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PDF | This paper includes a brief survey on Transport Layer Protocols like User Datagram Protocol (UDP), Transmission ControlProtocol (TCP). CSES. Washington University in St. Louis. Overview. ❑ Transport Layer Design Issues: ❑ Multiplexing/Demultiplexing. ❑ Flow control. ❑ Error control. ❑ UDP. Learn about Internet transport layer protocols: UDP: connectionless transport. TCP: connection-oriented reliable transport. TCP congestion control.


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Transport-layer services. ❒ Multiplexing and demultiplexing. ❒ Connectionless transport: UDP. ❒ Principles of reliable data transfer. segments into messages, passes to app layer. ❒ more than one transport protocol available to apps. ❍ Internet: UDP, TCP, SCTP application transport network. This chapter provides an overview of the most important and common protocols of the TCP/IP transport layer. These include: • User Datagram Protocol (UDP).

Bandwidth is one of the most important. The sender begins by sending an ACK Acknowledgement packet to the receiver requesting for a connection to be established. It also allows human interaction with the actual application to be used for communication. He is currently working on. Real-time applications react very poorly to congestion control. Bandwidth refers to the amount of information that can be transmitted over a network in a given amount of time, usually expressed in bits per second or bps [1.

Network layer has the Internet Protocol IP which is a best effort protocol. It only makes every effort to deliver the packets without any guarantees. To achieve the desired quality of service required by multimedia traffic, application developers need to carefully select which transport layer protocol to deploy for their application.

The application layer is the starting point of all communication sessions and defines the type of messages to be exchanged between the sender and receiver of a particular application process. It also allows human interaction with the actual application to be used for communication. After the application layer has played it role, the packets are passed down to the presentation layer. At this stage, offers encoding and decoding of data. This ensures that the correct data format is encoded and sent to the destination so that the right decoding system is used to read the data.

In the case of Multimedia applications; various video and audio formats exist. It is essential that the correct encoding scheme is used. Other functions the presentation layer offers is data compressing and encryption The session layer comes after the presentation layer and it allows for the control of dialogs or sessions. All applications running on a network must pass the data packets to the transport layer to provide process to process communication.

Among other things, the transport layer is responsible for identifying the sending and receiving applications; this ensures that the correct application gets the data that was intended for it. For this, the transport layer uses port number addressing to uniquely identify the application or process.

These protocol offer different services, with some being conducive for multimedia networking and some having a poor performance on the quality of service. After the transport layer has played its role, the packets are handed over to the Network layer. The network layer has the Internet Protocol which offers a best effort delivery service.

The challenges in multimedia network arise due to the distinct characteristics of audio and video data. Most multimedia applications are highly sensitive to end-to-end delay and jitter but can tolerate occasional loss of data.

The traffic flow of this data needs to be continuous. Any delays or breaks in the flow can have a negative impact on the user experience.

Delay and jitter are also very important in multimedia networking. Multimedia applications have a maximum time required from the time the packets are transmitted to the time they reach the destination; this is the delay time.

Delay should be very minimal and is measured in milliseconds ms. It is the variance in packet delays. End to end delay can be caused by queues and buffers created in the routers. End- to-end delay can be sub divided into various categories, and these are; propagation, processing and queueing delays. The quality of service required by multimedia data is regulated by many parameters. Bandwidth is one of the most important. Bandwidth refers to the amount of information that can be transmitted over a network in a given amount of time, usually expressed in bits per second or bps [1.

Pdf transport layer

The data rates in these applications is high and often has a lower bound to ensure continuous playback. The user should be able to play the audio or video with the content being fully downloaded. This is called continuous or streaming media. The traffic generated by multimedia applications is of a higher bit rates as compared to other usual network applications [2. Each class has its own set of service requirements. This media become interactive when the user is able to pause and fast- forward through the video content.

The user experience should be continuous video and audio play without any breaks or delays in receiving the content. These conversational systems enable users to conference with three or more parties. The interactivity allows users to communicate with others in real-time. This category of applications allow a bidirectional traffic to flow in both ways simultaneously.

These applications are very sensitive to jitter or the delay between packets. Streaming live media and stored media are both sensitive to delay and do not accept retransmissions. However, store media is on-demand and unicast while live streaming is multicast and is live [4. The transport layer provides a logical communication between applications or processes running on different machines.

This process to process means that the transport layer is in charge of the delivery of packets between the actual applications. Port Numbering To identify these applications, the transport layer deploys port numbers to uniquely identify the sending and receiving applications or processes.

When a service starts, a program listens on a port and informs the operating system it wishes to receive data directed to that port [5. Well-known port numbers range from 0 to 1, and these are assigned and controlled. Registered port numbers range from to 49, and are not assigned or controlled.

