INFO 331 Computer Networking Technology II

1
INFO 331Computer Networking Technology II

• Chapter 7
• Multimedia Networking
• Glenn Booker

2
Multimedia Networking

• Recent years have seen massive growth in Internet
  audio and video apps
• Streaming video, IP telephony, Internet radio,
  teleconferencing, interactive games, distance
  learning, etc.
• Older Internet apps (email, WWW, FTP) were very
  elastic in bandwidth needs, but multimedia is
  much fussier delay sensitive
• But theyre also more loss-tolerant

3
Multimedia Networking

• Multimedia apps are in three categories
• Streaming stored audio video (AV)
• Streaming live AV
• Real-time interactive AV
• This excludes download-stuff-and-play, such as
  MP3s, iTunes, etc.
• Download entire file before play starts
• FTP or HTTP are fine for them

4
Streaming stored AV

• Here, clients request on-demand compressed AV
  files that are stored on servers
• Content may include lectures, music, TV, etc.
• CNN video, YouTube, etc.
• Main features of this app type are
• Stored, prerecorded media pause, ffwd, rew
• Streaming, hence can play part of the media while
  downloading more of it
• Continuous play out should keep original timing

5
Streaming live AV

• This is live broadcast of radio or TV over the
  Internet
• Cant fast forward, since it hasnt happened yet,
  but local storage of whats been received can
  allow pausing and rewinding in some cases
• Often accomplished using IP multicasting or IPTV,
  but more often done via separate unicast streams
• Also has continuous play out, can tolerate some
  startup delay

6
IPTV

• IPTV (TV over IP) is a challenge in terms of
  bandwidth
• Traditional client/server cant work
• Often use P2P techniques (see CoolStreaming,
  PPLive), or content distribution networks (CDNs,
  which will be discussed later)

7
Real-time interactive AV

• This class of apps allows interaction between
  people at both ends (or many ends) of a
  connection, such as Internet telephony or
  teleconferencing
• Apps can be integrated, e.g. Web-phone
• Microsoft Live Messenger, Live Meeting (was
  NetMeeting), WebEx, GoToMeeting, Skype
• Transmission delays under 150 ms are good, under
  400 ms is okay, and over 400 ms is bad

8
Multimedia Challenges

• The Internet provides best-effort service
• No guarantees from IP on when stuff will get
  there, consistency in delay times, if it will get
  there, or getting there in order
• All packets are equal in the Internet!
• Streaming stored or live AV has been pretty
  successful, real-time interactive AV less so
• To help, use tricks to make transmission smoother

9
Multimedia Tricks

• At app level, well look at common tricks to make
  multimedia smoother, such as
• Use UDP to avoid TCP congestion control
• Delay playback by 100 ms to help allow for jitter
• Timestamp packets to know when they should be
  played
• Stored data can be fetched in advance, to help
  cover slow periods
• Send redundant data to help cover data losses

10
Fix the Internet!

• Some argue the Internet should allow for
  end-to-end bandwidth guarantees, like virtual
  circuit networks can provide
• Would require massive changes to routers to
  establish fixed paths for some service types
• Hard guarantees mean youre certain to receive
  the QoS you paid for
• Soft guarantees mean youre likely to receive the
  QoS you paid for

11
Fix the Internet!

• Others insist the Internet doesnt need massive
  changes
• Let ISPs upgrade bandwidth as customers demand it
• Increase ISP caching for common requested stored
  AV
• Add content distribution networks (CDNs) for paid
  stored media, conveniently near network edges

12
Fix the Internet!

• Create multicast overlay networks servers
  which help distribute streams to large audiences
• Third approach is to add pricing at the network
  transport layers, to pay for better service
  (Diffserv approach)

13
AV Compression

• Compression helps speed information transmission
  rate by reducing its volume
• Long done for static images (JPG, GIF)
• Each pixel is 24 bits of color data (RGB),
• A 1024x768 pixel image is 1024x768x24/8 2.36
  MB
• Compression can reduce image size a factor of 10
  without severe quality loss
• Key tradeoff more compression more losses
• Huge field (e.g. ISBN 1565921615)

14
Audio Compression

• Digital audio signals have three major
  characteristics
• The sampling rate (samples per second)
• The number of channels sampled (mono, stereo,
  etc.)
• The number of bits per sample
• These all affect the raw amount of data being
  sent, before compression is applied

15
Audio Compression

• Raw audio signals are recorded at some sample
  rate per second, per channel
• 8k, 16k, or 32k samples/sec (Hz) are common low
  grade rates
• 44.1k (CD quality), 48k, 96k, and even 192kHz
  sample rates are used for professional audio
• The number of channels used is typically one
  (mono), two (stereo), or 5 to 7 (surround)

16
Audio Compression

• Each sample is quantized into some number of bits
  to describe its relative strength
• 8 bits gives 256 values from silent to REALLY
  LOUD, which is typical for cheap built-in audio
• CD quality uses 16 bits per sample
• Pro audio typically uses 24 or 32 bits per sample
• So one minute of absurd quality 7 channel
  surround would use 192k sample/sec x 7 channels x
  32 bits/sample / 8 bit/byte 60 sec/min 322.56
  MB/min!

