1. 1 School of Computing Science Simon Fraser University CMPT 820: Multimedia Systems Introduction Instructor: Dr. Mohamed Hefeeda

2. 2 Course Objectives r Understand fundamentals of networked multimedia systems r Know current research issues in multimedia r Develop research skills

3. 3 Course Info r Course web page http://nsl.cs.sfu.ca/teaching/10/820/ r References m [SC07] Schaar and Chou (editors), Multimedia over IP and Wireless Networks: Compression, Networking, and Systems, Elsevier, 2007 m [Burg09] Burg, The Science of Digital Media, Prentice Hall, 2009 m [KR08] Kurose and Rose, Computer Networking: A top- down Approach Featuring the Internet, 4th edition, Addison Wesley, 2008 m [LD04] Li and Drew, Fundamentals of Multimedia, Prentice Hall, 2004 m Complemented by research papers

4. 4 Course Info: Grading r Class participation and Assignments: 50% m Few assignments and quizzes m Present one chapter/paper (important) m Read all Mandatory Reading and participate in discussion r Final Project: 50% m New Research Idea (publishable  A+) m Implementation and evaluation of an already-published algorithm/technique/system (Good demo  A+) m Quantitative comparisons between two already-published algorithms/techniques/systems. m A survey of a multimedia topic m … m Check wiki page for suggestions

5. 5 Course Info: Topics r Introduction m Overview of the big picture m QoS Requirements for Multimedia Systems r QoS in the Network m Principles m DiffServ and IntServ r Multimedia Protocols m RTP, RTSP, RTCP, SIP, … r Image Representation and Compression m Sampling, quantization, DCT, compression r Color Models m RGB, CMY, YIQ

6. 6 Course Info: Topics r Video Coding m Compression methods m MPEG compression m Scalable video coding r Error Control for Video Coding and Transmission m Tools for error resilient video coding m Error concealment r Internet Characteristics and Impact on Multimedia m Channel modeling m Internet measurement study r Multimedia Streaming m Fundamentals m On-demand streaming and live broadcast

7. 7 Course Info: Topics r Network-adaptive media transport m Rate-distortion optimized streaming r Wireless Multimedia m WLANs and QoS m Cross-layer design m QoS Support in mobile operating systems

8. 8 Introduction r Motivations r Definitions r QoS Specifications & Requirements

9. 9 Definitions and Motivations r “Multimedia” is an overused term m Means different things to different people m Because it touches many disciplines/industries Computer Science/Engineering Telecommunications Industry TV and Radio Broadcasting Industry Consumer Electronics Industry …. r For users m Multimedia = multiple forms/representation of information (text, audio, video, …)

10. 10 Definitions and Motivations r Why should we study/research multimedia topics? r Huge interest and opportunities m High speed Networks m Powerful (cheap) computers (desktops … cell phones) m Abundance of multimedia capturing devices (cameras, speakers, …) m Tremendous demand from users (mm content makes life easier, more productive, and more fun) m Here are some statistics …

11. Some video statistics r Growth of various video traffic [Cisco 2008][Cisco 2008] m Video traffic accounted for 32% of Internet traffic in 2008 and is estimated to account for 50% in 2012 m Y-axis in Petabytes (1000 Terabytes) per month. 11

12. 12 Some video statistics r YouTube: fastest growing Internet server in history m Serves about 300—400 million downloads per day m Has 40 million videos m Adds 120,000 new videos (uploads) per day r CBS streamed the NCAA March Madness basketball games in 2007 online m Had more than 200,000 concurrent clients m And at peak time there were 150,000 Waiting r AOL streamed 8 live concerts online in 2006 m There were 180,000 clients at peak time r Plus … m All major web sites have multimedia clips/demos/news/broadcasts/…

13. 13 Definitions and Motivations r Given all of this, are users satisfied? r Not Really! m We still get tiny windows for video m Low quality m Glitches, rebuffering m Limited scalability (same video clip on PDA and desktop) m Server/network outages (capacity limitations) r Users want high-quality multimedia, anywhere, anytime, on any device! r We (researchers) still strive to achieve this vision in the future!

