Multimedia Networking

1
Multimedia Networking

• Network Requirements and Protocols
• Ahrar Naqvi
• ahrar_at_palmchip.com

2
Agenda

• Overview of media characteristics from a
  networking standpoint
• Network requirements for multimedia
  communications
• Network architectures, techniques and protocols

3
Multimedia Classification

• Real Time Require bounds on end-to-end packet
  delay jitter. Subdivided into
• Discrete Media MSN/Yahoo Messenger, Stock quotes
• Continuous Media Continuous message stream with
  inter-message dependency. Further divided into
• Delay Tolerant e.g Internet webcast
• Delay Intolerant 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

4
Key Media Characteristics from Networking
Standpoint

• Bandwidth usage
• Sensitivity to error
• Real-time nature

5
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

6
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)
• Internet Webcast is more delay/jitter tolerant
  (Real-Time Tolerant)

7
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

8
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) video over ISDN
• H.263 (18-64 Kbps) video over POTS
• Error requirements and real-time characteristics
  similar to audio

9
Network Requirements

• Traffic Requirements Have implications for basic
  Internet infrastructure
• Limits on Delay Jitter
• Bandwidth
• Reliability
• Functional Requirements Require enhancements to
  TCP/IP stack in the form of additional network
  protocols
• Multicasting
• Single source of communication with multiple
  simultaneous receivers
• Can be one way (internet radio) or two way
  (multi-party audio/video conferencing)
• Security
• Mobility
• Session Management

10
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 acutal arrival
  time
• For a video stream skew could be a slower or
  faster moving picture

11
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

12
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

13
Delay Propagation Delay

• Flight time of packets limited by speed of
  light. Cant 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

14
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

15
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
• It is left to the discretion of the application
  to dynamically adapt to congestion
• 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

16
Bandwidth Related Metrics

• Cost of protocol processing operations relates
  more directly to packet processing rate than to
  the bandwidth in terms of bit rate
• In addition to a bit-rate oriented specification
  of bandwidth, packet processing rate is an
  important metric
• Cost of packet processing is largely dependent on
  number of packets and less so on packet size
• Packetization related metrics include maximum and
  average packet size and packet rate

17
Reliability

• Pertains to loss and corruption of data
• Can be measured in terms of loss probability
• Requires methods for dealing with erroneous/lost
  data

18
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)

19
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

20
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

21
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

22
Error Concealment

• Producing a replacement for a lost packet which
  is similar to the original
• Work for relatively small loss rates (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)

23
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

24
Admission Control

• Pro-active form of congestion control
• Takes requested traffic description as input
  including (in terms of leaky bucket parameters
  (b,r))
• Maximum burst size ( b bucket size)
• Peak rate
• Average rate
• Decides to accept or reject a flow including
  consideration of impact to existing flows

25
Admission Control

• Leaky bucket parameters provide a very loose
  upper bound on the traffic rate
• When traffic becomes bursty network utilization
  can become very low if admission control is
  solely based on parameters provided at call setup
  time
• Admission control unit must also use measurements
  of current network load and packet delay in its
  admission decisions

26
Traffice 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

27
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

28
Packet Scheduling

• FIFO scheduling traditionally used in routers
  needs to be replaced with more sophisticated
  queuing
• E.g. priority queuing. Lower priority queues
  served after higher priority queues
• 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

29
Packet Dropping

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

30
QoS Based Routing

• OSPF, RIP, BGP are best-effort routing protocols
  using single-objective optimization algorithms
  hop count or line cost
• All traffic routed to shortest path can lead to
  congestion on some links and leave other links
  under-utilized
• If link congestion is used to derive line cost
  such that congested routes are costlier, such
  algorithms can cause oscillations in the network,
  thus increasing jitter
• In QoS based routing paths are determined based
  on some knowledge of resource availability in the
  network as well as the QoS requirement of the
  flows

31
Integrated Services

• IntServ Internet Services model developed by IETF
• Requires applications to know their QoS
  requirements beforehand and signal intermediate
  network to reserve resources (bandwidth, buffers)
  
• Requires use of packet classifiers as well as
  packet schedulers
• Almost exclusively concerned with controlling the
  queuing component of end to end delay
• Has three service classes
• Guaranteed Service (RTI)
• Controlled Load (RTT)
• Best-effort

32
IntServ

• Flow descriptor specifies QoS requirements
• Filter spec specifies information for the
  identifying a packet with a given flow
• Flow spec specifies traffic spec in terms of
  token bucket parameters (Tspec) and QoS
  parameters (Rspec) in terms of bandwidth, delay,
  jitter and packet loss
• Resource Reservation Protocol (RSVP) used to
  signal network nodes about required resources

