CD 321 Multimedia Technology II

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CD 321 Multimedia Technology II

Lecture 3: Multimedia Protocols

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Multimedia O ver Internet

There are other ways to transmit multimedia data , D edicated links is a communications cable or other facility dedicated to a specific application. It is a communication path between two points. Networking Cables are networking hardware used to connect one network device to other network devices or to connect two or more computers to share printers, scanners etc. F or example coaxial cable, optical fiber cable, and twisted pair cables. Asynchronous Transfer Mode (ATM ) is a telecommunications standard defined by ANSI and ITU (formerly CCITT) for digital transmission of multiple types of traffic, including telephony (voice), data, and video signals in one network without the use of separate overlay networks.

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Dedicated links and cables are not practical because they require special installation and new software. Without an existing technology like LAN,WAN, the software development will be extremely expensive. ATM was said to be the ultimate solution for multimedia because it supports very high bandwidth, is connection-oriented and can tailor different level of quality of service to different type of applications. But at this moment, very few users have ATM networks reaching their organization, even fewer have ATM connections to their desktops.

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On the other hand, the Internet is growing exponentially. The well established LAN and WAN technologies based on IP protocol suite connect bigger and bigger networks all over the world to the Internet. Internet has became the platform of most networking activities. This is the primary reason to develop multimedia protocols over Internet.

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Another benefit of running multimedia over IP is that users can have integrated data and multimedia service over one single network, without investing on another network hardware and building the interface between two networks. At current time, IP and Ethernet seem to be more favoured in the desktops and LANs, with ATM in wide area networks.

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Multimedia Services

Multimedia services is a combination of two or more media components such as voice, data, video and still image in a single session delivered between two or more parties. Multimedia services can classified as Interactive services distribution services User interaction defines the behavior of interactive services. Distribution services are broadcasted-based

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Multimedia over Internet, Several Issues Must be Solved

As a shared datagram network, Internet is not naturally suitable for real-time traffic. Multimedia means extremely dense data and heavy traffic. The hardware has to provide enough bandwidth. Multimedia applications are usually related to multicast, i.e.,  the same data stream, not multiple copies, is sent a group of receivers. For example, in video conference, the video data need to be sent to all participants at the same time. Live video can be sent to thousands of recipients. The protocols designed for multimedia applications must take into account multicast in order to reduce the traffic.

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T he price tag attached shared network resources is unpredictable availability. But real-time applications require guaranteed bandwidth when the transmission takes place.  So there must be some mechanisms for real-time applications to reserve resources along the transmission path. Internet is a packet-switching datagram network where packets are routed independently across shared networks. The current technologies cannot guarantee that real-time data will reach the destination without being jumbled and jerky. Some new transport protocols must be used to take care of the timing issues so that audio and video data can be played back continuously with correct timing and synchronization. T here should be some standard operations for applications to manage the delivery and present the multimedia data

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Multimedia over Internet Solution

The Internet carries all types of traffic, each type has different characteristics and requirements. For example, a file transfer application requires that some quantity of data is transferred in an acceptable amount of time, while Internet telephony requires that most packets get to the receiver in less than 0.3 seconds. If enough bandwidth is available, best-effort service fullfils all of these requirements. When resources are limited, however, real-time traffic will suffer from the congestion.

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The solution for multimedia over IP is to classify all traffic, allocate priority for different applications and make reservations. The Integrated Services working group in the IETF (Internet Engineering Task Force) developed an enhanced Internet service model called Integrated Services that includes best-effort service and real-time service , see RFC 1633. Integrated services  or  IntServ  is an architecture that specifies the elements to guarantee quality of service ( QoS ) on networks. IntServ can for example be used to allow video and sound to reach the receiver without interruption.

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Real-time network services are designed to deliver real-time information with a high quality of service. Real-time information is live voice and video. Real-time services are designed to deliver real-time information in a timely manner . Resource Reservation Protocol (RSVP), together with Real-time Transport Protocol (RTP), Real-Time Control Protocol (RTCP), Real-Time Streaming Protocol (RTSP), provides a working foundation for real-time services. Integrated Services allows applications to configure and manage a single infrastructure for multimedia applications and traditional applications. It is a comprehensive approach to provide applications with the type of service they need and in the quality they choose.