17
Audio Compression

• This approach for audio is formally called Pulse
  Code Modulation (PCM)
• Mono speech recording at 8kHz sample rate and 8
  bits/sample equals 64 kbps of data
• CD quality (44.1 kHz, 16 bit, stereo) 1.411
  Mbps
• Modems cant handle 64 kbps, and most broadband
  users cant consistently get 1.4 Mbps, so
  compression is needed even for just audio

18
Audio Compression

• Common audio compression standards include
• GSM, G.729, and G.723.3 from ITU
• MPEG 1 layer 3, a.k.a. MP3, which compresses to
  96, 128, or 160 kbps
• From Drexel library, can get electronic copy of
  Compression technologies for video and audio,
  Jerry C. Whitaker, ISBN 0071391460

19
Video Compression

• Video is a series of images presented at 24 or 30
  images per second
• Hence the size of the images (X by 0.75X
  pixels), and the color resolution (number of bits
  per pixel) also affect the amount of raw data
• Widescreen image sizes have a 16x9 ratio instead
  of 4x3 ratio

20
Video Compression

• Video compression standards are mostly the MPEG
  family
• MPEG 1 for CD quality video at 1.5 Mbps
• MPEG 2 for DVD quality video, 3-6 Mbps
• MPEG 4 for object oriented video
• H.261 from ITU
• Proprietary formats, such as QuickTime (includes
  MPEG-4 and H.264), Real networks, etc.

21
Streaming Stored AV

• Streaming audio/video has become terribly popular
  (YouTube, AOL Video) because
• Disk space is dirt cheap (lt 15/GB)
• Internet infrastructure is improving
• Theres enormous demand to entertain me NOW
• In this multimedia mode, clients request
  compressed AV files that live on servers
• Files are segmented, with special headers used
  for encapsulating them

22
Streaming Stored AV

• Other examples of streaming stored AV
• Rhapsody, a proprietary client from RealNetworks
• MSN Video, usually played through Windows Media
  Player (WMP)
• Muze, an audio service to retailers

23
Streaming Stored AV

• Various protocols can be used
• RTP Real Time Protocol is used to encapsulate
  the segments
• RTSP Real Time Streaming Protocol is used for
  client/server interaction
• Users typically request files via a web browser
  which has a media player plug-in, such as Flash,
  Quicktime, Shockwave, RealPlayer or Windows
  Media Player

24
Streaming Stored AV

• The media player has several functions
• Decompress the audio or video files
• Buffer the incoming data to smooth out jitter
• Repair damage from lost packets as an attempt at
  error correction
• The media player has a GUI interface, which may
  be integrated into the rest of a web page, or
  appear as a new dedicated window

25
Accessing AV via Web Server

• Streaming stored AV can either be on a web
  server, or on dedicated streaming servers
• In the former case
• A TCP connection is established
• An HTTP message requests the desired file
• The audio file is encapsulated in the HTTP
  response message
• A video file may be separate from audio, so the
  media player may have to assemble them

26
Accessing AV via Web Server

• HTTP can have parallel downloads, so both audio
  and video can be downloaded at the same time
• Or audio and video might be in one file, making
  the process simpler
• The file(s) are passed to the media player, which
  decompresses them and plays them
• But this assumes the entire file is downloaded
  before playing begins grumble

27
Accessing AV via Web Server

• To avoid this, most media players have the server
  send the AV file directly to the media player
  process
• Done via a meta file, which tells what kind of
  file will be streamed
• The process becomes
• User click on link for the desired file
• Hyperlink is to the meta file
• HTTP response contains the meta file

28
Accessing AV via Web Server

• Clients web browser passes meta file to the
  media player
• The meta file tells the browser which media
  player to use!
• Media player sets up TCP connection with the HTTP
  server, and requests the actual AV file
• AV file is sent to the media player, which then
  streams it
• So this could work, but its slow (TCP) and
  doesnt allow pausing or rewinding easily

29
Accessing AV via Streaming Server

• To avoid the slowness of HTTP/TCP, AV can be sent
  via a dedicated streaming server over UDP
• Custom protocols can be used in place of HTTP
• Now one server has HTTP and meta files, and the
  streaming server has the AV files
• These could be two physical servers, or one

