Ch 5 : Multimedia Network Standardization, QoS , Access Media

1. Ch 5 :Multimedia Network • Standardization, QoS, Access Media Science and Technology Faculty Informatics Arini, ST, MT arinizul@gmail. Com arinizoel@yahoo.com

2. Contents • Multimedia Presentation • Charactersitic of Multimedia Application • Networked Multimedia Classification • Multimedia Networked • Consideration of Networked Multimedia • Standardizations • QoS • Metrics • Media Access

3. I. Multimedia Presentation • Local vs Networked multimedia • Local Multimedia : • Storage and presentation of multimedia information in standalone computers • Networked Multimedia • Involve transmission and distribution of multimedia information on the network (wired and wireless)

4. II. Networked Multimedia Classification • Real Time: • Require bounds on end-to-end packet delay & jitter. • Subdivided into: • Discrete Media: MSN/Yahoo Messenger • Continuous Media: Continuous message stream with inter-message dependency. Further divided into: • Delay Tolerant (called : streaming) e.g Internet webcast • Delay Intolerant (called : interactive) e.g. audio, video streams in conferencing systems • Non-Real Time: • No strict delay constraints (e.g. text, image files) • May be highly sensitive to errors

5. II. Networked Multimedia Classification • Streaming : • Live media transmission system (live broadcasting) • Send stored media across the network (On Demand file) • Hybrid (Progressive Download) : interactive

6. II. Networked Multimedia Classification • Streaming : • Live media transmission system (live broadcasting) • Capture, compress, and transmit the media on the fly (TV Streaming) • Captured” from live camera, radio, T.V. • 1-way communication, maybe multicast • Examples: concerts, radio broadcasts, lectures • RealPlayer,Media Player and Quicktime • Limited interactivity… • Delays of 1 to 10 seconds or so • Not so sensitive to jitter

7. II. Networked Multimedia Classification • Streaming : • Live media transmission system (live broadcasting) • Send stored media across the network (On Demand file) • Media is pre-compressed and stored at the server. • This system delivers the stored media to one or multiple receivers (video conferencing, Youtube) • Examples: pre-recorded songs, video-on-demand • RealPlayer,Media Player and Quicktime • Interactivity, includes pause, ff, rewind… • Delays of 1 to 10 seconds or so • Not so sensitive to jitter

8. II. Networked Multimedia Classification • Streaming : • Live media transmission system (live broadcasting) • Send stored media across the network (On Demand file) • Hybrid (Progressive Download) : (called : interactive) • 2-way communication • Examples: Internet phone, video conference • Very sensitive to delay < 150ms very good < 400ms ok > 400ms crappy

9. II. Networked Multimedia Classification

10. Data Farms / Storage Edge server Edge server E-Commerce Server Media Server Apps/DB Server Wireless Comm Server Web Server Digital Media: from desktop, to Internet, to hand-helds, to wireless, and to Peer-to-Peer

11. III. Characteristics of multimedia Application Interactive, High Performance, Enriched Media, Large data volume, Real-time property (Continuous display). How to transmit across network? Properties of current Internet Best effort network, cannot guarantee quality of multimedia applications Limitation of bandwidth Heterogeneity Different user requirements Different user network conditions

12. IV. Multimedia Networking • Multimedia networking deals with the design of networks that can handle multiple media types (see II. Networked Multimedia classification) with ease and deliver scalable performance • Points : • Required The Standardization (at a graph bellow) • Technology Multimedia Supporting • Consideration of Networked Multimedia • A measure of the ability of network and computing systems to provide different levels of services to selected applications and associated network flows is Quality of Service (QoS)

14. 4.2. Technologies of Multimedia Networking • Media compression – reduce the data volume Address the1st challenge • Image compression • Video compression • Audio compression • Multimedia transmission technology Address the 2nd and 3rd challenges • Protocols for real-time transmission • Rate / congestion control • Error control

15. 4.3. Consideration of Networked Multimedia • Network Requirements a. Traffic Requirements: Have implications for basic Internet infrastructure • Delay & Jitter • Bandwidth • Reliability b. Functional Requirements: Require enhancements to TCP/IP stack in the form of additional network protocols • Multicasting • Mobility • Session Management

