Communication Networks for Multimedia

Communication Networksfor Multimedia

The Evolution of Communication Networks • For over 100 years, the POTS (Plain Old Telephone System) has been the primary focus of conventional voice-band communications • POTS network is well designed and well engineered for the transmission and switching of 3-Khz voice calls – Real-time – Low-latency – High-reliability – Moderate-fidelity

Packet Networks • POTS network is not designed for other forms of communications, such as wide-band speech, audio, images, video, facsimile and data. • About 30 years ago, a second communications network was created with the goal of providing a better transport mechanism for data networking. The resulting network is called a packet network because data is transmitted and routed along the network in the form of units of information

Packet Networks (cont’d) • Packet networks evolved independently of telephone networks for the purpose of moving bursty, non-realtime data among computers. • Packets consist of a header (information about the source and destination addresses) and a payload (actual data being transmitted). • Packet networks are especially well-suited for sending data of various types, including messages, facsimile, and still images. Packet networks are not well suited for sending real-time communication signals such as speech, audio and video.

The Open Systems Interconnect (OSI) Architecture • Physical layer is concerned with the transmission and reception of unstructured bit stream over any physical medium. It deals with the mechanical aspects and signal voltage levels. Examples are RS-232-C and X.21 • Datalink layer ensures reliable transfer of data across the physical medium. It also provides access control to the media in the case of local area networks. Examples are High-level Data Link Control (HDLC), LLC and SDLC • Network layer provides the upper layers with independence from the switching technology. It is responsible for establishing, maintaining and terminating connections. It is also responsible for routing. Examples are X.25 and IP

The OSI Architecture (cont’d) • Transport layeris responsible for reliable and transparent transfer of data between end points, takes care of end-to-end flow control and end-to-end error recovery. An example is TCP • Session layer provides a means for establishing, managing and terminating connections between processes. It may also provide checkpoints, synchronization and restart of service • Presentation layer performs a transformation on the data to provide a standardized interface to applications. It help to resolve the syntactic differences when the internal representation of the data differs from machine to machine • Application layer provides services that can be used by user applications.

Media Access Control • Media Access Control (MAC)* systems may be divided into three categories: – Round robin: each station on the network, in turn, is given an opportunity to transmit. When it is finished it must relinquish its turn and the right to transmit passes to the next station in logical sequence. Control of turns may be centralized or distributed. Token ring is an example of such scheme * MAC: The lower sublayer of the OSIdata link layer. The interface between a node's Logical Link Control and the network's physical layer.

Media Access Control (cont’d) – Reservation. Typically, the time on the medium is divided into slots (time-division multiplexing). To transmit, a station reserves future slots for an extended or indefinite period. Shared satellite channel is an example of this scheme – Contention. No control is exercised to determine whose turn it is to transmit. These methods are likely to lead to collisions and may require retransmission. These techniques perform well when the network load is less, progressively drop off at moderate loads, and perform poorly at high loads. CSMA/CD is an example

Network and Transport Layers • Together, the network and transport layers establish a data pipe between the source computer and a process on the destination computer – the network layer is responsible for setting up routes from a source node to a destination node – the transport layer handles end-to-end issues between processes running on this nodes, such as error control, sequencing, flow control

Unicast and Multicast • Multimedia communications involve two basic modes: unicast and multicast • In unicast mode there are two communication partners, or peers, and the resulting mode is called peer-to-peer communications – Example: individual client-server applications (home shopping, video on demand) • Multicast mode involves 1 to N communications (peer-to-multipeer) as well as 1 to all communications (broadcast mode) – Examples: distance learning, multipeer teleconferences

Routing • Network: graph of nodes (subnetworks) and edges (links between subnetworks) • Problem: find an optimal path from a given source to a given destination node • Routing is the problem of the main task of the network layer and involves two major subproblems: – Find an optimal path in the routing graph, under changing network loads and perhaps even a changing network topology – Get all incoming packets through a router at runtime in an optimal way

Approaches to Routing • Connectionless: the pathfinding algorithm is executed every time a packet is injected into the network (e.g., IP). Each packet finds its way independent of other packets and carries a destination address. – Efficient for short connections (no connect and disconnect phases in the protocol) – Robust in the case of a node failure (no state information stays in the nodes) – Easy internetworking

Approaches to Routing (cont’d) • Connection-oriented (aka virtual circuit): a path from source to destination is computed only once for the duration of a connection. All packets of a connection follow the same path through the graph (e.g., X.25. Frame relay, ATM) – The connection set-up packet leaves a trace with routing information in each node on the path (a connection identifier plus an output port), and all subsequent packets follow the same path. – Efficient routing at runtime (no pathfinding algorithms to be executed during a connection) – The ability to use access control to avoid network congestion (a new call is rejected when the network is overloaded)

Routing Algorithms • Static routing: all routes are pre-computed for a given topology and are independent of the current network load – Each node has a table with entries in the form [source; destination; outgoing link] – An incoming packet contains the destination address (or, in the case of virtual circuits, the connection identifier) – Routing decision is reduced to a quick table look-up – When the network topology changes, a network control center re-computes the global routing table, and the new table is downloaded into all nodes

Routing Algorithms (cont’d) • Adaptive routing: the path-finding algorithm automatically takes into account new or obsolete nodes and links as well as the current load of nodes and links. Each node gets some limited information from neighboring nodes and/or extracts information from packets underway.