30
Accessing AV via Streaming Server

• Now the AV is sent over UDP, at a rate equal to
  its play rate (or drain rate)
• Playout is delayed 2-5 seconds, to allow for
  jitter
• Data is put in a client buffer, from which its
  played after the delay
• Or could send AV data over TCP, to get better
  sound quality, at the risk of sound pauses
  (client buffer starvation)

31
Accessing AV via Streaming Server
32
Client Buffer
33
RTSP

• The real time streaming protocol (RTSP) allows
  the user to control playback of audio or video
• Defined by RFC 2326
• It does NOT define compression schemes to be
  used, encapsulation (see RTP), transport protocol
  (TCP or UDP), or buffering approach
• Those are all application-level concerns

34
RTSP

• So what DOES it do?
• Allow user to pause/resume, reposition playback,
  fast forward, or rewind
• RTSP sends control messages out-of-band, using
  port 544, could be over TCP or UDP
• Recall FTP also used out-of-band control messages
• The actual media stream is a separate band
• Like HTTP, commands are plain text, with
  code-identified responses
• But RTSP maintains session client state
  information

35
RTSP Example

• An HTTP GET command requests the presentation
  (meta) file from the web server
• Then passes it to the media player to manage
  getting the media and playing it
• In this example, the audio and video files are
  separate, but are played together (lipsynch)

36
Meta file example

• lttitlegtTwisterlt/titlegt
• ltsessiongt
• ltgroup languageen lipsyncgt
• ltswitchgt
• lttrack typeaudio
• e"PCMU/8000/1"
• src
  "rtsp//audio.example.com/twister/audio.en/lofi"gt
  
• lttrack typeaudio
• e"DVI4/16000/2"
  pt"90 DVI4/8000/1"
• src"rtsp//audio.ex
  ample.com/twister/audio.en/hifi"gt
• lt/switchgt
• lttrack type"video/jpeg"
• src"rtsp//video.ex
  ample.com/twister/video"gt
• lt/groupgt
• lt/sessiongt

37
RTSP Example
38
RTSP

• Many RTSP methods are pretty self-explanatory
  PLAY, PAUSE, RECORD
• SETUP establishes the connection
• TEARDOWN closes the connection
• DESCRIBE identifies the media to be played
• ANNOUNCE updates the session description
• RTSP is used by Real Networks

39
Internet Phone

• Real-time apps, such as Internet phone and video
  conferencing, are very sensitive to packet delay,
  jitter, and packet loss
• See how these issues are handled for Internet
  phone
• Data is generated at 8 kBps
• Every 20 ms, a chunk of 160 bytes of data gets a
  header attached, and the packet is sent via UDP
• Ideally, these packets all get to the receiver

40
Internet Phone

• The receiver must decide
• When to play back a given packet
• What to do if a packet is missing
• Recall packets can be lost if they arrive at a
  router with a full input or output queue
• From 1-20 packet loss can be tolerated
• Forward error correction (FEC) can help make up
  for lost packets

41
Internet Phone

• End-to-end delay can be confusing for the
  receiver, if it simply takes too long for the
  data to get there
• Recall the under 150 ms, 150-400 ms, and over 400
  ms ranges for good, ok, and bad delays
• Internet phone may discard packets over 400 ms
  old
• Jitter is typically caused by variation in
  queuing delays

42
Internet Phone

• Some jitter problems can be removed by using
  sequence numbers, time stamps, and playout delays
  to make sure packets are in order as well as
  possible
• Playout delay can be fixed or adaptive
• For fixed playout delay, anything arriving after
  its planned play time is discarded
• The amount of delay is typically 150-400 ms, less
  under good network conditions

43
Adaptive playout delay

• Adaptive playout delay is often used because long
  delay is annoying to the users
• Want to make the delay as small as possible
  without losing a lot of packets
• Delay is reassessed for each talk spurt (period
  of transmission)
• Similar to calculation of timeout interval for
  TCP, use previous history, amended by the
  most recent data

44
Adaptive playout delay

• di (1-u)di-1 u(ri-ti)
• Where d is the average network delay
• u is a fixed value, e.g. 0.01
• ri is when the ith packet was received
• ti is the time that packet was timestamped by
  the sender
• Similarly the average deviation is
• vi (1-u)vi-1 uri ti di

45
Adaptive playout delay

• Then the playout time for the first packet in a
  talk spurt is
• pi ti di Kvi
• Where K is typically 4, much likeTimeoutInterval
  EstimatedRTT 4DevRTT
• This gives us an approach to calculate the
  playout delay, and keep adjusting it to
  compensate for network traffic conditions

46
Stored vs real time AV

• Streaming stored audio video, in contrast, uses
  the same techniques to help smooth out network
  jitter
• Sequence numbers, timestamps, playout delay
• But stored AV can tolerate much larger delays
  before playout begins (compared to real time
  AV), which gives the app developer much more
  design flexibility

