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Internet Video Protocols

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Jaeyeon Lee. Networked Media Laboratory. Dept. of Information & Communications ... Packet loss detection, packet re-ordering. Randomly selected initial sequence number ... – PowerPoint PPT presentation

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Title: Internet Video Protocols


1
Internet Video Protocols Applications
2003. 5. 27
2
Contents
  • Introduction
  • Building loss tolerant video delivery system
  • Handling Delay mechanism
  • Congestion Control
  • RTP RTCP
  • Case Study SALSA
  • Conclusion References

3
Introduction
  • Video delivery applications on Internet
  • Video as a part of Information-on-demand(IoD)
  • Applications involved transport of live video
  • Bi-directional interactive video applications
  • Video highest tolerance to packet losses
  • To make the best use of this loss tolerance in
    dealing with the delivery impairments
  • Important impairments of packet delivery over
    internet
  • Loss
  • Delay

4
Loss Tolerant Video Delivery System
  • Application Layer Framing (ALF)
  • Constrain the effect of a packet loss to a single
    packet
  • Increase error resilience
  • Fragmentation smaller than Max. Transmission Unit
    (MTU)
  • Repetition of higher layer headers within each
    packet
  • Reference Picture Selection
  • Proper Key Frame Updates
  • Increasing key frame rate ? reduce waiting time
    for playback
  • Slices or MBs instead of complete frames
  • Layered coding
  • Useful for addressing the heterogeneity issue
  • Not only video information is contained in base
    layer

5
Handling Delay Mechanism
  • Propagation delay on the wires
  • Queuing delay on the routers
  • Play-out buffers size management
  • Determine the initial minimum buffer size
  • Implement a mechanism to change the buffer size
    in real-time with minimal effect on the quality
    of the playback
  • Effective sampling rate modification based on
    frame repeat/drop
  • Global synchronization will be need
  • Synchronous delivery channels with low loss rate
  • Not suitable for multicast of heterogeneous
    receivers

6
Congestion Control
  • Bottleneck bandwidth
  • Upper limit on the speed of delivering data from
    end to end
  • Obstacles of rate control
  • Busty nature of the Internet traffic
  • Multicast applications
  • Rate adjustment algorithm based on a combination
    of the RTCP reports from all receivers
  • Sending different layers of the output of a
    layered codec to different multicast groups
  • Rate control of stored video
  • Require real-time transcoding
  • ? Increase the computational complexity
  • Storing the same video encoded at several rates

7
RTP RTCP
  • Payload type identification
  • Special field at the packet header
  • Packet sequence numbering
  • Packet loss detection, packet re-ordering
  • Randomly selected initial sequence number
  • Time Stamping (32 bit)
  • Encoder/decoder clock matching
  • Synchronization of several sources
  • Measuring packet arrival jitter
  • Source identification
  • Synchronization SouRCe identifier (SSRC)
  • RTCP
  • Periodic transmission of control packets from
    participants of a session

8
RTP RTCP (Cont.)
  • Feedback on the quality of distribution and
    timing
  • RTP reports among sender and receivers
  • Fraction of the lost RTP packets since the last
    report
  • Cumulative of packets lost since the beginning
    of reception
  • Packet inter-arrival jitter
  • Delay since receiving the last senders report
  • Not defined explicit flow control of RTP
  • Capable of generating high traffic rates causing
    network congestion
  • Receiver specific feedback for error recovery
  • Periodic RTCP
  • Contain 64bit Network Time Protocol (NTP)
    timestamps
  • Intra or inter-media synchronization with
    synchronized NTP timestamps

9
RTP RTCP (Cont.)
  • Participant Identification
  • Connection between the real identification of an
    RTP source
  • Using Canonical name (CNAME) current SSRC
  • Scale the control packet transmission
  • Less than 5 of the bandwidth allocated for a
    session

10
SALSA (System for Audio/visual Live Services and
Applications)
  • Two way interactive video over cable TV
  • Many applications of bi-directional video
  • Personal, tutoring, shopping, etc.
  • Layered coding to protect the video from packet
    losses
  • Temporal scalability
  • QoS guarantee between end points
  • Cable Modem Termination System (CMTS) located at
    the cable head-end
  • Data-over-Cable Service Interface Specification
    (DOCSIS) 1.1
  • H.263 G.728 ? transmitting base layer over QoS
    channel
  • Usable video under a 30 packet loss rate on the
    best effort channel