14. 14 Multimedia:The Big Picture [SN04]

15. QoS in Networked Multimedia Systems r Quality of Service = “well-defined and controllable behavior of a system according to quantitatively measurable parameters” r There are multiple entities in a networked multimedia system m User m Network m Local system (memory, processor, file system, …) 15

16. QoS in Networked Multimedia Systems r Different parameters belong to different entities  QoS Layers 16

17. 17 QoS Layers User Application System Local Devices Network Perceptual (e.g., window size, security) Media Quality (e.g., frame rate, adaptation rules) Traffic (e.g., bit rate, loss, delay, jitter) Processing (e.g., CPU scheduling, memory, hard drive)

18. QoS Layers r QoS Specification Languages m Mostly application specific m XML based m See: Jin & Nahrstedt, QoS Specification Languages for Distributed Multimedia Applications: A Survey and Taxonomy, IEEE MultiMedia, 11(3), July 2004 r QoS mapping between layers m Map user requirements to Network and Device requirements m Some (but not all) aspects can be automated m For others, use profiles and rule-of-thumb experience m Several frameworks have been proposed in the literature m See: Nahrstedt et al., Distributed QoS Compilation and Runtime Instantiation, IWQoS 2000 18

19. QoS Layers r QoS enforcement methods m The most important/challenging aspect m How do we make the network and local devices implement the QoS requirements of MM applications? r We will study (briefly) m Enforcing QoS in the Network (models/protocols) m Enforcing QoS in the Processor (CPU scheduling for MM) m When we combine them, we get end-to-end QoS r Notice: m This is enforcing application requirements, if the resources are available m If not enough resources, we have to adapt (or scale) the MM content (e.g., use smaller resolution, frame rate, drop a layer, etc) 19

20. 20 QoS Support in IP Networks r Principles r IntServ r DiffServ r Multimedia Protocols r Reading: Ch. 7 in [KR08]

21. 21 QoS in IP Networks: Two Models r Guaranteed QoS m Need to reserve resources r Statistical (or Differential) QoS m Multiple traffic classes with different priorities r In both models, network devices (routers) should be able to perform certain functions (in addition to forwarding data packets)

22. 22 Principles for QoS Guarantees r Let us explore these functions using a simple example m 1Mbps IP phone, FTP share 1.5 Mbps link. m bursts of FTP can congest router, cause audio loss m want to give priority to audio over FTP packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly Principle 1

23. 23 Principles for QoS Guarantees (more) r what if applications misbehave (audio sends higher than declared rate) m policing: force source adherence to bandwidth allocations r marking and policing at network edge: provide protection (isolation) for one class from others Principle 2

24. 24 Principles for QoS Guarantees (more) r Allocating fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn’t use its allocation While providing isolation, it is desirable to use resources as efficiently as possible Principle 3

25. 25 Principles for QoS Guarantees (more) r Basic fact of life: can not support traffic demands beyond link capacity Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs Principle 4

26. 26 Summary of QoS Principles Let’s next look at mechanisms for achieving this ….

27. 27 Scheduling And Policing Mechanisms r scheduling: choose next packet to send on link r FIFO (first in first out) scheduling: send in order of arrival to queue m discard policy: if packet arrives to full queue: who to discard? Tail drop: drop arriving packet priority: drop/remove on priority basis random: drop/remove randomly

28. 28 Scheduling Policies: more Priority scheduling: transmit highest-priority queued packet r multiple classes, with different priorities m class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc..

29. 29 Scheduling Policies: still more Weighted Fair Queuing: r generalized Round Robin r each class gets weighted amount of service in each cycle

30. 30 Policing Mechanisms Goal: limit traffic to not exceed declared parameters Three common-used criteria: r (Long term) Average Rate: how many pkts can be sent per unit time (in the long run) m crucial question: what is the interval length: 100 packets per sec and 6000 packets per min (ppm) have same average! r Peak Rate: e.g., m Avg rate: 6000 ppm m Peak rate: 1500 ppm r (Max.) Burst Size: max. number of pkts sent consecutively (with no intervening idle)

31. 31 Policing Mechanisms Leaky Bucket: limit input to specified Burst Size and Average Rate. r bucket can hold b tokens r tokens generated at rate r token/sec unless bucket full r over interval of length t: number of packets admitted less than or equal to (r t + b).