47
Recovering from packet loss

• The method for compensating for lost packets is
  the loss recovery scheme
• Here, lost means it never got there, or it got
  there after its planned play time
• Retransmitting lost packets doesnt make sense
  here (why?)
• Instead anticipate loss, using methods like
• Forward Error Correction
• Interleaving

48
Forward Error Correction

• Forward Error Correction adds a little redundant
  data to packets to help allow recreating what
  missing packets had in them
• RAT (Robust Audio Tool) is an open source
  Internet phone app that uses FEC and RTP
• There are lots of FEC approaches well look at
  two of them

49
Forward Error Correction

• First is to send a redundant encoded chunk of
  data after every n normal chunks
• The redundant chunk is obtained from XOR-ing the
  normal chunks
• If any ONE of the n chunks is missing, it can be
  reconstructed mathematically
• But if two or more packets are lost, tough luck
• This increases transmission rate 100/n

50
Forward Error Correction

• The second approach is sneakier
• Add a second, lower quality data stream in
  parallel with the primary stream (e.g. tack on a
  13 kbps stream to a 65 kbps stream)
• When loss occurs, use the low quality stream
• This increases playout delay little
• Many variations on this approach are possible,
  especially to allow for higher loss rates

51
Interleaving

• Interleaving breaks the stream of data into
  smaller chunks, and rearranges how the chunks are
  sent
• So if each chunk of data is broken into four
  pieces, then the first unit of data sent has
  chunks number 1, 5, 9, and 13 instead of just
  1-4
• The second chunk sent has 2, 6, 10, 14 the
  third chunk has 3, 7, 11, 15, etc.
• That way if a chunk is lost, the loss is
  distributed across a wider range of time

52
Interleaving

• This increases latency (have to break up
  reassemble chunks)
• Doesnt increase bandwidth of data no extra
  data is being sent
• So how can we fix the data if some is lost?

53
Repair of damaged audio

• Voice is relatively easy to fix if a little data
  is missing
• Loss rates under 15 for packets 4-40 ms long
• One technique is simply repetition copy the
  previous packet and play it again
• Easy to do and usually works ok
• Or can try to interpolate between before and
  after packets
• More tricky computationally, but sounds better

54
Content Distribution Networks

• Content Distribution Networks (CDNs) are to
  address the need for lots of people accessing the
  same stored media often and/or at once
• After all, a single server source would have
  terrible bandwidth and packet loss issues
• Often a content provider (CNN, MSN, etc.) will
  pay a CDN company (Akamai) to provide videos to
  users with as little delay as possible

55
Content Distribution Networks

• The CDN approach is simple make lots of copies
  of the media, and put them on lots of servers
  everywhere
• CDN servers get put throughout the Internet
• Often the CDN company will lease data center
  space to house the servers
• Data centers might be located at second or third
  tier ISPs

56
Content Distribution Networks

• The CDN gets source media (videos) from the
  customer, and copies them to the servers
• When a user requests content, the nearest (or
  most available) CDN server delivers it
• DNS redirection is used to find the correct CDN
  server
• This results in URLs with two addresses in them
  the first is the CDN, the second is the file name
• http//www.cdn.com/www.cnn.com/zoo/turtle.mpg

57
Content Distribution Networks

• The best server for each ISP is determined
  using the same approaches we saw for BGP routing
  tables
• CDNs may also be used within a large corporation
  to stream, for example, training videos locally

58
Dimensioning best-effort networks

• Many techniques weve seen can be used to improve
  the quality of service of multimedia apps
• We want low RTT, little jitter, few packets lost
• Another solution is to throw enough money at the
  network to avoid all these problems
• Given enough resources, none of these problems
  occur!

59
Dimensioning best-effort networks

• To do so requires adequate bandwidth
  provisioning, and at a larger scale, good network
  dimensioning issues include
• Model traffic demand between network end points
• Clear performance requirements
• Model performance for a given workload
• Model to predict minimum bandwidth to meet all
  requirements

60
Real-time Interactive Protocols

• Demand for real time interactive applications is
  huge, so there has been a lot of work on
  protocols to help make such apps easier to
  develop
• Here well look at three common ones
• RTP, the Real-time Transport Protocol
• SIP
• H.323

61
RTP

• RTP defines a standard AV chunk-of-data structure
  for data, sequence numbers, timestamps, etc.
• The actual data format can be a proprietary
  format, or other AV formats such as PCM, GSM,
  MP3, MPEG, H.263, etc.
• RTP is defined by RFC 3550, and usually runs over
  UDP (no specific port number)
• Often used for Internet telephony apps

62
RTP
63
RTP

• An RTP packet consists of the RTP header plus the
  data
• The RTP header is at least 12 B (3 rows x 4B ea.)
• Apps that both use RTP have a better chance of
  cooperating (e.g. if two users have different
  apps at each end of a phone call)
• RTP doesnt guarantee data delivery, or sequence
• Routers cant tell RTP data from anything else

