CE80N Introduction to Networks & The Internet Dr. Chane L. Fullmer UCSC Winter 2002

CE80NCE80NIntroduction to NetworksIntroduction to Networks

&&The InternetThe Internet

Dr. Chane L. Fullmer

UCSC

Winter 2002

February 28, 2002 CE80N -- Lecture #16, 2002 2

Class InformationClass InformationWeb page tutorial available on-lineWeb page submission:

– Email to [email protected]• Subject: cmpe080n-assgn4

Final Exam – Last class session

• March 14, 2002

mailto:[email protected]


Personal Web Page of the DayPersonal Web Page of the Day

A few brave souls….Presenting:

AKA: Multimedia Networking


Description of FunctionalityDescription of Functionality

Internet audio and video services make it possible to:– Send voice– Send live television images– Broadcast audio or video

signals– Allow viewing and editing a

document simultaneously


Audio And Video Require Audio And Video Require Special HardwareSpecial Hardware

Live audio or video require high bandwidth.– Requires a computer with:

• A microphone• A speaker• A camera• A high-speed processor

– Controls all devices electronically


Multimedia NetworkingMultimedia NetworkingOur goals: principles: network,

application-level support for multimedia– different forms of

network multimedia, requirements

– making the best of best effort service

– mechanisms for providing QoS

specific protocols, architectures for QoS

Overview: multimedia applications

and requirements– Audio– Video

making the best of today’s best effort service

Examples– RealPlayer– ISDN Videoconferencing


Digital Audio and VideoDigital Audio and Video

Audio: Air pressure converted

by microphone to voltage.– Magnitude represents

loudness or softness– Frequency

represents pitch, timbre.

Sampled: Voltage saved at discrete points in time

Quantized: Rounded off to a discrete value (x).

Video: Light converted by camera

into chemical deposition– Magnitude represents

brightness– Frequency represents edges,

contrast Sampled: Values saved at

each horizontal/vertical (x,y) position in time

Quantized: Rounded off to a discrete value (z) for each point (x,y).


Digital Audio numbersDigital Audio numbers Sample Size: 16 bit (2 byte) data representation Channels: 2 channels (stereo) Sampling Rate: 44,100 Samples Per Second

(CD quality) One Channel: 2 bytes per sample x 44,100

samples per second = 88,200 bytes per second Total Data Rate: 88,200 bytes per second x 2

channels = 176,400 bytes per second Total Bit Rate = 1,411,200 bits per second

– This is DS1 rate!.. ADSL is typically a 384K downlink


ImagesImages

YCC encoding scheme– Luminance = brightness (Y)– Chrominance = amount of color

• Cb = amount of blue• Cr = amount of red

Source: Peter Bourke http://astronomy.swin.edu.au/pbourke/colour/ycc/


Color Representation in YCCColor Representation in YCC

Luminance = brightness (Y)

Source: Peter Bourkehttp://astronomy.swin.edu.au/pbourke/colour/ycc/



Chrominance–Cr = amount of red




Chrominance–Cb = amount of blue



YCCYCCEncodingEncoding

Luminance stored for each pixel Chrominance stored for each 4x4 block of

pixelsSource: Peter Bourkehttp://astronomy.swin.edu.au/pbourke/colour/ycc/


Digital Video numbersDigital Video numbers 720 x 486 pixels per frame 29.97 frames per second (Fps) Sample Size - 8 bits per pixel data representation Sampling - 4:2:2 (Every two horizontal pixels = 2 Y : 1

Cr : 1 Cb) – Luminance (Y)

• 720 x 486 x 29.97 x 8 = 83,896,819.2 bits per second

– Chrominance R (Cr) • 360 x 486 x 29.97 x 8 = 41,948,409.6 bits per second

– Chrominance B (Cb) • 360 x 486 x 29.97 x 8 = 41,948,409.6 bits per second

Total = ~20 Megabytes per second– Phew…… !! Ethernet is only 1.25 Mbyte/s…


Compression Coding MethodsCompression Coding MethodsCompression:

– Transmission of the same information in fewer bits. Run-length coding: Encode as symbol followed by

number of symbols in a row.– “0,0,0,0,0,0,0,0,0 ….0” replaced by “0 256”

Huffman coding: (David Huffman, 1952)– Builds a tree of symbols, assigning shorter bit codes to the

more common symbols Arithmetic coding:

– Converts input symbols to a single real number Standards

– Images - JPEG (Joint Picture Experts Group)– Video - MPEG (Moving Picture Experts Group)

• MP3 for audio


Audio and Video CompressionAudio and Video Compression

Compression: Transmission of the same information in fewer bits.

