Delving into Internet Streaming Media Delivery:

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Delving into Internet Streaming Media Delivery
A quality and Resource Utilization Perspective

• Written by Lei Guo, Enhua Tan, Songqing Chen,
• Zhen Xiao, Oliver Spatscheck, Xiaodong Zhang
• Presented by Harris C.C. Sun
• Dicky Kwok

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Multimedia Downloading
Web browser
Web Server
Long start-up latency Potential waste of traffic
Media player
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Multimedia Pseudo Streaming
Web browser
Web Server
Media player
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Multimedia Streaming
Web browser
Web Server
Media player
Streaming Server
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Streaming Media

• Merits
• Thousands of concurrent streams
• Flexible response to network congestion
• Efficient bandwidth utilization
• High quality to end users
• Challenges
• Lack of QoS on the Internet
• Diverse network connection of users
• Prolonged startup latency
• Research and techniques
• Effective utilization of server and Internet
  resources
• Mechanism of streaming media delivery
• Protocol rollover,
• Fast Streaming,
• MBR and
• rate adaptation

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Motivation

• Modern streaming techniques concentrated on the
  techniques of
• media access pattern and
• user behaviors
• to improve streaming performance.
• Few focused on
• the improvement of techniques for quality
  streaming delivery and
• the effective utilization of media resource.

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Objective and approach

• Understand modern streaming techniques
• The delivery quality and resource utilization
• Collect a large streaming media workload
• From thousands of home users and business users
• Hosted by a large ISP (Gigascope)
• RTSP, RTP/RTCP, MMS, RDT packet headers instead
  of server logs
• Analyze commonly used streaming techniques
• Fast Streaming
• Protocol rollover
• MBR encoding and rate adaptation
• Propose a coordinated streaming mechanism
• Effectively coordinate the merits from caching
  and rate adaptation

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• Trace Collection and processing Methodology
• Traffic Overview
• Fast Streaming
• Rate Adaptation
• Protocol Rollover
• Coordinated Streaming
• Summary

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Trace Collection and processing methodology

• Collect streaming packet
• Capture all TCP packet within ports 554-555,
  7070-7071 9070 and 1755
• Keywords matching to collect RTSP and MMS packet
• Group TCP packet by IP addresses, port, TCP
  SYN/FIN/RST flag
• Extract streaming command from each request
• Identify media data and control packet

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• Trace Collection and processing Methodology
• Traffic Overview
• Fast Streaming
• Rate Adaptation
• Protocol Rollover
• Coordinated Streaming
• Summary

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Traffic Overview

• User communities
• Home users in a cable network
• Business users hosted by a big ISP
• Have different access patterns
• Media hosting services
• Self-hosting
• Third-party hosting

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Traffic Overview

• Access Pattern
• Business users access more audio than home users
• Business users tend to access longer audio/video
  files
• Business users tend to play audio/video longer
• Business users tend to access live audio/video
  longer
• Media Hosting Service
• Self-hosting yahoo.com, aol.com, wbr.com
• Third-party hosting akamai.com. LimeLight
  Networks, fplive.net

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• Trace Collection and processing Methodology
• Traffic Overview
• Fast Streaming
• Rate Adaptation
• Protocol Rollover
• Coordinated Streaming
• Summary

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Fast streaming

• A combination of techniques
• Fast start
• Fast cache
• Fast recovery
• Fast reconnect

Transmits data as fast as possible until the
play-out buffer is filled.
Streams media data up to 5 times the encoding rate
Rare in workload traffic, not include them in
study
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Fast streaming

• A combination of techniques
• Fast start
• Fast cache
• Fast recovery
• Fast reconnect

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Smooth bandwidth flucation

• Rebuffering ratio
• Less rebuffering ratio then normal TCP streaming

Rebuffer ratio rebuffer time / play time
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Production of extra traffic

• Oversupplies media to the client when client
  early terminate the media service
• Over-supplies about 54.8 of media data on
  average, while TCP streaming over-supplies about
  4.8 data only.

Normal TCP lt 5 oversupplied
Fast Cache gt 55 oversupplied
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Server Response time

• Longer time to be served
• 43 of request longer than 0.1 s running on Fast
  Cache while only 9 of request longer than 0.1 s
  in TCP streaming
• Fast cache is statistically longer than that on
  servers not running Fast Cache

Third party media service
Self-hosting media service
gt 40
20 ms
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Server load

• 3.6 times higher server load than normal TCP
  streaming

Server log
1 X 4 X
1 X 4 X
CPU
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Effectiveness ?

• Throughput (rebuffer ratio) of Fast Streaming
  similar to that of TCP streaming
• Only feasible when bandwidth is large enough
• Less possibility of congestion in this case

Encoding rate 200 320 K bps Bandwidth gt 500
Kbps
Fast Cache not resource-efficient
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• Trace Collection and processing Methodology
• Traffic Overview
• Fast Streaming
• Rate Adaptation
• Protocol Rollover
• Coordinated Streaming
• Summary

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Challenges of Streaming

• Bandwidth fluctuation
• Quality of media streaming may significantly
  degrade
• Connection of speed varies
• From dial-up to cable
• Prolonged startup latency

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Challenges of Streaming

• Under the situation of bandwidth fluctuation, a
  technique of Rate Adaptation is widely used by
  media players
• The basic concept is simple.
• modify the stream bit rate to adapt the various
  bandwidth
• Resume when bandwidth recovers
• Never higher than original bit rate