32. 32 Policing Mechanisms (more) r Leaky bucket + WFQ  provide guaranteed upper bound on delay, i.e., QoS guarantee! How? m WFQ: guaranteed share of bandwidth m Leaky bucket: limit max number of packets in queue (burst)

33. 33 IETF Integrated Services (IntServ) r architecture for providing QoS guarantees in IP networks for individual application sessions r resource reservation: routers maintain state info of allocated resources, QoS req’s r admit/deny new call setup requests:

34. 34 IntServ: QoS guarantee scenario r Resource reservation m call setup, signaling (RSVP) m traffic, QoS declaration m per-element admission control m QoS-sensitive scheduling (e.g., WFQ) request/ reply

35. 35 Call Admission Arriving session must: r declare its QoS requirement m R-spec: defines the QoS being requested r characterize traffic it will send into network m T-spec: defines traffic characteristics r signaling protocol: needed to carry R-spec and T- spec to routers (where reservation is required) m RSVP

36. 36 IntServ QoS: Service models [rfc2211, rfc 2212] Guaranteed service: r worst case traffic arrival: leaky-bucket-policed source r simple (mathematically provable) bound on delay [Parekh 1993, Cruz 1988] WFQ token rate, r bucket size, b per-flow rate, R D = b/R max arriving traffic

37. 37 IETF Differentiated Services Concerns with IntServ: r Scalability: signaling, maintaining per-flow router state difficult with large number of flows m Example: OC-48 (2.5 Gbps) link serving 64 Kbps audio streams  39,000 flows! Each require state maintenance. r Flexible Service Models: Intserv has only two classes. Also want “qualitative” service classes m relative service distinction: Platinum, Gold, Silver DiffServ approach: r simple functions in network core, relatively complex functions at edge routers (or hosts) r Don’t define service classes, provide functional components to build service classes

38. 38 Edge router:  per-flow traffic management  Classifies (marks) pkts  different classes  within a class: in-profile and out-profile Core router:  per class traffic management  buffering and scheduling based on marking at edge  preference given to in-profile packets DiffServ Architecture scheduling... r b marking

39. 39 Edge-router Packet Marking r class-based marking: packets of different classes marked differently r intra-class marking: conforming portion of flow marked differently than non-conforming one r profile: pre-negotiated rate A, bucket size B r packet marking at edge based on per-flow profile Possible usage of marking: User packets Rate A B

40. 40 Edge-router: Classification and Conditioning r Packet is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6 r 6 bits used for Differentiated Service Code Point (DSCP) and determine Per-Hop Behavior (PHB) that the packet will receive r 2 bits are currently unused

41. 41 Edge-router: Classification and Conditioning may be desirable to limit traffic injection rate of some class: r user declares traffic profile (e.g., rate, burst size) r traffic metered, shaped if non-conforming

42. 42 Core-router: Forwarding (PHB) r PHB result in a different observable (measurable) forwarding performance behavior r PHB does not specify what mechanisms to use to ensure required PHB performance behavior r Examples: m Class A gets x% of outgoing link bandwidth over time intervals of a specified length m Class A packets leave first before packets from class B

43. 43 Core-router: Forwarding (PHB) PHBs being developed: r Expedited Forwarding (EF): pkt departure rate of a class equals or exceeds specified rate m logical link with a minimum guaranteed rate m May require edge routers to limit EF traffic rate m Could be implemented using strict priority scheduling or WFQ with higher weight for EF traffic r Assured Forwarding: multiple traffic classes, treated differently m amount of bandwidth allocated, or drop priorities m Can be implemented using WFQ + leaky bucket or RED (Random Early Detection) with different threshold values. See Sections 6.4.2 and 6.5.3 in [Peterson and Davie 07]

44. Protocols For Multimedia Applications r To manage and stream multimedia data r RTP: Real-Time Protocol r RTSP: Real-Time Streaming Protocol r RTCP: Real-Time Control Protocol r SIP: Session Initiation Protocol 44

45. 45 Real-Time Protocol (RTP): FRC 3550 r RTP specifies packet structure for audio and video data m payload type identification m packet sequence numbering m time stamping r RTP runs in the end systems r RTP packets are encapsulated in UDP segments r RTP does not provide any mechanism to ensure QoS m RTP encapsulation is only seen at the end systems

46. 46 RTP Header Payload Type (7 bits): Indicates type of encoding currently being used: e.g., Payload type 0: PCM mu-law, 64 kbps Payload type 33, MPEG2 video Sequence Number (16 bits): Increments by one for each RTP packet sent, and may be used to detect packet loss Timestamp field (32 bytes long). Reflects the sampling instant of the first byte in the RTP data packet. SSRC field (32 bits long). Identifies the source of the RTP stream. Each stream in a RTP session should have a distinct SSRC.