Audio:– Silence is transmitted as blank space in run-

length coding.Video:

– Most objects stay fixed from frame to frame.– Differences between frames transmitted.


Image CompressionImage Compression

Can save a lot of bits!– Left-most picture: 24 bits per pixel– Right-most picture: 9 bits per pixel

Can you tell the difference?



Image CompressionImage Compression

Can be lossy– Left-most picture: Original– Center: 10:1 compression– Right-most picture: 45:1 compression

Can you tell the difference?

Source: Steven Smith http://www.dspguide.com/datacomp.htm


Audio quality and Transmission Audio quality and Transmission RatesRatesSound quality Bandwidth Mode Bitrate Reduction

ratio

Telephone 2.5 kHz Mono 8 kbps 96:1

Shortwave 4.5 kHz Mono 16 kbps 48:1

AM radio 7.5 kHz Mono 32 kbps 24:1

FM 11 kHz Stereo 56…64 kbps 26…24:1

Near-CD 15 kHz Stereo 96 kbps 16:1

CD >15 KHz Stereo 112..128kbps 14:12:1


Video quality and Transmission RatesVideo quality and Transmission Rates


Transmission RatesTransmission Rates

• Useful chart of transmission rates:http://the-mid-west-web.com/thespeed.htm

http://the-mid-west-web.com/thespeed.htm

http://the-mid-west-web.com/thespeed.htm


Multimedia, Quality of Service: Multimedia, Quality of Service: What is it?What is it?

Multimedia applications: network audio and video

network provides application with level of performance needed for application to function.

QoS

Source: http://www-net.cs.umass.edu/cs591/Professor Jim KuroseUsed with permissionSlides 18-31, 34-51

http://www-net.cs.umass.edu/cs591/








Multimedia Performance Multimedia Performance RequirementsRequirements

Requirement: deliver data in “timely” manner interactive multimedia: short end-end delay

– e.g., IP telephony, telecon, virtual worlds– excessive delay impairs human interaction

streaming (non-interactive) multimedia:– data must arrive in time for “smooth” playout– late arriving data introduces gaps in rendered

audio/video

reliability: 100% reliability not always required


Interactive, Real-Time Interactive, Real-Time Multimedia Multimedia

end-end delay requirements:– video: < 150 msec acceptable– audio: < 150 msec good, < 400 msec OK– includes application-level (packetization) and network

delays– higher delays noticeable, impair interactivity

applications: IP telephony, video conference, distributed interactive worlds


InteractiveInteractive Multimedia: Multimedia: VideoconferencingVideoconferencingIntroduce Internet Phone by way of an example

(note: there is no “standard” yet): speaker’s audio: alternating talk spurts and silent

periods. pkts generated only during talk spurts

– E.g., 20 msec chunks at 8 Kbytes/sec: 160 bytes data

application-layer header added to each chunk. Chunk+header encapsulated into UDP segment. application sends UDP segment into the network

every 20 msec during talkspurt.