16. 4.3. Consideration of Networked Multimedia • Requirements • Delay & Jitter • Metrics • Packet Processing delay • Packet Transmission delay • Propagation delay • Routing and Queuing Delay • Bandwidth • Reliability • Multicasting • Session Management • Security • Mobility

17. 4.3.1. Delay • Related Metrics • Maximum end to end delay • Delay variance • Jitter : non-monotonic variation in delay in given stream • For a video stream jitter would result in a shaky picture • Jitter can be removed by buffering at the receiver side • Skew: constantly increasing difference between the expected arrival time and the actual arrival time • For a video stream skew could be a slower or faster moving picture

18. 4.3.1. Delay • Packet Processing Delay • Constant amount of delay at both source and destination • A/D, D/A conversion time and time taken to packetize it through different layers of protocols • Typically a characteristic of the operating system and the multimedia application • Delay can become significant under high load conditions • Reductions in delay imply software enhancements including use of multimedia operating systems that provide enhanced resource, file and memory management with real-time scheduling

19. 4.3.1. Delay • Packet Transmission Delay • Time taken by the physical layer at the source to transmit packets. Depends on • Number of active sessions. Typically physical layer processes packets in FIFO order. Delay can become significant if OS does not support real-time scheduling for multimedia traffic • MAC access delay: Widespread Ethernet networks cannot provide any firm guarantees on medium access delay due to inherent indeterminism in CSMA/CD (carrier sense multiple access/collision detection). Isochronous Ethernet (802.9) integrated voice data LAN and demand priority Ethernet (802.12) provide QoS but market potential remains low

20. 4.3.1. Delay • Propagation Delay • Flight time of packets – limited by speed of light. Can’t do anything about it • For a distance of 20,000 km this would be about 0.067 sec • Significant part of a desirable ~200 msec delay budget

21. 4.3.1. Delay • Routing and Queuing Delay • Best-effort Internet treats every packet equally • Packets arriving at a queue have to wait a random amount of time depending on current router load • Delay is variable and is the major contributor to jitter • Techniques to reduce this include • IntServ • MPLS • DiffServ

22. Audio & Video Quality Requirements

23. 4.3.2. Bandwidth Requirements • Multimedia traffic streams have high bandwidth • Uncontrolled transmissions at high rates can cause heavy congestion in the network • Elastic applications that use TCP take advantage of built in congestion control • Most multimedia applications use UDP for transmitting media streams • To remove these shortcomings an enhanced internet service model would require • Admission control: application must first get permission from some authority to send traffic at a given rate with given traffic characteristics • Bandwidth reservation: if admission is given, appropriate resources (buffers, bandwidth) will get reserved along the path • Traffic policing mechanisms: to ensure that applications do not send data at a rate higher than what was negotiated

25. Bandwidth • Text • Bandwidth requirements depend on size • Can be easily reduced by compression techniques • Some text applications require complete freedom from loss & errors – use TCP. E.g. FTP • Others are error and loss tolerant – use UDP e.g. instant messaging

26. Bandwidth • Audio • Bandwidth requirements depend on dynamic range and/or spectrum • Narrowband speech (300-3300Hz) • 6.4 Kbps (G.723.3) to 64 Kbps (G.711) • Wideband audio (CD quality music) 10-220 KHz • 112-128 Kbps (MP3) • Can tolerate 1-2% packet loss • Real-time nature depends on extent of interactivity • VoIP requires strong bounds on delay/jitter (Real-Time Intolerant) • < 250 ms end to end delay • Internet Webcast is more delay/jitter tolerant (Real-Time Tolerant)

27. Bandwidth • Video • High bandwidth requirements • Efficient compression schemes • MPEG-I (1.2 Mbps) VCR quality compression • MPEG-II (3-100 Mbps) broadcast quality video, HDTV • MPEG-IV (64 Kbps) for low bandwidth video compression; supports audio, video, graphics, animation, text • H.261 (px64 Kbps) • H.263 (18-64 Kbps) • Error requirements and real-time characteristics similar to audio

28. Bandwidth • Graphics and Animation • Examples: Digital images, flash presentations • Large in size but lend themselves well to compression • Progressive compression techniques enable image to be initially displayed in low-quality and gradually improved as more information is received • Error-tolerant and can sustain packet loss provided application knows how to deal with packet loss • No real-time constraints