Dynamic Port numbers are from 49, to 65, and they are used temporary for private port numbering. Encapsulation and Decapsulation When a packet is processed at the transport layer, additional information such as socket addresses is added to the packet header and this is called encapsulation.

This takes place at the sender side. At the receiver side, the data that was added during the encapsulation process is read and the packet can be passed on to the appropriate process. This is called decapsulation.

Transport Layer Protocols And Services

These services add processing delays of packets and other overheads which can increase jitter cause problems for multimedia applications. Connection-Oriented TCP is connection-oriented protocol.

This means that before data is exchange, there must be a connection establishment to be made between the sender and the receiver. Establishing a connection involves a process known as the Three-Way-Handshake. The sender begins by sending an ACK Acknowledgement packet to the receiver requesting for a connection to be established. Either the sender or the receiver can initiate the termination of the connection.

It starts with sending a Fin Flag to the party. The other part then sends back an ACK of Fin and also sends a Fin flag back to the connection termination initiator. The initiated send a final ACK of Fin to completely terminate the connection.

This process of establishing and terminating a connection can cause unnecessary delays for multimedia applications. Flow control creates a balance between production and consumption of packets [6. However, when these buffers are packets are discarded.

The packets should be processed as soon as they get to the destination node. Error Control TCP is a reliable protocol. This means that it ensures that packets are delivered to the destination and are undamaged. This is achieved via the error control services in which TCP detects and discards corrupt packets, keeps track of lost and discarded packets and resends them. Multimedia applications have a tolerance to packet loss and resending packets can have a negative effect on the streaming of real-time packets [7.

TCP uses a retransmission mechanism to resend the packets that have not been acknowledged at destination host. TCP congestion control regulates and keeps the load below the capacity. Like flow control, congestion control deploys buffers at routers and switches. This simply means that the packets have to be stored or delayed in the buffer. TCP congestion control reduces the rate at the sender to a lower rate that the drain rate at the receiver. This has a severe impact on voice intelligibility at the receiver.

It is a reliable, message oriented transport layer protocol. SCTP is a multiple stream oriented protocol; meaning that multiple streams are allowed in each connection, and these streams are called associations.

This protocol is connectionless protocol and does not require a connection to be established for data transfer to begin. Apart from this, the packets are not numbered; UDP does not track down lost or delayed packets. In other words, UDP offers an unreliable service in which packets are not guaranteed of delivery.

It can be of no use to retransmit packets lost in a live multimedia session It relies on the application layer to offer a reliability of packet delivery in case the service is required. UDP does not offer congestion control mechanisms.

Real-time applications react very poorly to congestion control. This makes processing of packets very fast. As soon as an applications processes the data, it passes it to the transport layer via the UDP protocol, which will package the data inside the UDP segment and immediately pass the segment to the network layer without delay [ This simply means that the data on a network segment is multiplied and sent to a group of clients. It is applied in applications where there is need to broadcast a stream of media to more than one user.

Well-known port numbers range from 0 to 1, and these are assigned and controlled. Registered port numbers range from to 49, and are not assigned or controlled. Dynamic Port numbers are from 49, to 65, and they are used temporary for private port numbering. Encapsulation and Decapsulation When a packet is processed at the transport layer, additional information such as socket addresses is added to the packet header and this is called encapsulation. This takes place at the sender side. At the receiver side, the data that was added during the encapsulation process is read and the packet can be passed on to the appropriate process.

This is called decapsulation. These services add processing delays of packets and other overheads which can increase jitter cause problems for multimedia applications. Connection-Oriented TCP is connection-oriented protocol.

This means that before data is exchange, there must be a connection establishment to be made between the sender and the receiver. Establishing a connection involves a process known as the Three-Way-Handshake. The sender begins by sending an ACK Acknowledgement packet to the receiver requesting for a connection to be established. Either the sender or the receiver can initiate the termination of the connection.

It starts with sending a Fin Flag to the party. The other part then sends back an ACK of Fin and also sends a Fin flag back to the connection termination initiator.

(PDF) Multimedia Communication & Transport Layer Protocols | Mymemes Mugala - lesforgesdessalles.info

The initiated send a final ACK of Fin to completely terminate the connection. This process of establishing and terminating a connection can cause unnecessary delays for multimedia applications. Flow control creates a balance between production and consumption of packets [6.

However, when these buffers are packets are discarded. The packets should be processed as soon as they get to the destination node. Error Control TCP is a reliable protocol. This means that it ensures that packets are delivered to the destination and are undamaged.

This is achieved via the error control services in which TCP detects and discards corrupt packets, keeps track of lost and discarded packets and resends them. Multimedia applications have a tolerance to packet loss and resending packets can have a negative effect on the streaming of real-time packets [7.