64
RTP

• Every data source (video camera, microphone) can
  have an RTP data stream
• Video conference could have four RTP streams to
  send video each direction, and audio each way
• Some compression formats (MPEG-1 and -2) combine
  audio and video into one stream
• RTP can be used with unicast or multicast
• A group of streams from the same origin form an
  RTP session

65
RTP Header Fields

• The RTP header is very simple
• First line has
• Nine bits of misc. identifiers (version, how long
  the CSRC list will be, if custom extensions are
  used, etc.)
• Payload type (7 b)
• Sequence number (16 b)
• Timestamp (32 b)
• SSRC (32 b)
• CSRC list (0 to 15 lines of 32 b each)

66
RTP Header

• The Payload Type is a numeric code that
  identifies what format the data has PCM, GSM,
  MPEG, etc.
• The Sequence Number starts at a random value, and
  increments by one for each RTP data packet sent
  RFC 3550
• The Timestamp is when the first data was sampled
  change from one to the next is inversely
  proportional to the sampling rate

67
RTP Header

• The SSRC is the synchronization source identifier
  just a random number to identify the source
• Only goal is that no two SSRCs are the same in a
  session
• The CSRC list is the contributing sources for
  audio data
• Is often used for mixing, or to identify specific
  data sources (Fred vs. Wilma)

68
RTP App Development

• RTP can be implemented two ways
• Use RFC 3550 and manually implement RTP headers
  within your application
• Use existing API libraries (e.g. in C or Java) to
  implement RTP for you, and have your app call
  them
• Hmm, I wonder which is easier?
• RTP can be coded as an app-layer protocol, or
  used in conjunction with UDP via the API

69
RTP Control Protocol (RTCP)

• RTCP is also defined by RFC 3550
• It defines control packets that are sent by data
  senders and receivers in an RTP session via IP
  multicast
• Main purpose is to provide feedback on the
  quality of the data distribution
• It identifies the RTP source by a canonical name
  or CNAME
• Control rate of data transmission, identify
  users

70
RTCP

• RTCP uses the next port number above that used by
  RTP
• RTCP packets are sent periodically by all session
  systems
• They have no data attached just headers
• Sender Report (SR) has SSRC of the session,
  timestamp of most recent RTP packet, of packets
  and bytes sent in the stream

71
RTCP

• Sender reports can synch streams in a session
• The Reception Report (RR) has reception
  statistics
• SSRC of the RTP stream of interest
• Fraction of packets lost
• Last sequence number received
• Interarrival jitter
• There are source description packets, giving
  user name, email, app, SSRC

72
RTCP

• RFC 3550 has massive detail on the statistics
  which can be collected (bandwidth, packet size,
  jitter, transmission interval, packets lost,
  etc.) but leaves analysis and interpretation of
  the data to the app developer
• Sender and receiver reports can be stacked into a
  combined report

73
RTCP Scaling

• The amount of RTP traffic doesnt change with a
  large number of receivers, but the amount of RTCP
  traffic does grow
• RTCP tries to limit itself to 5 of session
  bandwidth(!)
• This results in different RTCP transmission
  periods (T) for senders versus receivers
• Two senders and 100 receivers will get
  T(receiver) 16.67T(sender)

74
SIP

• The Session Initiation Protocol (SIP) is defined
  by RFC 3261
• SIP can be over UDP or TCP, and uses port 5060
• It is designed to handle Internet telephone
  across LANs (not local phone exchanges)
• It allows calls to be placed over IP
• Allows caller to determine IP of callee (varies)
• Can add media streams during call

75
SIP Session Example

• Caller sends an INVITE message
• Tells caller and callee IP addresses, media
  format and encoding, type of encapsulation (RTP)
  and receiving port number
• Callee sends response msg
• Confirms IP address, encoding, encapsulation, and
  port number
• Caller sends ACK to callee, and the call can take
  place

76
SIP Session Example
77
SIP

• So what?
• The two parties can be using completely different
  encoding and encapsulation methods, yet carry on
  the call
• Also note that the control messages stay over
  port 5060, yet the media are over two other,
  negotiated ports so SIP uses out-of-band
  control messages
• SIP requires ACK of all messages, so either UDP
  or TCP can be used

78
SIP Addressing

• SIP addresses can look like email addresses, or
  cite IP addresses, or be a phone number, or even
  a full personal name
• bob_at_mydomainnotyours.com
• bob_at_213.33.129.97
• 12125551212_at_phone2net.com

79
SIP Messages

• The SIP protocol is huge 269 pages so only a
  brief overview is in order
• The INVITE message typically is addressed to an
  email-like address, e.g. bob_at_domain.com
• Each device the message passes through adds a
  Via header with that devices IP
• A SIP proxy finds the IP of the device the
  callee is currently using

