Computer Network
Architectures and Multimedia
Guy Leduc
Chapter 4
Multimedia Applications
& Transport
Sections 7.1 to 7.4 from
Computer Networking: A Top
Down Approach,
6th edition.
Jim Kurose, Keith Ross
Addison-Wesley, March 2012.
Also 7.4.2 and 7.4.7 from
Computer Networks - 4th edition
Andrew S. Tanenbaum
Prentice-Hall International, 2003
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp.
4-1
Multimedia networking: outline
4.1 multimedia networking applications
4.2 streaming stored video
4.3 voice-over-IP
4.4 protocols for real-time conversational
applications
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp.
4-2
1
Multimedia: audio


PCM (Pulse Code Modulation):
analog audio signal sampled at
constant rate
 telephone: 8,000
samples/sec
 CD music: 44,100
samples/sec
each sample quantized, i.e.,
rounded
 each quantized value
represented by bits,
 e.g., rounded to one of
28=256 values
 8 bits/sample
receiver converts bits back
to analog signal:
 some quality reduction
quantization
error
audio signal amplitude

quantized
value of
analog value
analog
signal
time
sampling rate
(N sample/sec)
4: Multimedia App. & Transp.
©From Computer Networking, by Kurose&Ross
4-3
Multimedia: audio
Examples:

Telephony:
 8,000 samples/sec,
8 bits/sample:
64 kbps
CD music:
 44,100 samples/sec,
16 bits/sample:
705.6 kbps
 Stereo: 1.411 Mbps
Other example rates


MP3: 96, 128, 160 kbps
Internet telephony: 5.3
kbps and up
©From Computer Networking, by Kurose&Ross
quantization
error
audio signal amplitude

quantized
value of
analog value
analog
signal
time
sampling rate
(N sample/sec)
4: Multimedia App. & Transp.
4-4
2
More on Audio Compression
The threshold of audibility
as a function of frequency
The frequency masking effect
MP3 (MPEG 1 audio layer 3) takes masking effects into account and
does not encode masked signals.
Can compress stereo CD down to 96-128 kbps.
4: Multimedia App. & Transp.
From Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
Multimedia: video
❒ video: sequence of images
displayed at constant rate
❍ e.g. 25 images/sec
❒ digital image: array of pixels
❍ each pixel represented
by bits
❒ coding: use redundancy
within and between images
to decrease # bits used to
encode image
❍ spatial (within image)
❍ temporal (from one image
to next)
spatial coding example: instead
of sending N values of same
color (all purple), send only two
values: color value (purple) and
number of repeated values (N)
……………………...…
……………………...…
frame i
temporal coding example:
instead of sending
complete frame at i+1,
send only differences from
frame i
frame i+1
©From Computer Networking, by Kurose&Ross
4-5
4: Multimedia App. & Transp.
4-6
3
Multimedia: video



CBR: (constant bit
rate): video encoding
rate fixed
VBR: (variable bit
rate): video encoding
rate changes as amount
of spatial, temporal
coding changes
examples:
 MPEG 1 (CD-ROM) 1.5
Mbps
 MPEG2 (DVD) 3-6 Mbps
 MPEG4 (often used in
Internet, < 1 Mbps)
spatial coding example: instead
of sending N values of same
color (all purple), send only two
values: color value (purple) and
number of repeated values (N)
……………………...…
……………………...…
frame i
temporal coding example:
instead of sending
complete frame at i+1,
send only differences from
frame i
frame i+1
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp.
4-7
Video - Digital Systems
❒ Consider a rectangular 4:3 grid of pixels, such as
❍ VGA: 640 x 480
❍ XGA: 1024 x 768
❒ Pixel = 8 bits for each of the RGB colours
❒ 25 frames per sec
❒ With XGA :
❍ 24 bits/pixel x 1024 x 768 x 25 frames/sec = 472 Mbps!
❒ Needs compression!
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp.
4-8
4
Data compression
❒ Encoding/decoding schemes
❒ Video on Demand (VoD)
❍
❍
❍
Encoding can be slow (done once)
Decoding must be fast (done many times)
Asymmetrical schemes
❒ Real-time multimedia (e.g. videoconference)
❍
Symmetrical schemes
❒ Lossy compression
❍
❍
Encode/decode is not neutral
When acceptable, leads to better compression ratios
❒ Two main compression schemes:
❍
❍
Entropy encoding (lossless)
Source encoding (lossy)
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp.