47. RTP Example r consider sending 64 kbps PCM-encoded voice over RTP. r application collects encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk. r audio chunk + RTP header form RTP packet, which is encapsulated in UDP segment r RTP header indicates type of audio encoding in each packet m sender can change encoding during conference. r RTP header also contains sequence numbers, timestamps. 47

48. Real-Time Streaming Protocol (RTSP) r RFC 2326 r client-server application layer protocol r Used to control a streaming session m rewind, fast forward, pause, resume, repositioning, etc… What it doesn’t do: r doesn’t define how audio/video is encapsulated for streaming over network r doesn’t restrict how streamed media is transported (UDP or TCP possible) r doesn’t specify how media player buffers audio/video 48

49. RTSP: out of band control FTP uses an “out-of- band” control channel: r file transferred over one TCP connection. r control info (directory changes, file deletion, rename) sent over separate TCP connection r “out-of-band”, “in- band” channels use different port numbers RTSP messages also sent out-of-band: r RTSP control messages use different port numbers than media stream: out-of-band. m port 554 r media stream is considered “in-band”. 49

50. RTSP Example r metafile communicated to web browser r browser launches player r player sets up an RTSP control connection, data connection to streaming server 50

51. Metafile Example Twister <track type=audio e="PCMU/8000/1" src = "rtsp://audio.example.com/twister/audio.en/lofi"> <track type=audio e="DVI4/16000/2" pt="90 DVI4/8000/1" src="rtsp://audio.example.com/twister/audio.en/hifi"> <track type="video/jpeg" src="rtsp://video.example.com/twister/video"> 51

52. RTSP Operation 52

53. RTSP Exchange Example (simplified) C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Transport: rtp/udp; compression; port=3056; mode=PLAY S: RTSP/1.0 200 OK Session 4231 C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=0- C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=37 C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 S: 200 OK 53

54. Real-Time Control Protocol (RTCP) r Also in RFC 3550 (with RTP) m works in conjunction with RTP m Allows monitoring of data delivery in a manner scalable to large multicast networks m Provides minimal control and identification functionality r each participant in RTP session periodically transmits RTCP control packets to all other participants. r each RTCP packet contains sender and/or receiver reports m report statistics useful to application: # packets sent, # packets lost, interarrival jitter, etc. m used to control performance, e.g., sender may modify its transmissions based on feedback 54

55. RTCP - Continued r Each RTP session typically uses a single multicast address r All RTP/RTCP packets belonging to session use multicast address r RTP, RTCP packets distinguished from each other via distinct port numbers r To limit traffic, each participant reduces RTCP traffic as number of conference participants increases 55

56. RTCP Packets Receiver report packets: r fraction of packets lost, last sequence number, average interarrival jitter Sender report packets: r SSRC of RTP stream, current time, number of packets sent, number of bytes sent Source description packets: r e-mail address of sender, sender's name, SSRC of associated RTP stream r provide mapping between the SSRC and the user/host name 56

57. Synchronization of Streams r RTCP can synchronize different media streams within an RTP session r consider videoconferencing app for which each sender generates one RTP stream for video, one for audio. r timestamps in RTP packets tied to the video, audio sampling clocks m not tied to wall-clock time r each RTCP sender-report packet contains (for most recently generated packet in associated RTP stream): m timestamp of RTP packet m wall-clock time for when packet was created. r receivers uses association to synchronize playout of audio, video 57

58. RTCP Bandwidth Scaling r RTCP attempts to limit its traffic to 5% of session bandwidth. Example r Suppose one sender, sending video at 2 Mbps. Then RTCP attempts to limit its traffic to 100 Kbps. r RTCP gives 75% of rate to receivers; remaining 25% to sender r 75 kbps is equally shared among receivers: m with R receivers, each receiver gets to send RTCP traffic at 75/R kbps. r sender gets to send RTCP traffic at 25 kbps. r participant determines RTCP packet transmission period by calculating avg RTCP packet size (across entire session) and dividing by allocated rate 58

59. SIP: Session Initiation Protocol [RFC 3261] SIP long-term vision: r all telephone calls, video conference calls take place over Internet r people are identified by names or e-mail addresses, rather than by phone numbers r you can reach callee, no matter where callee roams, no matter what IP device callee is currently using 59