29. 4.3.3. Reliability • Pertains to loss and corruption of data • Can be measured in terms of loss probability • Requires methods for dealing with erroneous/lost data • Error correction • Sender Based Repair • Active : ARQ • Passive : Interleaving, FEC • Error Concealment • Error Recovery for Different Applications • Admission Control • Traffic Shaping/Policing • Packet Classification • Packet Dropping

30. a. Error Correction • Sender Based Repair • Active Repair - Automatic retransmission request (ARQ) • Suitable for error intolerant applications • Passive Repair • Interleaving • FEC: Forward Error Correction. • Media Independent – independent of the content/nature of the stream • Media Dependent - use knowledge of the stream in the repair process • Error Concealment (Receiver Based Repair)

31. Passive Repair : Interleaving • Can be used when media unit size is smaller than packet size (as may be the case with audio) and end-to-end delay is not important • Units are resequenced before transmission so that originally adjacent units are separated by a guaranteed distance and returned to original order at the receiver • Disperses the effect of packet loss – loss of a single packet would causes multiple smaller gaps among original media units • In case of audio a phoneme originally encapsulated in one packet would get split across multiple packets • Loss of small parts of several phonemes is easier to deal with than loss of entire phonemes • Disadvantage: increased latency – not well suited for interactive applications • Advantage: does not increase bandwidth usage – does well for non-interactive use

32. Passive Repair : FEC • Introduce repair data in traffic from which lost packets may be recovered • Media Independent: use block or algebraic codes to produce additional packets which aid in loss recovery • Each code takes a codeword of k data packets and generates n-k additional check packets • i-th bit in check packet is generated from the i-th bits of each associated data packet • Parity Coding: XOR is applied across groups of packets to generate parity packets • Reed-Solomon Coding: Based on properties of polynomials over particular number bases • Take a set of codewords and use these as coefficients of a polynomial f(x) • The transmitted codeword is determined by evaluating the polynomial for all nonzero values of x over the number base • Disadvantage: Cause additional delay, increase bandwidth usage and exacerbate congestion

33. Passive Repair : Media Dependent FEC • Exploit media characteristics • For audio, could send each unit of audio in multiple packets • Primary encoding: first transmission • Secondary encoding: additional transmissions • Secondary encoding could be of lower bandwidth and quality than the primary coding • May not be necessary to transmit FEC for every packet due to nature of media • Advantage : low latency – only single packet delay added • Suitable for interactive applications * A Survey of Packet Loss Recovery Techniques for Streaming Audio, Colin Perkins et al IEEE Network Sep/Oct 1998

34. Error Concealment • Producing a replacement for a lost packet which is similar to the original • Work for relatively small loss rates (< 15%) and for small packets (4-40 ms) • Types in increasing order of computational cost and improved performance: • Insertion based: insert a fill-in packet that contains silence, noise or a repitition of an adjacent packet • Interpolation-based: some form of pattern matching and interpolation to derive the missing packet (waveform, pitch or timescale based) • Regeneration-based: derive decoder state from packets surrounding the loss and generate a lost packet from that (model based recovery)

35. b. Error Recovery for Different Applications • Non-interactive Applications • Multicasts (e.g. radio) • Interleaving is suitable (bandwidth efficient, though high latency) • Use error concealment – repetition with fading • Media-independent FEC better than a retransmission based scheme • Interactive Applications (e.g. IP telephony) • Media Dependent FEC • Error concealment using packet repetition

36. c. Admission Control • Pro-active form of congestion control • Takes requested traffic description as input including (in terms of leaky bucket parameters • Maximum burst size ( b = bucket size) • Peak rate • Average rate • Decides to accept or reject a flow including consideration of impact to existing flows • Admission control unit must also use measurements of current network load and packet delay in its admission decisions

37. d. Traffic Shaping/Policing • Token bucket algorithm is used for traffic shaping. • Limits the average rate and allows a degree of burstiness. • Token bucket depth ‘b’ in which tokens are collected at rate ‘r’ • When bucket becomes full extra tokens are dropped • Source can send data only if it can grab and destroy sufficient tokens from the bucket • Leaky bucket algorithm is used for traffic policing, in which excessive traffic is dropped • Bucket depth ‘b’ with hole at the bottom • If bucket is full extra packets are dropped