33
RSVP

• The sender sends a PATH message to the receiver
  specifying traffic characteristics
• Every intermediate router along the path forwards
  the path to the next hop determined by the
  routing protocol
• The receiver responds with RESV message

34
Disadvantages of IntServ

• Routers having to maintain per-flow state for
  every flow is a large overhead
• Does not scale in the core network
• Router state has to be refreshed at regular
  intervals increasing traffic overhead
• However, RSVP has a place at the edge of the
  network

35
DiffServ

• Removes some of the shortcomings of the IntServ
  architecture
• Divides network into regions called DS domains.
  Each domain is controlled by a single entity
• To provide service guarantees the entire path
  between source and destination must be in some DS
  domain
• Nodes in a DS domain can be of following types
• Boundary node
• Interior node

36
DiffServ Boundary Node

• Performs admission control to limit the number
  of flows in a domain
• Performs packet classification by marking each
  packet with a service class called Behavior
  Aggregate
• Each Behavior Aggregate is assigned an 8-bit code
  word called a DS code point
• IP ToS field is updated with the code-point

37
DiffServ Interior Node

• Lies completely within a DS domain. Connects with
  other interior nodes or boundary nodes within the
  domain
• Only performs packet forwarding
• Packets are forwarded according to some
  pre-defined rule associated with the packet class
  (as indicated by the code point)
• These pre-defined rules are called Per-Hop
  Behaviors (PHBs)

38
DiffServ PHB

• Two commonly used PHBs are
• Assured Forwarding (AF) Divides incoming traffic
  into four classes where each class guarantees a
  minimum bandwidth and buffer space
• Within each class packets are further assigned
  one of three drop priorities
• Expedited Forwarding (EF) Departure rate of a
  traffic class must equal or exceed the configured
  rate
• Queuing delay is guaranteed to be bounded
• Used to provide Premium Service
• Requires strict admission control and traffic
  policing

39
MPLS

• A router determines the next hop of a packet by
  doing a longest prefix match of an IP destination
  address against entries in a routing table
• This introduces some latency as routing tables
  can be very large
• Same process repeated for every packet even if
  these are in the same flow
• IP switching gets around this
• Short label is attached to every packet and is
  updated at every hop
• This label is used at the next hop as an index
  into the routing table to get the next hop
  (happens in constant time) and next label
• Label is replaced and packet is forwarded to the
  next hop
• Lends itself to being done in (low-cost) hardware
  resulting in very high speeds

40
MPLS

• Like DiffServ MPLS network is divided into
  domains with boundary nodes called Label Edge
  Routers (LER) and interior nodes called Label
  Switching Routers (LSR)
• Packets entering an MPLS domain are assigned a
  label at the ingress LER and are switched inside
  a domain by a simple label lookup
• Labels determine QoS
• Labels may get stripped off at egress LER and
  then get routed conventionally outside the domain
• A sequence of LSR to be followed by a packet in
  an MPLS domain is called a Label Switched Path
  (LSP)
• To guarantee QoS both source and destination have
  to be attached to the same domain or the
  different domains have to have service level
  agreements among them

41
MPLS

• A group of packets that are forwarded in the same
  manner are said to belong to the same Forward
  Equivalence Class (FEC)- FECs form the basis of
  service differentiation
• No limit on number and granularity of FECs
• Labels only have local significance
• Two LSRs agree upon using a particular label for
  a given FEC
• Necessary to do label assignments including label
  allocation and label to FEC binding on every hop
  of the LSP before the traffic flow can use the
  LSP
• Labels can be stacked in FILO order can be used
  in tunneling applications
• In a domain only the topmost label is used to
  make forwarding decisions
• Can be useful in providing mobility
• Home agent can push a label on incoming packets
  and forward them to a foreign agent
• Foreign agent pops of its label and forwards the
  packet to the destination mobile host

42
MPLS Label Assignment

• Label assignment can be done in the following
  ways
• Topology-driven LSPs for every possible FEC are
  automatically set up between every pair of LSR
  (zero call setup delay)
• Request driven LSPs set up based on explicit
  requests.
• RSVP can be used
• Traffic driven LSPs set up only when LSR
  identifies traffic patterns requiring a new LSP
• Traffic patterns not requiring an established LSP
  use the normal routing method
• Combines advantages of above two

43
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)

44
Multicasting Overlay Network (MBONE)

• Virtual network implemented on top of some
  portions of the Internet
• Islands of multicast capable networks are
  connected by virtual links called tunnels
• Tunneled packets are encapsulated as IP over IP
  such that they look like normal unicast packets
  to intervening routers