4-9
Entropy encoding
❒ Lossless
❒ Three typical examples:
❍ Run-length encoding
• repeated symbols are encoded as “Special symbol + number
of occurrences”
❍
Statistical encoding
• short codes for frequent symbols
❍
Look up table
• e.g. CLUT (Colour Look Up Table)
• define the table of the colours actually used
• send table index instead of a 24-bit colour value
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-10
5
Source encoding
❒ Lossy
❒ Three main examples:
❍ Differential encoding
• sequence of values are encoded by representing the differences
from the previous values
• makes sense if differences are encoded with less bits
• lossy when there are large jumps between two values and a fixed
number of bits per difference
• lossless if variable-length encoding is used
❍
Transformation
• e.g. Fourier or DCT Transform
• lossy since only the first amplitudes are sent
❍
Variant of Look Up Table with approximations to closest value
4: Multimedia App. & Transp. 4-11
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
JPEG
❒ Joint Photographic Experts Group
❒ ISO/IEC and ITU standard for compressing still pictures
❒ Compression ratio 20:1 is typical
❒ Roughly symmetrical scheme (decoding as long as encoding)
❒ Lossy sequential mode:
❍
6 steps
Block
preparation
Discrete
Cosine
transform
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
Quantization
Differential
quantization
Run-length
encoding
Statistical
Output
encoding
4: Multimedia App. & Transp. 4-12
6
JPEG - Step 1
Block
preparation
Discrete
Cosine
transform
Quantization
Differential
quantization
Run-length
encoding
Statistical
Output
encoding
❒ Step 1: block preparation
❍
Translate RGB into luminance (Y) and 2 chrominance (I,Q) values
• gives better compression
• we get 3 matrices of pixels
❍
Average square blocks of 4 pixels for I and Q
• lossy but unnoticeable
❍
❍
Subtract 128 from each element (0 is middle)
Divide up frame into 8x8 blocks
4: Multimedia App. & Transp. 4-13
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
JPEG - Step 2
Block
preparation
Discrete
Cosine
transform
Quantization
Differential
quantization
Run-length
encoding
Statistical
Output
encoding
❒ Step 2: DCT (Discrete Cosine Transformation) to each block
❍
Sort of 2 dimensional Discrete Fourier Transform
• Advantage: most of the spectral power in the first few terms
❍
❍
Output: block of 8x8 elements (coefficient of DCT)
Slightly lossy in practice (round-off errors)
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-14
7
JPEG - Step 3
Block
preparation
Discrete
Cosine
transform
Quantization
Differential
quantization
Run-length
encoding
Statistical
Output
encoding
❒ Step 3: Quantization
❍ Apply sort of low pass filter to coefficients (lossy)
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-15
JPEG - Steps 4, 5 and 6
❒ Step 4: Differential quantization
❍
Replace upper-left (DC) coefficient by its difference with
corresponding element of previous block
❒ Step 5: Run-length encoding
❍
Applied to a zig-zag scanning pattern
❒ Step 6: Statistical output encoding (Huffman)
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-16
8
MPEG
❒ Motion Picture Experts Group - ISO standard
❒ Audio and video
❒ MPEG-1
❍ Video-recorder quality (CD-ROM)
❍ 1.2 Mbps output
❒ MPEG-2
❍ Broadcast quality
❍ 4-6 Mbps output is typical but higher for HDTV
❒ MPEG-4
❍ Medium-resolution videoconferencing with low frame rate
• 10 frames/sec
4: Multimedia App. & Transp. 4-17
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
MPEG-1
Audio signal
Audio
encoder
System
multiplexer
Clock
Video signal
MPEG-1 output
Video
encoder
❒ Audio and video encoders work independently
❒ Timestamps included in both flows for
synchronization at receiver
❒ Audio compression (MP3)
❍
Also, exploitation of redundancy in the 2 channels of a
stereo stream
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-18
9
MPEG-1 - Video compression
❒ Exploit spatial and temporal redundancies
❒ Spatial redundancy: like JPEG
❒ But adds temporal redundancy
❍
Many common parts in the following three consecutive frames!
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-19
MPEG-1 - Video compression (2)
❒ Temporal redundancy: four kinds of frames: I, P, B, D
❍ I frames (Intracoded)
• self-contained JPEG-encoded still pictures
• should appear periodically in the output (initial synch, resynch on
error, fast forward or rewind)
❍
P frames (Predictive)
• block-by-block difference with the previous frame
• search for a macroblock (Y,I,Q) in previous frame which is equal or
slightly different
• encode the offset in position and difference
❍
❍
B frames (Bidirectional): same as P but search also in next I or P
frame
D frames (DC-coded): block averages for fast forward (low
resolution)
❒ Example of part of an MPEG sequence
❍ IBBPBBI
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-20
10
MPEG-2
❒ Similar to MPEG-1
❒ Better quality (10 x 10 DCT coefficients instead
of 8 x 8)
❒ Several resolution levels (lowest one is comparable
to MPEG-1)
❒ Several profiles (e.g. no B frames to simplify
encoding)
❒ Usually 3-4 Mbps, but can go up to 100 Mbps
(HDTV)
From
Computer
Networks,
by Tanenbaum
© Prentice Hall
©From
Computer
Networking,
by Kurose&Ross
4: Multimedia App. & Transp. 4-21
Multimedia networking:
3 application types
❒
streaming, stored audio, video
❍
❍
❍
❒
conversational voice/video over IP
❍
❍
❒
streaming: can begin playout before downloading entire
file
stored (at server): can transmit faster than audio/video
will be rendered (implies storing/buffering at client)
e.g., YouTube, Netflix, Hulu
interactive nature of human-to-human conversation limits
delay tolerance
e.g., Skype
streaming live audio, video
❍
e.g., live sporting event
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-22
11
MM Networking Applications
Fundamental characteristics:
❒ typically delay sensitive
❍
❍
end-to-end delay
delay jitter
Jitter is the variability
of packet delays within
the same packet stream
❒ loss tolerant: infrequent losses cause
minor glitches
❒ antithesis of data, which are loss
intolerant but delay tolerant
QoS (Quality of Service) refers to performance
metrics such as delay, bandwidth, jitter and loss
4: Multimedia App. & Transp. 4-23
©From Computer Networking, by Kurose&Ross
Multimedia Over Today’s Internet
TCP/UDP/IP: “best-effort service”
❒
no guarantees on delay, bandwidth, jitter, loss (if UDP)
?