TCP uses a retransmission mechanism to resend the packets that have not been acknowledged at destination host. TCP congestion control regulates and keeps the load below the capacity. Like flow control, congestion control deploys buffers at routers and switches.

This simply means that the packets have to be stored or delayed in the buffer.

Pdf transport layer

TCP congestion control reduces the rate at the sender to a lower rate that the drain rate at the receiver. This has a severe impact on voice intelligibility at the receiver. It is a reliable, message oriented transport layer protocol. SCTP is a multiple stream oriented protocol; meaning that multiple streams are allowed in each connection, and these streams are called associations. This protocol is connectionless protocol and does not require a connection to be established for data transfer to begin.

Apart from this, the packets are not numbered; UDP does not track down lost or delayed packets.

In other words, UDP offers an unreliable service in which packets are not guaranteed of delivery. It can be of no use to retransmit packets lost in a live multimedia session It relies on the application layer to offer a reliability of packet delivery in case the service is required.

UDP does not offer congestion control mechanisms. Real-time applications react very poorly to congestion control. This makes processing of packets very fast.

As soon as an applications processes the data, it passes it to the transport layer via the UDP protocol, which will package the data inside the UDP segment and immediately pass the segment to the network layer without delay [ This simply means that the data on a network segment is multiplied and sent to a group of clients. It is applied in applications where there is need to broadcast a stream of media to more than one user.

For multimedia streaming applications to perform accordingly, there should be very minimal or no delays during packet processing. Both TCP and SCTP are connection oriented protocol, with flow control, error and congestion control services slows down packet processing.

Pdf transport layer

UDP relies on the application layer to offer this service and other required services. These protocols provide services for data with real-time characteristics.

Multimedia applications use different encoding schemes and each of these have various trade-offs between quality, bandwidth and computational cost. RTP is responsible to negotiate and select which scheme the sending and receiving applications should deploy. RTP adds the payload type in the header of each packet. This indicates the media data type and the encoding scheme. This allows the receiver to know how to decode the data. For applications to perform well, the jitter caused by routers and other network constrains should be handled.

Jitter changes the original spacing between the packets and this cause the packets not to play at the right time. RTP is responsible to sort out this spacing and advises the receiver about the timing relationships among the received packets. To achieve this, RTP deploys timestamps.

The timestamps records the instant when the first part of the packet was sampled and is set by the sender. Multimedia data packets, like any other packets, required to be delivered in an orderly manner.

However, UDP does not offer this facility. On the other hand, RTP offers the sequence numbers for the packets. This allows the receiver to ensure that the packets can be reconstructed in order by the receiver and also to know which packet comes first and last or in case of buffering, the packets can be stored and played according the sequence number given.

Sequence numbers also allow the completion of the function of time stamping. RTCP delivers quality of service feedback for media distribution through recurrent statistical data transmission to streaming multimedia contributors [ Quality of service come into play when the application receives this information and changes its parameters by regulating packet flow or using a different encoding scheme.

The protocols focus is to provide out-of-band statistics and control information for RTP sessions. The RTCP packets contain direct information for quality of service monitoring.

Using adaptive encoding, these types of packets can be used to deploy a flow control mechanism resembling TCP at the application level. The Source Description provides information about the source, this includes the email address, phone number, and full name of the source. To keep track of each session, RTCP deploys the transport-level identifier known as the canonical name. This protocol controls multimedia servers and streams delivery.

RTSP is used for establishing and controlling media sessions between client and server. In other words, RTSP sets up and controls streams of continuous audio and video media. This protocol work hand in hand with RTP, RSVP and RTCP to offer client services that facilitate real-time control of playback such as play, random-seek, pause multimedia files from the server and also offer a complete streaming server over a network.

The Real Time Streaming Protocol establishes and controls either a single or several time-synchronized streams of continuous media such as audio and video [ In the first step, the client sends a Describe request to the server.

This requests the server to issue out information regarding the media type image, video or audio , frame rate, codec and other parameters. In the second step, the client send a Setup request to the server.

Layer pdf transport

The session is only setup when the server sends back a session ID to the client. This identifier is what the server uses to maintain the connection. The third step involves requesting and receiving the actual media. At this stage the server transmits the content using RTP. The client can then store, play, pause, fast- forward and rewind the media.

The client sends feedback RTCP packets to the server to offer quality of service information. On the client is done, it sends this packet to the server requesting to terminate the session. RSTP is responsible for setting up conferencing communication. A media server can be invited to join an existing conference where there are more than three media servers participating in the session.

This includes both audio and video conferencing applications such as skype. RTSP is able to keep track of the concurrent communication sessions using a state identifier and for this reason it is a statefull protocol. This job is given to the SRTP. This protocol provides encryption, message authentication and integrity, and replay protection to the RTP data [