45
Application Layer Multicasting

• SIP and H.323 support multicasting through a
  multi-point control unit that provides mixing and
  conferencing functionality

46
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

47
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 format

48
Session Management - Session Announcement

• Allows participants to announce future sessions
• E.g. for Internet radio stations to distribute
  information about scheduled shows

49
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

50
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

51
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

52
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

53
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

54
H.323

• Umbrella recommendation that specifies
  components, protocols and procedures multimedia
  conferencing over a packet network
• Defines four components
• Terminals These are the endpoints
• Gateway For interoperation between clients using
  different H.32x flavors
• Gatekeeper Control functions including admission
  control, bandwidth management, call routing
• Multi-point Control Unit For point to multipoint
  conferencing capability
• Uses H.245 to determine common capabilities of
  terminals
• Two kinds of call control models
• Gatekeeper routed (preferred mode in carrier
  environments)
• Direct (not scalable)

55
H.323 Protocol Stack
56
H.323 Stack

• RTP RTCP used for media
• H.225 RAS (Registration, Admission and Status)
  used by endpoints and gateways to
• Gatekeeper discovery and registration
• Requesting call admission, bandwidth allocation
• Clearing a call
• Q.931 signaling protocol is used for call setup
  and teardown between two endpoints lightweight
  version of the ISDN protocol
• H.245 media control protocol is used for
  negotiating media processing capabilities such as
  A/V codec
• T.120 is used for data-conferencing capabilities
  such as whiteboard sharing, instant messaging

57
Session Initiation Protocol - SIP

• Application-layer signaling protocol for
  initiating, modifying and terminating interactive
  sessions. Defined in RFC 3261
• Does not define what a session is. Session is
  carried opaquely in SIP messages.
• Text-encoded protocol based on elements from HTTP
  and SMTP
• SIP supports five facets of establishing and
  terminating multimedia communications
• User location determination of the end system to
  be used for communication
• User availability determination of the
  willingness of the called party to engage in
  communications
• User capabilities determination of the media and
  media parameters to be used
• Session setup "ringing", establishment of
  session parameters at both called and calling
  party
• Session management including transfer and
  termination of sessions, modifying session
  parameters, and invoking services (RFC 3261)

58
SIP Key Capabilities

• A stateful SIP server can split or "fork" an
  incoming call so that several extensions can be
  rung at once
• The first extension to answer can take the call
• SIP can return different media types within a
  single session
• Participants can be invited to existing sessions
• Media can be added (removed from) an existing
  session
• Supports mobility

59
Elements of a SIP Network

• Three main elements in a SIP network
• User Agent end device in a SIP network. User
  Agent Client (UAC) initiates requests. User Agent
  Server (UAS) responds to requests. Roles may
  change in the course of a session.
• Server There are three main types
• Proxy Receives requests from UAs or other proxy
  and forward the request to another location
• Redirect Receives a request from a UA or proxy
  and returns a redirect response (3XX) indicating
  where the request should be retried
• Registrar Receives SIP registration requests and
  updates UAs information to a location server
  (e.g. LDAP server) or other database
• Location Server General term for a database.
  Non-SIP protocol is used to interact with it.

60
SIP Methods

• SIP Methods are commands supported by SIP
• INVITE Invites a user to a call
• ACK Used to facilitate reliable message exchange
  for INVITEs
• BYE Terminates a connection between users or
  declines a call
• CANCEL Terminates a request, or search, for a
  user
• OPTIONS Solicits information about a server's
  capabilities
• REGISTER Registers a user's current location
• INFO Used for mid-session signallingSIP
  responses
• The following are SIP responses
• 1xx Informational (e.g. 100 Trying, 180 Ringing)
• 2xx Successful (e.g. 200 OK, 202 Accepted)
• 3xx Redirection (e.g. 302 Moved Temporarily)
• 4xx Request Failure (e.g. 404 Not Found, 482 Loop
  Detected)
• 5xx Server Failure (e.g. 501 Not Implemented)
• 6xx Global Failure (e.g. 603 Decline)

61
SIP Signaling
62
SIP H.323 Comparison

• SIP is largely equivalent to the Q.931 and H.225
  components of H.323 (sipcenter.com)

63
SIP H.323 Comparison
64
Security

• Integrity
• Authenticity
• Encryption
• Intellectual rights protection
• Digital watermarking techniques embed extra
  information into multimedia data
• Imperceptible to normal user and irremovable

65
Security

• At the IP layer security can be provided by IPSec
• Secure RTP (RFC 3711)
• Provides 128 bit AES encryption
• Confidentiality, authentication and replay
  protection
• SHA-1 (Secure Hash Algorithm) for authentication
• Does not deal with key exchange