80
SIP Messages

• Every SIP user is associated with a SIP
  registrar
• When a SIP device is launched, it registers with
  the registrar to give its current IP address
• So the SIP proxy asks the callees registrar for
  their IP address
• The proxy might be redirected to another
  registrar if the callee is not nearby

81
SIP applications

• SIP has been described for voice over IP, but can
  also be used for any media video, even text
  messaging
• SIP software is widely available for many common
  functions, on various platforms (PC/Mac/Linux)

82
H.323

• H.323 (catchy name, huh?) is an alternative to
  SIP
• Computers can use SIP or H.323 to connect to
  plain telephones, as well as do IP-only calling
• H.323 supports audio and optionally, video
• Audio must support at least G.711 speech
  compression at 56 or 64 kbps

83
H.323

• Video, if used, must support at least QCIF H.261,
  which is a massive 176x144 pixels
• H.323 is a suite of protocols
• Includes a separate control protocol (H.245),
  signaling channel (Q.931), and uses RAS for
  registering with the gatekeeper
• The gatekeeper is between the IP and plain
  telephone networks
• Whereas, SIP only manages connections

84
H.323 vs SIP

• H.323 is from the ITU (mainly telephone basis),
  whereas SIP is from IETF (the RFC folks)
• SIP can work with RTP, G.711, and H.261, but
  doesnt require any of them
• SIP is simple compared to H.323

85
Beyond Best Effort

• So lots of techniques have been used to get the
  most out of the Internets best effort approach
• The current quality of service (QoS) has no
  guarantees
• How can we improve on that?
• Change the architecture of the Internet!
• Look at a simple network to examine the
  problems

86
A Sample Network
87
A Sample Network

• In this example, the local networks are assumed
  much faster than the 1.5 Mbps connection between
  them
• Two apps are competing for that 1.5 Mbps of
  bandwidth, e.g. leaving the first router (R1)
• Scenario 1 If hosts H1 is sending FTP to H3,
  and then H2 tries to send audio traffic to H4,
  the router R1 will already be full, making for
  lost audio packets

88
A Sample Network

• But IF we could mark audio packets versus FTP
  packets, we could give priority to audio packets,
  since they are delay sensitive
• Principle 1 packet marking lets a router
  distinguish between different classes of traffic
• Scenario 2 what if the FTP traffic was on a
  high priority (paid) connection, but the audio
  traffic was normal free Internet
• Principle 1a packet classification lets a router
  distinguish between different classes of traffic

89
A Sample Network

• The distinction is important
• Marking packets for type of payload is one way to
  classify them, but hardly the only one
• The packets could be prioritized by the
  originating IP address (which could imply whether
  premium service was used)
• Scenario 3 Bad application
• What if the audio app should get priority, but
  it abuses that and hogs the whole link?

90
A Sample Network

• Instead of allowing 1.0 Mbps for audio, the app
  tries to take the whole 1.5 Mbps, thereby
  shutting off the FTP service
• Just like circuit switching, we want to isolate
  each traffic flow, so a misbehaving app doesnt
  ruin the link for everyone
• Principle 2 Some degree of isolation is
  desirable to protect traffic flows from each
  other
• This might imply the need for traffic cops, to
  ensure app compliance

91
A Sample Network

• So two approaches could address these concerns
  and improve the QoS
• Mark packets, and police app behavior to share
  bandwidth fairly
• Or, logically isolate traffic flows, and
  designate some specific bandwidth for each
• But if the flows are isolated, need to make sure
  full bandwidth is used if a traffic flow
  isnt active

92
A Sample Network

• Principle 3 When traffic flows are isolated,
  need to use resources (bandwidth, router buffers)
  as efficiently as possible
• Scenario 4 Two 1.0 Mbps audio apps over a 1.5
  Mbps link
• Both audio apps cant get the full bandwidth
  needed each would lose 25 of their packets
• To maintain any decent QoS level, the network
  should allow or block the flow
• Telephones have done this for years!

93
A Sample Network

• So to guarantee a minimum level of QoS we need a
  call admission process
• Principle 4 a call admission process is needed
  to admit or block calls from the network, based
  on the calls QoS requirements
• Now that we have the principles of guaranteeing
  QoS, look at how these are implemented in
  the Internet

94
Scheduling and Policing

• Recall at the link level, packets from various
  flows are multiplexed and queued on the output
  buffers to go onto a link
• The link-scheduling discipline is the process for
  queuing packets, which is vital for making QoS
  guarantees
• Look at FIFO, priority, round robin, and WFQ
  approaches for link-scheduling