Broadcast Routing • Send a distinct packet to each destination – Bandwidth wasteful – Requires the source to have a complete list of all destinations • Flooding – Every incoming packet in a node of the subnet is sent out to every outgoing line except the one it arrived on – Must have a way to dump the number of duplicate packets • E.g., each router can keep track of which packets in a sequence have already been sent

Broadcast Routing (cont’d) • Multi-destination Routing – Each packet contains a list of destinations – New copies of the packets are generated at each router for the output lines that are needed – After a sufficient number of hops, each packet will carry only one destination and can be treated as a normal packet • Spanning Tree Broadcasting – Spanning Tree (ST) = subset of nodes of the subnet that includes all the routers but no loops – If each router knows which of its lines belong to the spanning tree, it can copy an incoming broadcast packet onto the ST lines except the one it arrived from – Most efficient but all routers must know the ST

CBR and VBR Traffic • Multimedia traffic can be characterized as constant bit rate (CBR) or variable bit rate (VBR) • For CBR applications, it is important that the network that transports the data streams has a constant throughput (otherwise, extensive buffering would be required at each end of the system) • VBR traffic often occurs in bursts or spurts (typical case: video compression) – A good measure of burstiness is given by the ratio of peak traffic rate over mean traffic rate over a given period of time

CBR and VBR Traffic (cont’d) • Even with CBR networks, the throughput may vary with time due to the following reasons: – Node or link failure – Network congestion (when the demand for network capacity exceeds the availability) • Throughput decreases with increasing load, especially when bottlenecks are present in the network – Flow control . It is an end-to-end protocol that places limits on the rate of data transmission between two end-systems connected through a network in order to prevent loss of data at the receiving end-system due to buffer overflow

Congestion and Flow Control • Congestion happens when too many packets are present in (a part of) a subnet (performance degrades) • Congestion control: makes sure that the subnet can carry the offered traffic. • Flow control: makes sure that a fast sender cannot continuously send data faster than the receiver can absorb it (involves feedback from the receiver)

Congestion and Flow Control (cont’d) • Examples: – Fiber optic network at 1000 Gb/s on which a fast computer is trying to transfer a file to a PC at 1Gb/s • There is no congestion but flow control is required – Network with 1 Mb/s lines and 1000 computers, half of which are trying to transfer files at 100 Kb/s to the other half • There are not fast senders overpowering slow receivers but the total offered traffic exceeds what the network can handle (congestion)

Congestion Control: Traffic Shaping • Sender and network agree on average rate and burstiness of data transmission – It is not so important for file transfer but very important for real-time data (audio/video) which do not tolerate congestion well • Needs traffic policing to monitor the traffic flow and make sure that the customer is following the agreement

Example: Leaky Bucket • The Leaky Bucket algorithm consists of a token counter and a timer. The counter is incremented by one at each T units of time and can reach a maximum value C. A packet is admitted into the network if and only if the counter is positive. Each time a packet is admitted, the counter is decremented by one. • The traffic generated by a Leaky Bucket regulator consists of a burst of up to C packets followed by a steady stream of packets with minimum inter-packet time of T • Parameters: – Capacity C (packets or bytes) – Flow ρ (packets or bytes per second)

Leaky Bucket (cont’d) • Example: – A computer can produce data at 25 MB/s. – The routers can handle this data rates only for short intervals. For longer intervals, they work best at less than 2MB/s. – Data comes in 1MB bursts (one 40 ms burst every second) – To reduce the average rate to 2MB/s, we could use a leaky bucket with ρ=2MB/s and capacity C=1MB • This means that bursts up to 1MB can be handled without data loss, and that such bursts are spread out over 500 ms, no matter how fast they come in.

Flow Control • Flow control is typically performed using the Sliding Window mechanism – The sliding-window algorithm allows the sender to transmit packets at its own speed until a window of size W is used up. It then has to stop and wait until acknowledgments from the receiver open the window again. – In the TCP protocol, W is not counted in terms of packet but in terms of bytes in transfer

Sliding Window

Requirements and Performance • The three most important parameters of a communication network for multimedia communications are: – Throughput – Error rate – Delay • They form the basic parameters of the Quality of Service (QoS)

Throughput • The throughput of a network corresponds to its effective bandwidth or bit rate, i.e., the physical link bit rate minus the various overheads • Example: ATM technology over a SONET (Synchronous Optical NETwork) fiber optics transmission system. The network carrier’s provisioned bit rate is 155.52 Mb/s. Principal overheads are approximately 3% for SONET and 9.5% for ATM. Thus, the maximum throughput of this network is actually 136 Mb/s • Other factors that affect throughput are network congestion, bottlenecks, node or line faults