60. SIP Services r Setting up a call, SIP provides mechanisms... m for caller to let callee know she wants to establish a call m so caller, callee can agree on media type, encoding m to end call r determine current IP address of callee: m maps mnemonic identifier to current IP address r call management: m add new media streams during call m change encoding during call m invite others m transfer, hold calls 60

61. Setting up a call to known IP address  Alice’s SIP invite message indicates her port number, IP address, encoding she prefers to receive (PCM ulaw)  Bob’s 200 OK message indicates his port number, IP address, preferred encoding (GSM)  SIP messages can be sent over TCP or UDP; here sent over RTP/UDP.  default SIP port number is 5060. 61

62. Setting up a call (more) r codec negotiation: m suppose Bob doesn’t have PCM ulaw encoder m Bob will instead reply with 606 Not Acceptable Reply, listing his encoders Alice can then send new INVITE message, advertising different encoder r rejecting a call m Bob can reject with replies “busy,” “gone,” “payment required,” “forbidden” r media can be sent over RTP or some other protocol 62

63. Example of SIP message INVITE sip:bob@domain.com SIP/2.0 Via: SIP/2.0/UDP 167.180.112.24 From: sip:alice@hereway.com To: sip:bob@domain.com Call-ID: a2e3a@pigeon.hereway.com Content-Type: application/sdp Content-Length: 885 c=IN IP4 167.180.112.24 m=audio 38060 RTP/AVP 0 Notes: r HTTP message syntax r sdp = session description protocol r Call-ID is unique for every call.  Here we don’t know Bob’s IP address. Intermediate SIP servers needed.  Alice sends, receives SIP messages using SIP default port 5060  Alice specifies in header that SIP client sends, receives SIP messages over UDP 63

64. Name translation and user locataion r caller wants to call callee, but only has callee’s name or e-mail address. r need to get IP address of callee’s current host: m user moves around m DHCP protocol m user has different IP devices (PC, PDA, car device) r result can be based on: m time of day (work, home) m caller (don’t want boss to call you at home) m status of callee (calls sent to voicemail when callee is already talking to someone) Service provided by SIP servers: r SIP registrar server r SIP proxy server 64

65. SIP Registrar REGISTER sip:domain.com SIP/2.0 Via: SIP/2.0/UDP 193.64.210.89 From: sip:bob@domain.com To: sip:bob@domain.com Expires: 3600 r when Bob starts SIP client, client sends SIP REGISTER message to Bob’s registrar server (similar function needed by Instant Messaging) Register Message: 65

66. SIP Proxy r Alice sends invite message to her proxy server m contains address sip:bob@domain.com r proxy responsible for routing SIP messages to callee m possibly through multiple proxies. r callee sends response back through the same set of proxies. r proxy returns SIP response message to Alice m contains Bob’s IP address r proxy analogous to local DNS server 66

67. Example Caller jim@umass.edu with places a call to keith@upenn.edu (1) Jim sends INVITE message to umass SIP proxy. (2) Proxy forwards request to upenn registrar server. (3) upenn server returns redirect response, indicating that it should try keith@eurecom.fr (4) umass proxy sends INVITE to eurecom registrar. (5) eurecom registrar forwards INVITE to 197.87.54.21, which is running keith’s SIP client. (6-8) SIP response sent back (9) media sent directly between clients. Note: also a SIP ack message, which is not shown. 67

68. Comparison with H.323 r H.323 is another signaling protocol for real-time, interactive r H.323 is a complete, vertically integrated suite of protocols for multimedia conferencing: signaling, registration, admission control, transport, codecs r SIP is a single component. Works with RTP, but does not mandate it. Can be combined with other protocols, services r H.323 comes from the ITU (telephony). r SIP comes from IETF: Borrows much of its concepts from HTTP m SIP has Web flavor, whereas H.323 has telephony flavor. r SIP uses the KISS principle: Keep it simple stupid. 68

69. Summary: Protocols r Several protocols to handle multimedia data r RTP: Real-Time Protocol m Packetization, sequence number, time stamp r RTSP: Real-Time Streaming Protocol m Establish, Pause, Play, FF, Rewind r RTCP: Real-Time Control Protocol m Control and monitor sessions; synchronization r SIP: Session Initiation Protocol m Establish and manage VoIP sessions m Simpler than the ITU H.323 r NONE enforces QoS in the network 69