95
FIFO Scheduling

• FIFO (first-in first-out), also called first-come
  first-served (FCFS) is the McDonalds approach to
  scheduling
• Packets are queued in the order in which they
  arrived at the device
• If the output buffer gets full, the
  packet-discarding policy is used to decide which
  packets are dropped

96
Priority Queuing

• Here, packets are grouped by their priority a
  separate queue for each level of priority
• How they are grouped could be by type of service,
  source IP, or other methods
• The highest priority queue is emptied before the
  next priority queue is transmitted
• If the high priority queue is empty, a lower
  priority packet will still be sent immediately
• Within each queue, FIFO applies

97
Round Robin Queuing

• In round robin, packets are sorted into classes
  which have no relative priority
• One packet from class 1 is sent, then one from
  class 2, etc.
• Then go back to class 1 and keep repeating the
  cycle
• A work-conserving round robin tries to keep the
  link busy, so will check the next class if
  one is empty

98
Weighted Fair Queuing (WFQ)

• WFQ is a variation on round robin queuing
• Here each class is assigned a weight, which
  determines how much transmission time they get
• In short, some classes are more equal than others
  
• The weights for all classes total 100
• Incoming packets are classified into the correct
  class, which has its own queue
• WFQ is very widely used

99
Policing

• We may need to control the rate data are allowed
  to enter the network a traffic cop
• Could control three aspects
• Long term average data rate
• Peak or max rate
• Burst size max number of packets in a short
  time interval

100
The Leaky Bucket Analogy

• Bucket holds b tokens (packets)
• New tokens are created at rate r tokens per sec
• If bucket is full, new tokens are discarded

101
The Leaky Bucket Analogy

• The token generation rate, r, limits the long
  term average rate
• In any time t, the max number of tokens which
  could be added to the network is rt b, so
  that limits the peak rate
• The size of the bucket limits the max burst size
  
• No more than b tokens can be removed in a short
  time

102
WFQ Leaky buckets

• If we combine the WFQ scheduling with the leaky
  bucket policing options, what happens to the
  delay for any class of packets?
• If the class bucket is full, and b1 packets
  arrive, the rate the bucket is emptied is at
  least Rwi/sum(w) where R is the link rate, wi is
  the weight of this class, and sum(w) is the sum
  of all active classes weights

103
WFQ Leaky buckets

• So the time for the bi packets to be handled is
  time quantity/rate
• di bi/Rwi/sum(w)
• Hence there is a provable maximum delay
• As long as the rate this class can handle packets
  is less than Rwi/sum(w), the above di equation
  is the max delay in a WFQ queue

104
Integrated vs Differentiated Services

• Now weve covered the types of mechanisms needed
  to provide QoS guarantees in the Internet
• Cool. So, how is this being implemented?
• Two major architectures are trying to provide an
  answer Intserv and Diffserv
• Integrated Services Intserv
• Differentiated Services Diffserv

105
Intserv

• Intserv provides a way for QoS to be specified
  for a given session
• Kind of like setting up a virtual circuit
• Intserv depends on two key features
• Must know what resources are already occupied by
  routers along a desired path
• Must do call setup to prepare for a session

106
Intserv

• Call setup includes
• A session must define the QoS needs
• An Rspec defines the type of QoS service desired
• A Tspec defines the traffic the session will be
  sending
• See RFC 2210 and 2215 for Rspec and Tspec
• Then a signaling protocol, RSVP, is used to set
  up along the paths routers
• Each router (element) decides whether to allow
  the session

107
Intserv

• Intserv provides two different types of QoS
• Guaranteed QoS the bounds on queuing delays for
  each router are assured
• See RFC 2212 for gory details how this is done
• Controlled-load Network Service the session
  will receive a close approximation to the QoS
  an unloaded element would provide
• Oddly, this QoS is not quantified
• RFC 2211, since you were wondering

108
Diffserv

• Diffserv (RFC 2475) was in response to problems
  with Intserv
• Scalability problems with per-flow (session)
  reservations
• Gets very time consuming for large networks
• Flexibility in service classes Intserv only
  provides specific service classes
• Want ability to grade or rank service classes
  (low/med/high, silver/gold/platinum, etc.)

109
Diffserv

• Diffserv assumes the whole network will be aware
  of Diffserv protocols, and that most traffic
  flows are under Diffserv
• You will be assimilated!
• Two main functions under Diffserv
• At the edges of the network, hosts or first hop
  routers mark the DS field in the packet header
• Within the core of the network, traffic is
  handled based on its class per-hop behavior

110
Diffserv

• Analogy
• The marking process is like getting your hand
  stamped at a night club, or having a VIP or
  backstage pass, or getting comped for spending
  too much money at a casino
• Packets are classified based on source or
  destination IP or port, or (transport) protocol
• There could be different marking algorithms for
  various classes of flows