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Multimedia Transport Protocols

The goal of multimedia transport protocols is to transmit multimedia signals from one point to another point. These points are connected by communication network employing specific protocols. Generally , multimedia original signals are encoded to reduce the bit rate. When the encoded stream is to be sent to another location in the network, the transport protocols are responsible for the packetization and the delivery of that stream. At the other side, the encoded multimedia stream is reconstructed from the stream of delivered packets and then decoded to produce a useful multimedia signal to be played back or stored for further use.

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The Internet Protocol (IP) is a packet-based network protocol used to exchange data over networks. It is the underlying network protocol, i.e. other protocols are built over IP. The most used higher level protocol is the Transport Control Protocol (TCP), which is a reliable transport protocol designed for data transmission and extensively used in Internet services. TCP is not suitable for real-time applications as the retransmissions can lead to high delay and cause delay jitter, which significantly degrades the quality .  In addition, it does not support multicast. Also , congestion control mechanisms, namely slow start, are not suitable for audio or video media transmission

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User Datagram Protocol (UDP )

In the other hand the transport protocol that is generally used for real-time multimedia transmission is the User Datagram Protocol (UDP). UDP does not guarantee the arrival of the packet, I t is up to the application or higher level protocols to take care of the sent data. The most used, for real time applications, protocol which is built over UDP is Real time Transport Protocols (RTP).

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Real time Transport Protocols (RTP )

The most important variables governing the operation of RTP are the Time Stamp (TS) and the Sequence Number (SN). The TS is responsible for placing the incoming packets in correct timing order . The SN is used to detect packet loss occurrences. It is increased by one for each packet in the stream. It should be mentioned that for a video frame that is split into multiple RTP packets, these packets share the same value of TS but use different SN. There is a separate control protocol that is generally used with RTP, which is named Real Time Control Protocol (RTCP).

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Real Time Control Protocol (RTCP )

RTCP synchronizes across different media streams by feedback messages (e.g. the sender report). It also provides feedback on the quality of data transmission by using lost packet counts in the Receiver Report. In addition, it identifies and keeps track of the participants. RTCP reports are sent periodically (every 5 sec.) between participants with the restriction that its traffic should not exceed 5% of the total data traffic. RTP supports multicasting, payload type identification, time stamping, sequence numbering, delivery monitoring.

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Five RTCP Packet Types

RR :  receiver report. Receiver reports are generated by participants that are not active senders. They contain reception quality feedback about data delivery, including the highest packets number received, the number of packets lost, inter-arrival jitter, and timestamps to calculate the round-trip delay between the sender and the receiver. SR : sender report. Sender reports are generated by active senders. In addition to the reception quality feedback as in RR, they contain a sender information section, providing information on inter-media synchronization, cumulative packet counters, and number of bytes sent.

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SDES : source description items. They contains information to describe the sources. BYE : indicates end of participation. APP : application specific functions. It is now intended for experimental use as new applications and new features are developed.

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In addition, the underlying UDP protocol supports multiplexing and checksum services. Even if RTP is the most used protocol for real-time applications, it has some problems. First , it does not support the timely delivery of data or any other QoS guarantee. In-time delivery requires lower layers that have control over resources in switches or routers (e.g. Resource Reservation Protocol, RSVP). Second , it does not guarantee delivery, so packets may be delivered out of order or get lost. Also , there is no mechanism to recover from packet loss.

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RSVP(Resource Reservation Protocol)

RSVP is the network control protocol that allows data receiver to request a special end-to-end quality of service for its data flows. Real-time applications use RSVP to reserve necessary resources at routers along the transmission paths so that the requested bandwidth can be available when the transmission actually takes place. RSVP is a main component of the future Integrated Services Internet which can provide both best-effort and real-time service.