?
?
?
?
?
?
But you said multimedia apps require ?
QoS and level of performance to be
? effective!
?
?
Today’s Internet multimedia applications
use application-level techniques to mitigate
(as best possible) effects of delay, loss
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-24
12
Chapter 4: outline
4.1 multimedia networking applications
4.2 streaming stored video
4.3 voice-over-IP
4.4 protocols for real-time conversational
applications
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-25
Internet multimedia: simplest approach
Media player
❒ jitter removal
❒ decompression
❒ error concealment
❒ graphical user interface
with controls for
interactivity
❒ audio or video stored in file
❒ files transferred as HTTP object
received in entirety at client
then passed to player
audio, video not streamed in this scenario:
❒ no, “pipelining,” long delays until playout!
❍
❍
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-26
13
Internet multimedia: streaming approach
❒ browser GETs metafile
❒ browser launches player, passing metafile
❒ player contacts server
❒ server streams audio/video to player
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-27
Streaming from a streaming server
❒ allows for non-HTTP protocol between server and media player
❒ UDP or TCP for step (3), more shortly
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-28
14
Streaming Multimedia: client rate(s)
1.5 Mbps encoding
28.8 Kbps encoding
Q: how to handle different client receive rate
capabilities?
A: server stores, transmits multiple copies
of video, encoded at different rates
4: Multimedia App. & Transp. 4-29
©From Computer Networking, by Kurose&Ross
Cumulative data
Streaming stored video:
1. video
Recorded
(e.g., 30
frames/sec
)
©From Computer Networking, by Kurose&Ross
2. video
sent
network delay
(fixed in this
example)
3. video received,
played out at client
(30 frames/sec) time
streaming: at this time, client
playing out early part of video,
while server still sending later
part of video
4: Multimedia App. & Transp. 4-30
15
Streaming stored video: challenges
continuous playout constraint: once client
playout begins, playback must match original
timing
 … but network delays are variable (jitter),
so will need client-side buffer to match
playout requirements
 other challenges:
 client interactivity: pause, fast-forward,
rewind, jump through video
 video packets may be lost, retransmitted

4: Multimedia App. & Transp. 4-31
©From Computer Networking, by Kurose&Ross
Streaming stored video: revisited
client video
reception
variable
network
delay
client playout
delay
❒
constant bit
rate video
playout at client
buffered
video
Cumulative data
constant bit
rate video
transmission
time
client-side buffering and playout delay:
compensate for network-added delay, delay jitter
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-32
16
Client-side buffering, playout
buffer fill level,
Q(t)
playout rate,
e.g., CBR r
variable fill
rate, x(t)
client application
buffer, size B
video server
client
4: Multimedia App. & Transp. 4-33
©From Computer Networking, by Kurose&Ross
Client-side buffering, playout
buffer fill level,
Q(t)
playout rate,
e.g., CBR r
variable fill
rate, x(t)
video server
client application
buffer, size B
client
1. initial fill of buffer until playout begins at tp
2. playout begins at tp,
3. buffer fill level varies over time as fill rate
x(t) varies and playout rate r is constant
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-34
17
Client-side buffering, playout
buffer fill level,
Q(t)
playout rate,
e.g., CBR r
variable fill
rate, x(t)
client application
buffer, size B
video server
x < r: buffer may empty, causing freezing of video
playout until buffer again fills
initial playout delay tradeoff: buffer starvation less
likely with larger delay, but larger delay until user
begins watching
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-35
Streaming multimedia: UDP
❒ server sends at rate appropriate for client
often: send rate = encoding rate = constant rate
transmission rate can be oblivious to congestion levels
❒ short playout delay (2-5 seconds) to remove network jitter
❒ error recovery: application-level, time permitting
❒ encapsulation of audio/video chunks in RTP (Real-Time
Transport Protocol, RFC 3550) and then in UDP
❍
❍
❍
see later for details
❒ needs a control connection in parallel to pause, resume
reposition, etc:
❍
Real-Time Streaming Protocol (RTSP, RFC 2326)
❒ issue: UDP may
not go through firewalls
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-36
18
User Control of Streaming Media: RTSP
HTTP
❒ does not target multimedia
content
❒ no commands for fast
forward, etc.
RTSP
❒ Real-Time Streaming
Protocol
❒ client-server application
layer protocol
❒ user control: rewind, fast
forward, pause, resume,
repositioning, etc.