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Rate Adaptation

• In order to adapt to bandwidth fluctuation, major
  media services support three kinds of techniques
  for rate adaptation.
• Stream switch (also known as Intelligent
  Streaming in WM and SureStream in RealNetwork)
• Stream thinning
• Video Cancellation

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Rate Adaptation

• Stream Switch
• Multiple bit rates (MBR) must be used
• According to the statistic of the paper, MBR
  encoding technique is widely used in media
  authoring

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MBR Encoding
on-demand audio
audio stream in video objects
live audio
video stream in video objects
42 on-demand video are MBR encoded with at least
two video stream The maximum numbers of streaming
is 20
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How it works
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How it works

• When bandwidth drops
• WM sends command with a body specifying current
  stream and new stream
• REAL sends UNSUBSCRIBE to cancel the current
  stream, and SUBSCRIBE to switch to new stream

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How it works

• During transmission, if the bandwidth decreases,
  the server automatically detects the change and
  switches to a stream with a lower bit rate. If
  bandwidth improves, the server switches to a 
  stream with a higher bit rate, but never higher
  than the original bit rate. (from Microsoft page)

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Problem Occurs

• After extracting and analyzing information from
  RTSP/MMS commands, switch latency occurs
• Switch latency occurs as the freezing duration
  between old stream and new stream
• User has to wait for re-buffering
• Low quality duration appears

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Problem Occurs

• 3040 of stream switches have latency greater
  than 3 seconds
• 1020 of stream switches have latency greater
  than 5 seconds
• 60 of sessions have low quality duration less
  than 30 seconds
• 85 are shorter than 40 seconds
• Non-trivial for end users

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Stream Thinning

• Similar to stream switch
• If the bandwidth can no longer support the
  streaming video, the image quality will be
  degraded in order to avoid buffering
• Thinning interval is defined as the interval
  between two consecutive stream thinning events

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Stream Thinning

• 70 of the thinning duration are shorter than 30
  seconds
• 70 in the home users and 82 in the business
  users, the thinning intervals are longer than 30
  seconds

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Video Cancellation

• When the bandwidth is too low to transmit the key
  frame of video stream, the client may send a
  TEARDOWN command to cancel the video streaming
• After that, the server maintains the continuous
  audio steam only
• If the bandwidth increases, the client may set up
  and request the video streaming again

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• Trace Collection and processing Methodology
• Traffic Overview
• Fast Streaming
• Rate Adaptation
• Protocol Rollover
• Coordinated Streaming
• Summary

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Protocol Rollover

• Streaming protocol UDP, TCP, HTTP
• Due the wide deployment of NAT routers/firewalls
  in both home and business users, protocol
  rollover results in great affect of startup
  latency

X
X
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Protocol Rollover
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Protocol Rollover

• In the collected data, if protocol rollover
  occurs, it tried to establish UDP from 1 to 3
  times before switching to TCP
• Protocol rollover takes non-trivial time,
  increase the startup latency
• The default protocol in client is usually UDP
• However, some interesting results are revealed by
  the data.

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Protocol Rollover

• In home user workload, only 7.37 of streaming
  session trying UDP first then switching to TCP.
• In business user workload, only 7.95 in the
  streaming session.
• These imply that TCP is directly used without
  protocol rollover in most streaming.
• What happened?

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Protocol Selection Rollover Avoidance

• The phenomenon can be explained as following
• Windows streaming service allows the
  specification of protocol in URL modifier at
  client side or server side
• Content provider use URL modifier to specify
  protocol in the meta file
• rtspt//xxx.xxx.com/xxx.wmv (TCP) gt70
• rtspu//xxx.xxx.com/xxx.wmv (UDP) rarely
• After extracting URL modifier, 70 of streaming
  session is specified as TCP

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Protocol Selection Rollover Avoidance

• The conjecture is
• Content providers are aware of NAT/firewalls
• Understand UDP is mostly shielded by clients
• They actively use TCP to avoid shielding or
  protocol rollover
• Even if UDP is supported, the streaming is
  delivered over TCP directly

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Proof

• Further investigate the NAT usage with MMS
• Different from RTSP, clients report local IP
  address to server
• Most users report private IPs
• Indicate that clients access internet through NAT

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• Trace Collection and processing Methodology
• Traffic Overview
• Fast Streaming
• Rate Adaptation
• Protocol Rollover
• Coordinated Streaming
• Summary

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Coordinating caching and rate adaptation

• Fast Cache aggressively buffer data in advance
• Over-utilize CPU and bandwidth resources
• Neither performance effective nor cost-efficient
• Rate adaptation conservatively switch to lower
  bit rate stream
• Switch handoff latency
• Coordinated Streaming

high rate stream
low rate stream
Upper bound Prevent aggressive buffering
Lower bound Prevent switch latency
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Results

• Rebuffering ratio is close to zero
• Reduces 77 over-supplied traffic produced by
  Fast Cache, though still not as good as TCP
  streaming
• Switch handoff latency is nearly zero

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• Trace Collection and processing Methodology
• Traffic Overview
• Fast Streaming
• Rate Adaptation
• Protocol Rollover
• Coordinated Streaming
• Summary

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What We Learn and What We Think

• This paper mainly aim at investigating the
  current streaming technique used by media players
  
• Most people enjoy video streaming but seldom know
  the theory behind
• Modern streaming services over-utilize CPU and
  bandwidth resource
• Coordinated Streaming is suggested
• Ideal theory but not sure if it works in reality.

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What We Learn and What We Think

• More research papers are written by same group of
  people
• More study are needed

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Thank You!