111
Diffserv

• Classes of flows could follow a traffic profile
• May limit their peak transmission rate, or
  burstiness
• A metering function could be used to control flow
  rate into the network, or shape flow rates
• The differentiated services (DS) field (RFC 3260)
  replaces the type of service and traffic class
  fields from IPv4 and IPv6
• The packets are marked using the DS field

112
Diffserv

• Within the network, each class per-hop behavior
  (PHB) is critical to controlling its handling
• PHB influences the forwarding behavior
  (performance) at a Diffserv node
• No method for implementing behavior is given
• Only key is that they are externally measurable
• Two PHBs have been defined expedited forwarding
  (EF), and assured forwarding (AF)

113
Diffserv

• Expedited forwarding (EF, RFC 3246) requires the
  departure rate of a class must equal or exceed a
  certain rate
• Implies isolation from other traffic classes,
  since the absolute rate is guaranteed
• Assured forwarding (AF, RFC 2597) divides traffic
  into four classes
• Each is guaranteed bandwidth and buffer space
• Per class there are three levels of drop
  preference

114
Intserv, Diffserv, and Reality

• So while techniques exist to provide QoS
  guarantees, they arent widely used
• Why? The Internet is highly distributed, and not
  everyone will agree 1) to implement QoS methods,
  2) on the type of QoS needed, and 3) on how to
  pay for it
• And if traffic levels are low, best effort will
  provide the same service as Intserv/Diffserv!

115
RSVP Protocol

• In order to reserve resources in the Internet, a
  signaling protocol is needed
• The Resource reSerVation Protocol (RSVP, RFC
  2205) does so
• Here we focus on bandwidth reservation, but RSVP
  can be used for other applications
• To work, hosts and routers need to implement the
  RSVP protocol

116
RSVP Protocol

• RSVP provides reservations for bandwidth in
  multicast trees (unicast is easy then)
• The receiver of the data controls getting the
  reservations (not the sender)
• In a session, each sender can control many flows
  (audio and video, e.g.)
• Each flow has the same multicast address
• To distinguish flows, use the flow identifier
  (IPv6)

117
RSVP Protocol

• RSVP does not specify how the bandwidth is
  obtained see earlier scheduling methods
  (priority scheduling, WFQ, etc.)
• RSVP depends on other routing protocols to
  determine the path needed, and hence the specific
  links in the path
• If the path changes during the session, RSVP
  re-reserves the path

118
RSVP Protocol

• In multicast, receivers could have many different
  speeds dial up, cable, DSL, etc.
• Sender doesnt need to know the rate of all
  receivers, only the rate of the fastest one
• Encode the media in layers at different data
  rates (e.g. 20 kbps 100 kbps)
• A slow base layer (dial up), and add other layers
  to enhance the image/signal for faster receivers
• Each receiver picks up the layers they can use

119
RSVP Protocol

• The source of a multicast advertises content
  using RSVP path messages through a multicast
  tree
• Tells the bandwidth required, timeout interval,
  and upstream path to the sender
• Receivers send RSVP reservation messages
• Tells what rate they want the data
• Message is passed upstream to the source, where
  each router allocates the bandwidth

120
RSVP Protocol

• Routers can merge requests for the same multicast
  event saves having many reservations for the
  same signal!
• Hence routers only need to reserve the maximum
  downstream receiver rate
• Routers only send one reservation message
  upstream

121
RSVP Protocol

• In a conference call situation, all participants
  are typically senders and receivers of video, so
  the bandwidths are added
• For audio only, its rarer for many to speak at
  once, to typically if one user is transmitting at
  rate b bps, reserving 2b bps is typically
  adequate bandwidth

122
RSVP Protocol

• When a router gets a reservation request, it must
  do an admission test
• If there isnt enough bandwidth to handle the
  reservation, the router rejects the reservation
  and returns an error message to the receiver
• RSVP is very receiver-oriented, since they
  request reservations

123
Soft Hard State Reservations

• In general, a signaling protocol can have soft or
  hard state reservations
• Soft state reservations will time out unless
  renewed periodically
• You have to say Im alive! or Ill assume
  youre dead
• A system could crash and reboot, and it wouldnt
  lose its reservation, if it responded fast enough
• RSVP, PIM, SIP, and IGMP use soft state

124
Soft Hard State Reservations

• Hard state signaling means the reservation stays
  forever, until explicitly removed by an
  uninstall, delete, tear-down, or other message
• Requires a means to look for orphaned states
  (its installer is dead or gone)
• Typically used for reliable systems (not best
  effort) so few Internet protocols use it
• RSVP has optional removal of reservations

125
Summary

• Multimedia is huge in the Internet, and getting
  bigger!
• May replace circuit switched telephones
• Here we reviewed the types of multimedia apps,
  looked at methods for best-effort networking,
  added methods for QoS guarantees, and outlined
  architectures for providing QoS