What it doesn’t do:
❒ doesn’t define how
audio/video is
encapsulated for
streaming over network
❒ doesn’t restrict how
streamed media is
transported (UDP or
TCP possible)
❒ doesn’t specify how
media player buffers
audio/video
4: Multimedia App. & Transp. 4-37
©From Computer Networking, by Kurose&Ross
RTSP: out-of-band control
FTP uses an “out-ofband” control channel:
❒ file transferred over
one TCP connection
❒ control info (directory
changes, file deletion,
rename) sent over
separate TCP
connection
❒ “out-of-band”, “inband” channels use
different port
numbers
©From Computer Networking, by Kurose&Ross
RTSP messages also sent
out-of-band:
❒ RTSP control
messages use
different port
numbers than media
stream: out-of-band
❍ port 554
❒ media stream is
considered “in-band”
4: Multimedia App. & Transp. 4-38
19
RTSP example
Scenario:
❒ metafile communicated to web browser (1)
❒ browser launches player (2)
❒ player sets up an RTSP control connection, data connection to
streaming server (3)
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-39
Metafile Example
<title>Twister</title>
<session>
<group language=en lipsync>
<switch>
<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">
</switch>
<track type="video/jpeg"
src="rtsp://video.example.com/twister/video">
</group>
</session>
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-40
20
RTSP Operation
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-41
RTSP exchange example
C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0
Transport: rtp/udp; compression; port=3056; mode=PLAY
S: RTSP/1.0 200 1 OK
Session 4231
C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0
Session: 4231
Range: npt=0C: 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 3 OK
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-42
21
Streaming multimedia: TCP
❒ multimedia file retrieved via HTTP GET
❒ send at maximum possible rate under TCP
variable
rate, x(t)
video
file
TCP send
buffer
server
TCP receive
buffer
application
playout buffer
client
❒ fill rate fluctuates due to TCP congestion control,
retransmissions (in-order delivery)
❒ larger playout delay: smooth TCP delivery rate
❒ HTTP/TCP passes more easily through firewalls
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-43
Streaming multimedia: DASH
DASH: Dynamic, Adaptive Streaming over HTTP
❒ server:
❒
❍
❍
❍
❒
divides video file into multiple chunks
each chunk stored, encoded at different rates
manifest file: provides URLs for different chunks
client:
❍
❍
periodically measures server-to-client bandwidth
consulting manifest, requests one chunk at a time
• chooses maximum coding rate sustainable given
current bandwidth
• can choose different coding rates at different points
in time (depending on available bandwidth at time)
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-44
22
Streaming multimedia: DASH
❒ DASH: Dynamic, A daptive Streaming over
HTTP
❒ “intelligence” at client: client determines
when to request chunk (so that buffer
starvation, or overflow does not occur)
❍ what encoding rate to request (higher quality
when more bandwidth available)
❍ where to request chunk (can request from URL
server that is “close” to client or has high
available bandwidth)
❍
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-45
Content distribution networks
❒
❒
challenge: how to stream content (selected from
millions of videos) to hundreds of thousands of
simultaneous users?
option 1: single, large “mega-server”
❍
❍
❍
❍
single point of failure
point of network congestion
long path to distant clients
multiple copies of video sent over outgoing link
… quite simply: this solution doesn’t scale
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-46
23
Content distribution networks
❒
❒
challenge: how to stream content (selected from
millions of videos) to hundreds of thousands of
simultaneous users?
option 2: store/serve multiple copies of videos at
multiple geographically distributed sites (CDN)
❍
enter deep: push CDN servers deep into many access networks
• close to users
• used by Akamai, 1700 locations
❍
bring home: smaller number (10’s) of larger clusters in POPs
near (but not within) access networks
❍
Google uses both, in addition to its “mega data centers”
responsible for serving dynamic content
• used by Limelight
4: Multimedia App. & Transp. 4-47
©From Computer Networking, by Kurose&Ross
CDN: “simple” content access scenario
Bob (client) requests video http://video.netcinema.com/6Y7B23V
actually stored in a KingCDN content distribution server
1. Bob gets URL for video
http://video.netcinema.com/6Y7B23V
from netcinema.com
2
web page
1
6. request video from 5
KingCDN server,
streamed via HTTP
3.
netcinema’s
DNS returns
netcinema.com
a1105.kingcdn.com
3
netcinema
authoritative DNS
KingCDN content
distribution server
©From Computer Networking, by Kurose&Ross
2. resolve video.netcinema.com
via Bob’s local DNS that relays to
netcinema’s authoritative DNS server
4
4&5. Resolve
a1105.kingcdn.com
via KingCDN’s authoritative DNS,
which returns IP address of KingCDN
distribution server with video
KingCDN
authoritative DNS
4: Multimedia App. & Transp. 4-48
24
CDN cluster selection strategy
❒
challenge: how does CDN DNS select “good” CDN node to stream to
client
❍
❍
CDN learns the IP address of the client’s local DNS via the client’s DNS
lookup
CDN can then implement a selection strategy to dynamically direct clients
to a “suitable” server cluster or data center
❒ Possible strategies:
❍
❍
❍
❒
pick CDN node geographically closest to client
pick CDN node with shortest delay (or min # hops) to client (CDN nodes
periodically ping access ISPs, reporting results to CDN DNS)
IP anycast: the CDN assigns the same IP address to each of its clusters,
and uses standard BGP to advertise this IP address from each of the
different cluster locations. When a BGP router receives multiple route
advertisements for this same IP address, it treats them as providing
several paths to the same physical location and picks the “best”
alternative: let client decide - give client a list of several CDN servers
❍
❍
client pings servers, picks “best”
Netflix approach
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-49
Case study: Netflix
❒ 30% downstream US traffic in 2011
❒ owns very little infrastructure, uses 3rd party
services:
❍ own registration, payment servers
❍ Amazon (3rd party) cloud services:
• Netflix uploads studio master to Amazon cloud
• create multiple version of movie (different encodings) in
cloud
• upload versions from cloud to CDNs
• Cloud hosts Netflix web pages for user browsing
❍
three 3rd party CDNs host/stream Netflix
content: Akamai, Limelight, Level-3
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-50
25
Case study: Netflix
Amazon cloud
Netflix registration,
accounting servers
2. Bob browses
Netflix video 2
upload copies of
multiple versions of
video to CDNs
3. Manifest file
returned for
requested video
Akamai CDN
Limelight CDN
3
1
1. Bob manages
Netflix account
Level-3 CDN
4. DASH
streaming
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-51
Chapter 4: outline
4.1 multimedia networking applications
4.2 streaming stored video
4.3 voice-over-IP
4.4 protocols for real-time conversational
applications
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-52
26
Voice-over-IP (VoIP)
❒
VoIP end-end-delay requirement: needed to maintain
“conversational” aspect
❍
❍
❍
❍
higher delays noticeable, impair interactivity
< 150 msec: good
> 400 msec: bad
includes application-level (packetization,playout), network
delays
session initialization: how does callee advertise IP
address, port number, encoding algorithms?