11
Summary Conclusion
  • Outline the Internet packet video delivery
  • Review some possible remedies to address
    feasibility of using the Internet video
  • To design a cost effective video delivery system
    for the Internet is needed
  • QoS guaranteed network infrastructure for
    interactive applications
  • Associated price structure
  • Familiarity with the transport protocol is
    essential for video codec designers

12
References
  • M. R. Civanlar, "Internet video - Protocols
    applications," in Proc. Packet Video Workshop
    2001, 2001

13
Real-Time Internet Video using resilient
Scalable Compression and TCP-Friendly Transport
Protocol
2003. 5. 27
14
Contents
  • Introduction
  • Error-resilient bandwidth-scalable video
    compression
  • TCP-friendly transport protocol
  • Conclusion
  • references

15
Introduction
  • Discrete Wavelet Transform (Subband Coding)
  • DC subband contains most of the energy
  • Generate several layers ? various spatial and
    quality resolutions
  • Multi-priority system

downsample
upsample
?2
?2
H0(z)
G0(z)
X(z)
Y(z)
?
?2
?2
H1(z)
G1(z)
Coding transmission
16
Hierarchical block coding (HBC)
  • Significance Map
  • Binary map indicating whether each coeff. is
    significant at the current bit-plane
  • 44 ? 22 ? indicating each bit

17
Bandwidth Scalable Packetization
  • Each component contains coeff. Blocks which are
    independently compressed using progressive
    quantization by hierarchical block coding
  • Successive bit-planes of a coeff. Block are
    intercoded to produce small codewords
  • ? Fine granularity for layered packetization

18
Encoding procedure
  • Every time the encoder finishes subband analysis,
    it computes and parametized the energy
    distribution of the subbands into 8 parameters
    compactly transmitted per packet

19
Two source of latency in TCP
  • Backlog of data when throughput temporarily drops
    below the data generating rate
  • ? filtering a scalable video bit-stream to meet
    the instantaneous throughput
  • Retransmission delay
  • ? not performing retransmission

20
Throughput estimation of TCP
  • On a lightly loaded network,
  • TCP sending window ? amount of buffer space (B)
  • ? Calculating average throughput (T) with
    buffer space
  • TCP throughput to the packet loss rate (p)
  • (kconstant, MSS Max. segment size)

21
Throughput estimation of TCP (Cont.)
  • Packet losses within one RTT are considered as a
    single congestion event, then
  • C of congestion events in an observation
    window
  • D amount of data in units of MSS (Max. Segment
    Size) transmitted in the same observation window
  • Throughput estimated by (1) and (2) will show
    large variability
  • ? Throughput is calculated using an additional
    RTT estimate that is measured using an accurate
    clock

22
TCP Friendly Transport Protocol
  • Modulated by flow controller not to exceed
    throughput given by (1) and (2)
  • If source attempts to make a fixed of
    transmissions per second
  • If the size of each transmission
  • Reduce transmission overhead by combining
    consecutive small transmission

23
Performance comparison
(proposed)
(T) compared
(T) scalable compression scheme producing
linearly dependent packet
24
FEC Packet Replication
(P) Ave. MSE 100.53
(T) Ave. MSE 201
  • Higher ave. distortion of scheme (T) compared
    with scheme (P) with or without FEC

25
Conclusion
  • Low latency video transmission scheme consisting
    of a TCP-friendly flow controller
  • By elimination of buffering in flow control
  • Overall latency propagation delay decoding
    time
  • Bandwidth-scalable compression scheme producing
    individually decodable packets
  • Produce relatively constant video quality in face
    of packet losses as compared to MPEG and other
    schemes
  • Independently decoded packets reduces the
    resulting distortion even after an adaptive FEC
    scheme is applied to protect the non-resilient
    bitstream

26
References
  • Wai-tian Tan and Avideh Zakhor. "Real-Time
    Internet Video Using Error Resilient Scalable
    Compression and Tcp-Friendly Transport Protocol",
    in IEEE Trans. on Multimedia, 1999
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