Internet Phone: Packet Loss and Internet Phone: Packet Loss and DelayDelay 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, losses can be concealed, packet loss rates between 1% and 10% can be tolerated.


constant bit ratetransmission

Cum

ula

tive

data

time

variablenetwork

delay(jitter)

clientreception

constant bit rate playout at client

client playoutdelay

bu

ffere

ddata

Delay JitterDelay Jitter

Client-side buffering, playout delay compensate for network-added delay, delay jitter


Internet Phone: Fixed Playout DelayInternet Phone: Fixed Playout Delay

Receiver attempts to playout each chunk exactly q msecs after chunk was generated.– chunk has time stamp t: play out chunk at

t+q .– chunk arrives after t+q: data arrives too

late for playout, data “lost” Tradeoff for q:

– large q: less packet loss– small q: better interactive experience


Fixed Playout DelayFixed Playout Delay

packets

tim e

packetsgenerated

packetsreceived

loss

r

p p '

playout schedulep' - r

playout schedulep - r

• 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’


Recovery From Packet LossRecovery From Packet Loss

loss: pkt never arrives or arrives too late real-time constraints: little (no) time for

retransmissions!– What to do?

Forward Error Correction (FEC): add error correction bits (recall parity bits?)– e.g.,: add redundant chunk made up of exclusive OR of n

chunks; redundancy is 1/n; can reconstruct if at most one lost chunk

Interleaving: spread loss evenly over received data to minimize impact of loss


InterleavingInterleaving

Has no redundancy, but can cause delay in playout beyond Real Time requirements

Divide 20 msec of audio data into smaller units of 5 msec each and interleave

Upon loss, have a set of partially filled chunks


Piggybacking Lower Quality Piggybacking Lower Quality StreamStream


Streaming MultimediaStreaming Multimedia

Streaming: media stored at source transmitted to client streaming: client playout begins

before all data has arrived

timing constraint for still-to-be transmitted data: in time for playout


Streaming: what is it?Streaming: what is it?

1. videorecorded

2. videosent

3. video received,played out at client

Cum

ula

tive

data

streaming: at this time, client playing out early part of video, while server still sending laterpart of video

networkdelay

time


Streaming MultimediaStreaming Multimedia (more) (more)

Types of interactivity: none: like broadcast radio, TV

– initial startup delays of < 10 secs OK VCR-functionality: client can pause, rewind,

FF– 1-2 sec until command effect OK

timing constraint for still-to-be transmitted data: in time for playout


Multimedia Over Today’s Multimedia Over Today’s InternetInternetTCP/UDP/IP: “best-effort service” no guarantees on delay, loss

Today’s Internet multimedia applications use application-level techniques to mitigate

(as best possible) effects of delay, loss

But you said multimedia apps requiresQoS and level of performance to be

effective!

?? ???

?

? ??

?

?


Streaming Internet MultimediaStreaming Internet Multimedia

Application-level streaming techniques for making the best out of best effort service:– what is streaming?– client side buffering– multiple rate encodings of multimedia

….. let’s look at these …..


Internet multimedia: simplest Internet multimedia: simplest approachapproach

audio or video stored in file files transferred as HTTP

object– received in entirety at

client– then passed to player

audio, video not streamed: no, “pipelining,” long delays until playout!


Internet multimedia: streaming Internet multimedia: streaming approachapproach

browser GETs metafile browser launches player, passing metafile player contacts server server streams audio/video to player


Downloadable vs. StreamingDownloadable vs. StreamingDownloadable: Download and save Listen later or send

to others Standards-based Better quality Bandwidth burden

on user Downloading takes

time

Streaming: Disposable Listen “on-the-fly” live Proprietary Lower quality Bandwidth burden on the

developer More resources/time/cost No time lost in

downloading Live broadcasting


constant bit rate videotransmission

Cum

ula

tive

data

time

variablenetwork

delay

client videoreception

constant bit rate video playout at client

client playoutdelay

bu

ffere

dvid

eo

Streaming Multimedia: Client Streaming Multimedia: Client BufferingBuffering



Streaming Multimedia: Client Streaming Multimedia: Client BufferingBuffering


bufferedvideo

variable fillrate, x(t)

constant drainrate, d


Streaming Multimedia:Streaming Multimedia: client rate(s) client rate(s)

Q: how to handle different client receive rate capabilities?– 28.8 Kbps dialup– 100Mbps Ethernet

A: server stores, transmits multiple copies of video, encoded at different rates

1.5 Mbps encoding

28.8 Kbps encoding


User control of streaming multimediaUser control of streaming multimedia

Real Time Streaming Protocol (RTSP): RFC 2326 user control: rewind, FF, pause, resume, etc… out-of-band protocol:

– one port (544) for control msgs– one port for media stream

TCP or UDP for control msg connection

Scenario: metafile communicated to web browser browser launches player player sets up an RTSP control connection, data

connection to server


Metafile ExampleMetafile 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>


RTSP OperationRTSP Operation


Summary: Summary: Internet Multimedia: bag Internet Multimedia: bag of tricksof tricks

use UDP to avoid TCP congestion control (delays) for time-sensitive traffic

client-side playout delay: to compensate for network delay/jitter

server side matches stream bandwidth to available client-to-server path bandwidth– chose among pre-encoded stream rates– dynamic server encoding rate

error recovery (on top of UDP)– FEC– retransmissions, time permitting– mask errors: repeat nearby data


Example: RealPlayerExample: RealPlayer

RealNetworks1995, first streaming Internet audio

(Progressive Networks)1997

– RealSystem (RealVideo, RealAudio, and RealFlash)

– RealServer (client/server software)Uses RTSP


Example: RealPlayerExample: RealPlayer

Applications: Internet server

– Wide audience (most complex/expensive)– video commercials/e-commerce capabilities

Intranet server – Internal to business– desk-top training for employees

Commerce Solution – Secure transmissions to small groups– B2B, distance-learning, briefings


Radio Programs On The InternetRadio Programs On The Internet

A radio station uses computer equipment that converts the audio signal to digital form.– Requires software that

contacts the station– Extracts digitized audio

from the packets– Converts the data to sound– Plays the sound

KPIG Radio on the Internet

http://www.kpig.com/kpig_128.ram


Real-Time Transmission Is Real-Time Transmission Is Called WebcastingCalled Webcasting

To enable a browser to play real-time audio or video, the browser must be extended with a plugin.– Consists of software

• Ie, Real Audio, M$ Media Player

– Keeps a list of packets in memory– Enables the browser to receive and play

audio/videoMonterey Bay Aquarium live videocast

http://www.montereybayaquarium.org/efc/efc_fo/fo_ottr_cam.asp


Software for CollaborationSoftware for Collaboration

Software exists that permits a group of users to examine and edit a single document.– Called a

whiteboard service


Marking A DocumentMarking A Document

A participant uses a mouse to control interaction with the whiteboard service.– Allows all users to see changes on their

own screens, as well as reflected on the screens of others


The Participants Discuss And The Participants Discuss And Mark A DocumentMark A Document

Usually participants may also engage in an audio teleconference at the same time.

Changes:– Can be noted in colors– Saved to a word processing document


Video TeleconferencingVideo Teleconferencing

Video teleconferencing enables face-to-face communication.– Requires software to

start a video session• M$ NetMeeting• Vic/vat for Unix

– Incorporates both video and audio technology


Video Teleconference Among Video Teleconference Among Groups Of PeopleGroups Of People

The screen cannot hold individual images from many cameras.– Gather people in smaller

groups with a:• Camera• Large screen monitor or TV• Microphones for audio


A Combined Audio, Video, And A Combined Audio, Video, And Whiteboard ServiceWhiteboard Service

Teleconferencing becomes more interesting when combined with a whiteboard service.– Provides the most flexibility


ConclusionConclusion

How do you think telephone companies view Internet telephone service? (What!? Free long distance?)

Would you be willing to watch the evening news on your computer?


GlossaryGlossary

Audio Teleconference – A service that allows a group of users to

exchange audio information over the Internet similar to a telephone conference call.

Whiteboard Service – A service that permits a group of users to

establish a session that enables all of them to see and modify the same display.


GlossaryGlossary

Bandwidth – The capacity of a network, usually measured in

bits per second. Video Teleconference Service

– A service that allows a group of users to exchange video information over the Internet.

Delay Jitter– The variance in delay seen at the receiver of an

audio/video stream

Documents

CE80N Introduction to Networks & The Internet Dr. Chane L. Fullmer UCSC Winter 2002