❒ value-added services: call forwarding, screening,
recording
❒ emergency services: 112 (Europe), 911 (North America)
❒
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-53
VoIP characteristics
❒ speaker’s audio: alternating talk spurts, silent
periods.
❍
64 kbps during talk spurt
❍
pkts generated only during talk spurts
❍
20 msec chunks at 8 Kbytes/sec: 160 bytes of data
❍
so, 20 msec of packetization delay
❒ application-layer header added to each chunk
❒ chunk+header encapsulated into UDP (or TCP)
segment
❒ application sends segment into socket every 20 msec
during talkspurt
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-54
27
VoIP: packet loss, delay
❒ network loss: IP datagram lost due to
network congestion (router buffer overflow)
❒ delay loss: IP datagram arrives too late for
playout at receiver
delays: processing, queueing in network;
end-system (sender, receiver) delays
❍ typical maximum tolerable delay: 400 ms
❍
❒ loss tolerance: depending on voice encoding,
loss concealment, packet loss rates between
1% and 10% can be tolerated
4: Multimedia App. & Transp. 4-55
©From Computer Networking, by Kurose&Ross
constant bit
rate
transmission
variable
network
delay
(jitter)
client
reception
constant bit
rate playout
at client
buffered
data
Cumulative data
Delay jitter
time
client playout
delay
❒ end-to-end delays of two consecutive packets:
difference can be more or less than 20 msec
(transmission time difference)
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-56
28
VoIP: fixed playout delay
❒ receiver attempts to playout each chunk exactly
q msecs after chunk was generated
❍ chunk has timestamp t: play out chunk at t+q
❍ chunk arrives after t+q: data arrives too late
for playout, data “lost”
❒ tradeoff in choosing q:
❍ large q: less packet loss
❍ small q: better interactive experience
4: Multimedia App. & Transp. 4-57
©From Computer Networking, by Kurose&Ross
VoIP: fixed playout delay
• sender generates packets every 20 msec during talk spurt
• first packet received at time r
• first playout schedule: begins at p
• second playout schedule: begins at p’
packets
loss
packets
generated
packets
received
playout schedule
p-r
playout schedule
p’ - r
time
r
©From Computer Networking, by Kurose&Ross
p
p'
4: Multimedia App. & Transp. 4-58
29
Adaptive playout delay (1)
goal: low playout delay, low late loss rate
❒ approach: adaptive playout delay adjustment:
❒
❍
❍
❍
estimate network delay, adjust playout delay at beginning
of each talk spurt
silent periods compressed and elongated
chunks still played out every 20 msec during talk spurt
❒ adaptively estimate packet delay:
(EWMA - exponentially weighted moving average, recall TCP
RTT estimate):
di = (1−α)di-1 + α (ri – ti)
delay estimate
after ith packet
small constant,
e.g. 0.1
time received - time sent
(timestamp)
measured delay of ith packet
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-59
Adaptive playout delay (2)
also useful to estimate average deviation of delay, vi :
vi = (1−β)vi-1 + β |ri – ti – di|
❒
❒
estimates di, vi calculated for every received
but used only at start of talk spurt
packet,
for first packet in talk spurt, playout time is:
playout-timei = ti + di + Kvi
❒
remaining packets in talkspurt are played out
periodically
Q: does it require clock synchronization?
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-60
30
Adaptive playout delay (3)
Q: How does receiver determine whether packet is
first in a talk spurt?
❒ if no loss, receiver looks at successive timestamps
❍ difference of successive stamps > 20 msec -> talk spurt
begins
❒ with loss possible, receiver must look at both time
stamps and sequence numbers
❍
difference of successive stamps > 20 msec and sequence
numbers without gaps -> talk spurt begins
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-61
VoIP: recovery from packet loss (1)
Challenge: recover from packet loss given small tolerable
delay between original transmission and playout
❒ each ACK/NAK takes ~ one RTT
Forward Error Correction (FEC)
send enough bits to allow recovery without retransmission
(recall two-dimensional parity)
❒ alternative:
❍
simple FEC
n chunks, create redundant chunk by exclusive
OR-ing n original chunks
❒ send n+1 chunks, increasing throughput by factor 1/n
❒ can reconstruct original n chunks if at most one lost chunk from
n+1 chunks, with playout delay
❒ called “erasure” code
❒ for every group of
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-62
31
VoIP: recovery from packet loss (2)
❒ increasing throughput:
❍ by factor 1/n
❒ increasing playout delay:
❍ need enough time to receive all n+1 packets
❒ tradeoff:
increase n, less bandwidth waste
❍ increase n, longer playout delay
❍ increase n, higher probability that 2 or more chunks
will be lost
❍
4: Multimedia App. & Transp. 4-63
©From Computer Networking, by Kurose&Ross
VoIP: recovery from packet loss (3)
FEC: Reed-Solomon (RS) scheme
❒ RS is a more sophisticated
error correcting code, which
can be used as erasure code
❒ An (n,k) RS code encodes k
source packets into n > k
packets
❒ Systematic code: the n
transmitted packets contain
verbatim copies of the k source
packets
❍
❍
❒
+ n-k new packets
no decoding if no source packet
loss!