38. e. Packet Classification • In order to prevent all packets from being treated equally some mechanism to distinguish between real-time and non-real time packets is needed • Done by packet marking e.g. use Type of Service (ToS) field in IP header • MPLS uses short labels

39. f. Packet Scheduling • FIFO scheduling traditionally used in routers needs to be replaced with more sophisticated queuing • Disadvantage: possible starvation of low priority flows • Weighted Fair Queuing has different queues for different classes. • However every queue is assigned a certain weight. • Packets in that queue get a fraction of the total bandwidth proportional to their weight

40. g. Packet Dropping • Routers can randomly drop packets under congestion • This can be a problem since certain packets may carry more information than others

41. 4.3.4. Multicasting – IP Multicast • Can be done in several ways • Send packets to multicast IP address (Class D) • Hosts willing to receive multicast messages for particular multicast groups inform immediate-neighboring routers using IGMP • Multicast routers exchange group information using a variety of algorithms: • Flooding • Spanning tree • Reverse path broadcasting • Reverse path multicasting • Protocols that use some of these algorithms include • Distance Vector Muticast Routing Protocol (DVMRP) • Multicast extension to Open Shortest Path First (MOSPF) • Protocol Independent Multicast (PIM)

42. 4.3.4. Multicasting – IP Multicast • Application Layer Multicasting • SIP and H.323 support multicasting through a multi-point control unit that provides mixing and conferencing functionality

43. 4.3.5. Session Management • Media Description • Session Description Protocol • Session Announcement • Session Announcement Protocol • Session Control

44. 4.3.5. Session Management • Media Description • Enables application to distribute session information • Media Type • Encoding Scheme • Session Start Time • Session Stop Time • IP Addresses of involved hosts

45. 4.3.5. Session Management • Session Description Protocol • SDP developed by IETF can be used to describe media type, media encoding used for session • More of a description syntax than a protocol – augmented by SIP for media negotiation • Media descriptions encoded in text format • SDP message contains a series of lines called fields with single letter abbreviations. Each field has a <tag>=<value> format • Session Announcement • Allows participants to announce future sessions • E.g. for Internet radio stations to distribute information about scheduled shows

46. 4.3.5. Session Management • Session Announcement Protocol • Used for advertising multicast conferences and sessions • SAP announcer periodically multicasts announcement packets to a well-known multicast address and port (9875) with the same scope as the session being announced • Recipients of announcement are also potential recipients of sessions being advertised • Multiple announcers may announce a single session for more robustness • Announcement interval chosen to ensure total bandwidth used by announcements is below a pre-configured limit • Each announcer is expected to listen to other announcements in order to determine the total number of sessions being announced on a group • Involves large startup delay before complete set of announcements is heard by a listener • Contains mechanisms for ensuring integrity, authenticating the origin and encryption of announcements

47. 4.3.5. Session Management • Session Control • Information in multiple media streams may be inter-related • Network must guarantee to maintain such relationships – Multimedia Synchronization • Can be achieved by putting timestamps in every media packet • Internet multimedia users may want to control playback of continuous media – similar to what a VCR or CD player provides • E.i : • RTP, RTCP, RTSP, H.323, SIP

48. A. Session Control - RTP • RTP runs on top of UDP • Carries chunks of real-time (audio/video) data • Provides • Sequencing: sequence number in RTP header helps detect lost packets • Payload Identification: payload identifier included in each RTP packet describes encoding of the media • Frame Indication: video and audio sent in logical units called frames. A frame marker bit indicates the beginning and end of a frame • Source Identification: To identify the originator of a frame in a multicast session a Synchronization Source (SSRC) identifier • Intramedia Synchronization: To compensate for different delay and jitter for packets within the same stream RTP provides timestamps, which are needed by play-out buffers • Additional media information can be inserted using profile headers and extensions

49. B. Real-Time Control Protocol - RTCP • Real-Time Control Protocol - RTCP • RTCP is a control protocol that works in conjunction with RTP • Provides useful statistics: packets sent, lost, jitter, round-trip time • Sources can use this to adjust their data rate • Other information includes email address, phone number, name – allow users to know the identities of other users in the session

50. C. Real-Time Streaming Protocol • RTSP is an out-of-band control protocol that allows the media player to control the transmission of the media stream including functions such as • Pause • Resume • Repositioning • Playback