Error Rate • Defined in terms of the bit (packet) error rate, i.e., the ratio of the average number of corrupted bits (packets) to the total number of bit (packets) transmitted • Examples: – In fiber optics transmission, the bit error rate range from 10-8 to 10-12 – In satellite transmission systems, the bit error rate is on the order of 10-7

Causes of Errors in Packet-Switching Systems • Individual bits in packets are inverted or lost – Error-correction codes are able to correct the error or detect it and request retransmission – Bit error recovery is based on error detection and retransmission – The sender learns about bit error in one of two ways: • The receiver sends a negative acknowledgment (NACK) • The sender signals a time-out unless a positive acknowledgment is received within a predefined interval

Causes of Errors in Packet-Switching Systems (cont’d) • Packets are lost in transit (inadvertent error), dropped by an intermediate node (deliberate error) or delayed – In a connection-oriented network, when packets are lost or dropped, the receiving end-system is usually able to detect such a situation and inform the sending side • Packet loss recovery is based on sequence numbers – In the case of connectionless networks, packet loss or dropped packets are difficult, if not impossible to detect • The primary reason for packets being dropped or lost in high-speed networks is insufficient buffer space at the receiving end-system due to congestion in the network

Causes of Errors in Packet-Switching Systems (cont’d) • Packets arrive out-of-order – It is the job of the receiving end-system to rearrange the received packets in the numerical sequence in which they were originally sent • IMPORTANT: packet retransmission (especially if it has to be carried out on an end-to-end basis) significantly increases latency – For real-time video or audio transmission, delay is a more important performance issue than error rate, so in many cases it is preferable to forget the error and simply work with the received data stream as is

Delay (Latency) • End-to-end delay is formed by: – Network delay, composed of • transit delay which depends on the physical distance between the two ends • transmission delay which is the time required to transmit a block of data and depends on the bit rate and on processing delays in the intermediate nodes, including routing and buffering – Interface delay, which is the delay incurred between the time a sender is ready to begin ending a block of data and the time the network is ready to transmit the data

Delay (Latency) (cont’d) • For connection-oriented networks, when end-to-end acknowledgments are required, round-trip delay is useful • Round-trip delay is defined as the total time required for a sender to send a block of data through a network and receive an acknowledgment that the block was received correctly

Delay Variation (Jitter) • Extremely important for synchronous multimedia streams (e.g., audio and video) • Network traffic can be: – Asynchronous (no upper bound to the latency) – Synchronous (an upper bound to the latency exists) – Isochronous (there is a constant transmission delay for each message, i.e., if two data streams traverse the network at essentially the same rate and arrive at the destination at the same time) • Isochrony may be recovered by an appropriate playout buffer at the destination

Quality of Service (QoS) • QoS indicates how well a network performs in dealing with a multimedia application • Individual applications have different expectations of how well the network carries out its tasks – Real-time conferencing may impose QoS requirements on latency and throughput – Downloading a video might require small error rates but not have tight restrictions on latency or throughput

QoS (cont’d) ‧ Resource Reservation and Scheduling ‧ If an application knows in advance that it requires certain QoS resources it can make a reservation with the network for those resources for the period in question. The network can either deny the request or schedule the application for that period and reserve the resources requested ‧ Resource Negotiations ‧ If the network administrator feels that the requested resources might overtax the capabilities of the network, it can negotiate with the requester and offer lower QoS parameters. A mutually acceptable set of QoS parameters can then be negotiated.

QoS (cont’d) • Admission Control – If the QoS demands of the particular application are so high that the network cannot meet them, the network has the choice of not letting the application on to the network. ‧ Guaranteed QoS ‧ The user may expect a guaranteed level of service from the network. Whether these guarantees are statistical or absolute depends upon the negotiations between the user and the network.

Example of QoS Requirements for Audio

Example of QoS Requirements for Video

Media Filtering • In a multicast scenario, not all receivers have the same QoS requirements – E.g., a PC connected via a telephone line will not be able to receive video at the same rate as a highend UNIX workstation connected via ATM • A solution: media filtering – The internal network nodes implement media filters, so that the sender needs to create only one flow satisfying the maximum QoS, saving considerable bandwidth

Media Filtering - Example

Media Scaling • A problem with a static “QoS” contract between the sender and receivers of a multicast stream is the variance of many parameters throughout the duration of the transmission, both at the end notes and within the network. • It would be desirable to adjust the QoS parameters during a multimedia connection. When applied to a multimedia data stream, this is called media scaling.

Media Scaling (cont’d) • Media scaling allows control of parameters other than just the data rate (which was already supported by traditional connection-oriented protocols via flow control mechanisms). – Example: image quality in video stream • To implement media scaling, the interface between the application and the network must be extended to pass control information. – Example: if the network signals increasing congestion, an MPEG video encoder can adjust its data rate out via any of its scalability techniques.

Communication Networks for Multimedia