Optimal code: Original k
packets can be recovered
provided that any k packets
among n are received
©From Computer Networking, by Kurose&Ross
❒ Linear code: coding/decoding
represented by matrix
operations:
❍
❍
❍
❍
x is the vector of k source
packets
G is a n x k matrix
y is the vector of n transmitted
packets
y=Gx
❒ Decoding:
❍
❍
❍
y’ vector of any k received
packets
G’ is the k x k submatrix of G
with rows corresponding to
these packets
x = G’-1 y’
4: Multimedia App. & Transp. 4-64
32
VoIP: recovery from packet loss (4)
2nd FEC scheme
 “piggyback lower
quality stream”
 send lower resolution
audio stream as
redundant information
 e.g., nominal
stream PCM at 64 kbps
and redundant stream
GSM at 13 kbps.
non-consecutive loss, receiver can conceal the loss
generalization: can also append (n-1)st and (n-2)nd low-bit rate
chunks


©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-65
VoIP: recovery from packet loss (5)
Interleaving to conceal loss
❒ audio chunks divided into smaller
units
❒ for example, four 5 msec units
per 20 ms audio chunk
❒ packet contains small units from
different chunks
©From Computer Networking, by Kurose&Ross
most
of every chunk
❒ no redundancy overhead, but
increases playout delay
❒ if packet lost, still have
4: Multimedia App. & Transp. 4-66
33
Voice-over-IP: Skype
Skype clients (SC)
❒ proprietary application-
layer protocol (inferred
via reverse engineering)
❍ encrypted msgs
❒ P2P components:
 clients: skype peers
connect directly to
each other for VoIP
call
 super nodes (SN):
skype peers with
special functions
 overlay network: among
SNs to locate SCs
Skype
login server
supernode (SN)
supernode
overlay
network
 login server
4: Multimedia App. & Transp. 4-67
©From Computer Networking, by Kurose&Ross
P2P voice-over-IP: skype
skype client operation:
1. joins skype network by
contacting SN (IP address
cached) using TCP
2. logs-in (username,
password) to centralized
skype login server
Skype
login server
3. obtains IP address for
callee from SN, SN
overlay
 or client buddy list
4. initiate call directly to
callee
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-68
34
Skype: peers as relays
Problem: both Alice, Bob
are behind “NATs”
❍
❍
NAT prevents outside peer
from initiating connection
to insider peer
inside peer can initiate
connection to outside
relay solution: Alice, Bob
maintain open connection
to their SNs
 Alice signals her SN to
connect to Bob
 Alice’s SN connects to Bob’s
SN
 Bob’s SN connects to Bob
over open connection Bob
initially initiated to his SN
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-69
Chapter 4: outline
4.1 multimedia networking applications
4.2 streaming stored video
4.3 voice-over-IP
4.4 protocols for real-time conversational
applications: RTP/RTCP, SIP
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-70
35
Real-Time Protocol (RTP)
❒ RTP specifies packet
structure for packets
carrying audio, video
data
❒ RFC 3550
❒ RTP packet provides
❍ payload type
identification
❍ packet sequence
numbering
❍ time stamping
❒ RTP runs in end systems
❒ RTP packets
encapsulated in UDP
segments
❒ interoperability: if two
Internet phone
applications run RTP,
then they may be able
to work together
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-71
RTP runs on top of UDP
RTP libraries provide transport-layer interface
that extends UDP:
• port numbers, IP addresses
• payload type identification
• packet sequence numbering
• time-stamping
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-72
36
RTP Example
❒ consider sending 64
kbps PCM-encoded
voice over RTP
❒ application collects
encoded data in
chunks, e.g., every 20
msec = 160 bytes in a
chunk
❒ audio chunk + RTP
header form RTP
packet, which is
encapsulated in UDP
segment
©From Computer Networking, by Kurose&Ross
❒ RTP header indicates
type of audio encoding
in each packet
❍
sender can change
encoding during
conference
❒ RTP header also
contains sequence
numbers, timestamps
4: Multimedia App. & Transp. 4-73
RTP and QoS
❒ RTP does not provide any mechanism to ensure
timely data delivery or other QoS guarantees
❒ RTP encapsulation is only seen at end systems
(not by intermediate routers)
❍ routers provide best-effort service, making no
special effort to ensure that RTP packets
arrive at destination in timely manner
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-74
37
RTP entities
Different encodings
Multiple streams
End System
SSRC = 53
Translator
Mixer
SSRC = 19
Single stream
SSRC = 19
CSRC = 53 77
End System
SSRC = 77
❒
End system: application that actually generates/consumes the content
carried in RTP packets
❍
SSRC: Synchronisation Source identifier
Translator: intermediate system that changes the encoding scheme without
altering the timing. It may also convert multicast into multiple unicast
streams
❒ Mixer: intermediate system that receives multiple streams and combines
them in some manner. The new stream has its own timing (new SSRC)
❒
❍
CSRC: Contributing Source identifier
4: Multimedia App. & Transp. 4-75
©From Computer Networking, by Kurose&Ross
RTP Header
payload
type
sequence
number type
time stamp
Synchronization
Source ID
Miscellaneous
fields
Payload Type (7 bits): Indicates type of encoding currently being
used. If sender changes encoding in middle of conference, sender
informs receiver via payload type field
•Payload type 0: PCM µ-law, 64 kbps
•Payload type 3: GSM, 13 kbps
•Payload type 7: LPC, 2.4 kbps
•Payload type 26: Motion JPEG
•Payload type 31: H.261
•Payload type 33: MPEG2 video
Sequence Number (16 bits): incremented by one for each RTP
packet sent, detects packet loss and restores packet sequence
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-76
38
RTP Header (2)
payload
type
❒
Synchronization
Source ID
time stamp
Miscellaneous
fields
Timestamp field (32 bytes): sampling instant of
first byte in this RTP data packet
❍
❍
❒
sequence
number type
for audio, timestamp clock typically increments by one for
each sampling period (for example, each 125 µsecs for 8
KHz sampling clock)
if application generates chunks of 160 encoded samples,
then timestamp increases by 160 for each RTP packet when
source is active. Timestamp clock continues to increase at
constant rate when source is inactive
SSRC field (32 bits): identifies source of RTP stream.
