Transport protocols Introduction

Transcription

1 Transport protocols 12.1 Introduction All protocol suites have one or more transport protocols to mask the corresponding application protocols from the service provided by the different types of network protocols. In the TCP/IP suite, the transport protocols used are the transmission control protocol (TCP) and the user datagram protocol (UDP). When the application involves the transfer of streams of audio and/or video in real time, the timing information required by the receiver to synchronize the incoming streams is provided by the real-time transport protocol (RTP) and its associated real-time transport control protocol (RTCP) CYH/MMT/TP/p.1

2 12.2 TCP/IP protocol suite Applications such as file transfers and electronic mail require a reliable service. TCP converts the best-effort service provided by IP into a reliable service. For other applications such as interpersonal applications that involve the transfer of streams of audio or video, a simple best-effort service is acceptable and hence they use UDP. CYH/MMT/TP/p.2

3 With most networked applications, the client-server paradigm is used. All protocol data units (PDUs) relating to the protocols that use the services of the IP layer are transferred in an IP datagram. The protocol field in each IP datagram header is used for IP to identify the protocol to which the contents of the datagram relate. The source and destination port numbers that are present in the header of the PDUs of TCP/UDP are used to identify the application protocol to which the PDU contents relate. Both the source port number and the source IP address from the IP datagram header are used to identify the client by the server. Client port numbers are called ephemeral ports as they are short lived. A new port number is allocated for each new transfer request (1024~5000). The port numbers of the peer application protocols in the server application protocols are fixed and are known as well-known port numbers (0~1023). CYH/MMT/TP/p.3

4 The protocols in both the application and transport layers communicate on an end-to-end basis. The IP protocol in each host has local significance and it talks to its peer protocol in the destination host indirectly. CYH/MMT/TP/p.4

5 12.5 RTP and RTCP When an application involves the transfer of a real-time stream of audio and/or video over a packet network, the timing information that is required by the receiver to output the received packet stream at the required rate is provided by the real-time transport protocol (RTP). For applications that involve both audio and video streams, the real-time transport control protocol (RTCP) is used to synchronize the two media streams prior to carrying out the decoding operation RTP Time is critical in real-time application, so UDP instead of TCP is used. Consequences of using UDP: packet lost, jitter, not in sequence. Missing packets must be detected and compensated for. Delay variations in the packet arrival times must be allowed for. CYH/MMT/TP/p.5

6 Each of the participants that contributes to a multicast call is called a contributing source (CSRC) and is typically identified by the IP address of the source. Streams from different CSRC may be multiplexed together for transmission purposes. To enable the receiver to relate each block/frame to the appropriate participant, the CSRC identifier for each CYH/MMT/TP/p.6

7 block/frame is included in the header of the mixed packet. The number of CSRC identifiers present in the packet is given in the CSRC count (CC) field. Associated with the marker (M) bit is a profile which enables the receiver to interpret the packet data on the correct block/frame boundaries. The payload type field indicates the type of encoder that has been used to encode the data in the packet. The sequence number is used for the destination host to detect lost or out-of-sequence packets. How to deal with lost or out-of-sequence packets: Replacing the lost packet with the last correctly received packet. Buffering a number of packets before playout starts The time-stamp indicates the time reference when the packet was created. Use of time-stamp: Determine the current mean transmission delay, the level of jitter that is being experienced. CYH/MMT/TP/p.7

8 This information, together with the number of lost packets, forms the current QoS of the path through the network. This information is periodically returned to the sending RTP by the related RTCP. The synchronization source (SSRC) identifier identifies the source device that has produced the packet contents RTCP The RTCP adds additional system-level functionality to its related RTP such as the means for a receiving RTP to integrate and synchronize the individual packet streams together and for a sending RTP to be informed of the currently-prevailing network QoS. CYH/MMT/TP/p.8

9 The RTCP operates alongside of RTP and shares information with it. Each RTCP has a different (UDP) port number associated with it so that it can operate independently of RTP. The RTCP in all the systems involved in a call periodically exchange messages with one another. Info that is exchanged: Integrated media synchronization: Let all involved applications use a common clock for synchronization QoS reports: analyzed result of the network's QoS Participation reports: who's in and who's out during a conference call Participation details: information of each participant CYH/MMT/TP/p.9

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