist for personal computers to allow users to record, display, edit, mix, and store sound waves ECE Dept.

Large amounts ECE Dept. For transmission purposes, it is now sufficient to encode just the ECE Dept. The computer runs experiment 1 again, but this time with a constant-amplitude ECE Dept. Communications: Applications,. Networks, Protocols. Covers all of the main subject areas associated with multimedia communications in one volume.

Applications, Networks, Protocols, and Standards. My library Help Advanced Book Search. Customise existing Pearson eLearning content to match the specific needs of your course. Multimesia This Product Features Table of Contents Backcover Copy Preface PDF Courses About This Product Description Multimedia Communications by Fred Halsall addresses the main subject areas associated with multimedia communications applications, networks, protocols, and standards at a level that enables the reader to develop an in-depth understanding of the technical issues associated with this rapidly evolving subject.

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Looking for technical support for your Pearson course materials? If you have a separate IRC account, please log in using that login name and password. Standards for Multimedia Communications 6. Please visit our Technical Support site. About This Product Features Table of Contents Backcover Copy Preface PDF Courses About This Product Description Halsapl Communications by Fred Halsall addresses the main subject areas associated with multimedia communications applications, networks, multimedai, and standards at a level that enables the reader to develop an in-depth understanding of the technical issues associated with this rapidly evolving subject.

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