Each stream in RTP session should have distinct SSRC.
4: Multimedia App. & Transp. 4-77
©From Computer Networking, by Kurose&Ross
Real-Time Control Protocol (RTCP)
❒ works in conjunction
with RTP
❒ each participant in
RTP session
periodically
transmits RTCP
control packets to
all other
participants
©From Computer Networking, by Kurose&Ross
❒ each RTCP packet
contains sender and/or
receiver reports
❍
report statistics useful
to application: #
packets sent, # packets
lost, interarrival jitter,
etc.
❒ feedback can be used
to control
performance
❍
sender may modify its
transmissions based on
feedback
4: Multimedia App. & Transp. 4-78
39
RTCP: multiple multicast senders
sender
RTP
RTCP
RTCP
RTCP
receivers
 each
RTP session: typically a single multicast address; all
RTP /RTCP packets belonging to session use multicast
address
 RTP, RTCP packets distinguished from each other via
distinct port numbers
 to limit traffic, each participant reduces RTCP traffic as
number of conference participants increases
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-79
RTCP: packet types
Receiver Report (RR)
packets:
❒ fraction of packets lost,
last sequence number,
average interarrival
jitter
Sender Report (SR)
packets:
❒ SSRC of RTP stream,
current time, number of
packets sent, number of
bytes sent
©From Computer Networking, by Kurose&Ross
Source Description
(SDES) packets:
❒ e-mail address of
sender, sender's name,
SSRC of associated
RTP stream
❒ provide mapping
between the SSRC and
the user/host name
4: Multimedia App. & Transp. 4-80
40
RTCP: stream synchronization
❒ RTCP can synchronize
different media streams
within a RTP session
❒ e.g., videoconferencing app:
each sender generates one
RTP stream for video, one
for audio
❒ timestamps in RTP packets
tied to the video, audio
sampling clocks
❍ not tied to wall-clock
time
❒ each RTCP sender-report
packet contains (for most
recently generated packet
in associated RTP stream):
❍
❍
timestamp of RTP packet
wall-clock time for when
packet was created
❒ receivers use association
to synchronize playout of
audio, video
4: Multimedia App. & Transp. 4-81
©From Computer Networking, by Kurose&Ross
RTCP: bandwidth scaling
❒ RTCP attempts to limit its
traffic to 5% of session
bandwidth
Example
❒ one sender, sending video
at 2 Mbps
❒ RTCP attempts to limit its
traffic to 100 kbps
❒ RTCP gives 75% of rate to
receivers; remaining 25%
to sender
©From Computer Networking, by Kurose&Ross
❒ 75 kbps is equally shared
among receivers:
❍
with R receivers, each
receiver gets to send RTCP
traffic at 75/R kbps
❒ sender gets to send RTCP
traffic at 25 kbps
❒ participant determines RTCP
packet transmission period by
calculating average RTCP
packet size (across entire
session) and dividing by
allocated rate
4: Multimedia App. & Transp. 4-82
41
SIP: Session Initiation Protocol [RFC 3261]
SIP long-term vision:
❒ all telephone calls, video conference calls take
place over Internet
❒ people are identified by names or e-mail
addresses, rather than by phone numbers
❒ you can reach callee (if callee so desires), no
matter where callee roams, no matter what IP
device callee is currently using
4: Multimedia App. & Transp. 4-83
©From Computer Networking, by Kurose&Ross
SIP Services
❒ SIP provides
mechanisms for call
setup:
❍ for caller to let
callee know she
wants to establish
a call
❍ so caller and
callee can agree
on media type,
encoding
❍ to end call
©From Computer Networking, by Kurose&Ross
❒ determine current IP
address of callee:
❍
maps mnemonic
identifier to current IP
address
❒ call management:
❍ add new media streams
during call
❍ change encoding during
call
❍ invite others
❍ transfer, hold calls
4: Multimedia App. & Transp. 4-84
42
Example: setting up a call to known IP address
Bob
Alice
167.180.112.24
193.64.210.89
INVITE bo
b@193.64
.2
c=IN IP4
167.180.11 10.89
2.24
m=audio
38060 RT
P/AVP 0
port 5060
port 5060
Bob's
terminal rings
200 OK
.210.89
c=IN IP4 193.64
RTP/AVP 3
m=audio 48753
ACK
port 5060
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

µ
µ Law audio
port 38060
GSM
Alice’s SIP invite
message indicates her
port number, IP address,
encoding she prefers to
receive (PCM µlaw)

default SIP port
number is 5060

port 48753
time
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-85
time
Setting up a call (more)
❒ codec negotiation:
suppose Bob doesn’t
have PCM µlaw
encoder
❍ Bob will instead reply
with 606 Not
Acceptable Reply,
listing his encoders
❍ Alice can then send
new INVITE
message, advertising
different encoder
❍
©From Computer Networking, by Kurose&Ross
❒ rejecting a call
Bob can reject with
replies “busy,”
“gone,” “payment
required,”
“forbidden”
❒ media can be sent over
RTP or some other
protocol
❍
4: Multimedia App. & Transp. 4-86
43
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
Alice sends, receives
SIP messages using
SIP default port 5060

Alice specifies in
“Via:” header that SIP
client sends, receives
SIP messages over
UDP

Notes:
❒ HTTP message syntax
❒ sdp = session description protocol
❒ Call-ID is unique for every call
©From Computer Networking, by Kurose&Ross
Here we don’t know
Bob’s IP address.
-> Intermediate SIP
servers needed

4: Multimedia App. & Transp. 4-87
Name translation and user location
❒ caller wants to call
callee, but only has
callee’s name or e-mail
address
❒ need to get IP address
of callee’s current
host:
❍
❍
❍
user moves around
DHCP protocol
user has different IP
devices (PC,
smartphone, car device)
©From Computer Networking, by Kurose&Ross
❒ result can be based on:
❍ time of day (work, home)
❍ caller (don’t want boss to
call you at home)
❍ status of callee (calls sent
to voicemail when callee is
already talking to
someone)
Service provided by SIP
servers
4: Multimedia App. & Transp. 4-88
44
SIP Registrar
registrar
❒ when Bob starts SIP client, client sends SIP
REGISTER message to Bob’s registrar server
❒ one function of SIP server:
Register Message:
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
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-89
SIP Proxy
proxy
❒ Alice sends invite message to her proxy server
❒ another function of SIP server:
❍
contains address sip:bob@domain.com
❒ proxy responsible for routing SIP messages to
callee Bob
❍
possibly through multiple proxies
❒ Bob sends response back through the same set of
proxies
❒ proxy returns Bob’s SIP response message to
Alice
❍
contains Bob’s IP address
❒ SIP proxy analogous to local DNS server plus TCP
setup
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-90
45
SIP example:
jim@umass.edu calls keith@poly.edu
2. UMass proxy forwards request
to Poly registrar server
2
3
UMass
SIP proxy
Poly SIP
registrar
3. Poly server returns redirect response,
indicating that it should try keith@eurecom.fr
4. Umass proxy forwards request
to Eurecom registrar server 4
1. Jim sends INVITE
8
message to UMass
SIP proxy.
1
128.119.40.186
7
6-8. SIP response returned to Jim
9
Eurecom SIP
registrar
5. eurecom
5 registrar
6
forwards INVITE
to 197.87.54.21,
which is running
Keith’s SIP client
9. Data flows between clients
197.87.54.21
Note: also a SIP ack message from Jim, which is not shown
4: Multimedia App. & Transp. 4-91
©From Computer Networking, by Kurose&Ross
Comparison with H.323
❒ H.323 is another signaling
protocol for real-time,
interactive
❒ H.323 is a complete,
vertically integrated suite
of protocols for multimedia
conferencing: signaling,
registration, admission
control, transport, codecs
❒ SIP is a single component.
Works with RTP, but does
not mandate it. Can be
combined with other
protocols, services
©From Computer Networking, by Kurose&Ross
❒ H.323 comes from the ITU
(telephony)
❒ SIP comes from IETF:
borrows much of its
concepts from HTTP
❍ SIP has Web flavor,
whereas H.323 has
telephony flavor
❒ SIP uses the KISS
principle: Keep It Simple
Stupid
4: Multimedia App. & Transp. 4-92
46
Chapter 4: Summary
Principles
❒ audio and video coding
❒ multimedia applications types over IP
❍
streaming stored audio video, real-time conversational voice/video
❒ UDP versus TCP streaming
❒ making the best of best effort service
❍
❍
❍
❍
DASH: Dynamic, Adaptive Streaming over HTTP
CDN: Content Distribution Networks
adaptive playout delay
loss recovery (FEC, retransmissions) and concealment
Protocols
❒ RTSP
❒ RTP/RTCP
❒ SIP
©From Computer Networking, by Kurose&Ross
4: Multimedia App. & Transp. 4-93
How should the Internet evolve to better
support multimedia?
Laissez-faire
Differentiated services
philosophy:
❒ just put more capacity where
needed
❒ fewer changes to Internet
❒ no major changes in network,
infrastructure, yet provide
let apps handle it
1st and 2nd class service
❒ content distribution networks,
application-layer multicast
Integrated services philosophy:
❒ fundamental changes in
Internet so that apps can
reserve end-to-end
bandwidth
❒ requires new, complex
software in hosts & routers
©From Computer Networking, by Kurose&Ross
What’s your opinion?
4: Multimedia App. & Transp. 4-94
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