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Multimedia Networking Sound, Video

Multimedia Networking Sound, Video. Applications Downloadable files Streaming Video Streaming Audio VoIP (Voice over IP) –Internet Phone

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Multimedia Networking

Sound, Video

Applications

• Downloadable files

• Streaming Video

• Streaming Audio

• VoIP (Voice over IP)– Internet Phone

Problems for multimedia

• Best-effort-service– No promises about delay

• Packet jitter– Variable packet delays

• Dropped packets– Time delay caused by out of sequence

packets

Fixes

• Faster internet ISPs– More bandwidth for the Internet

• Content distribution networks (CDNs)– Distribute the data over a wider area (edges of the

internet)

• Multicasting (multicast overlay network)– Not IP multicast but a server based distributed

multicast.

Human Interface

• Hearing limitations– Ear response 20 to 20,000 Hertz– Networking speeds are up to 100,000,000,000

hertz– Phonemes are from 5-100msec– Ear does not respond to delays < 150msec

• Visual limitations– The eye retention is about 60 msec.– Any transfer faster than 16 frames per second will

not appear to flicker.

Data Preparation

• Audio and Video compression

– Sampling rate

– Quantization (rounding samples to finite values)

– Fit the quantization values into a fixed size

– Pulse Code Modulation (PCM)

• Sample rate * # of bits * 2 (if stereo)

– Compression

• GSM (Global System Mobile) or G.723.3 (ADPCM)

• MPEG 1 layer 3 or MP3

• Coding/Decoding

– Codecs

Video Compression

• MPEG1 CD-ROM quality

• MPEG 2 for DVD quality

• MPEG4 for object oriented

• H.261

• Proprietary– Apple Quick Time– Real Networks

Protocols

• RTP (Real Time Protocol)

• RTSP (Real Time Streaming Protocol)

• These are not IP protocols but are accessed via help applications in browsers

• They provide:– Decompression– Jitter Removal– Error Correction

Types of transfer

• Web server file access– Download the multimedia file– Play the file on Windows media player or your

favorite version of player.– Requires local storage of the file

• Streaming– Access the audio/video stream via web server– Playback portions of the stream in real time

via the streaming server.

RTSP

• Real Time Streaming protocol– Does not set compression schemes– Does not define encapsulation– Does not restrict the transport– Does not restrict buffering

• Uses out-of-band signaling (separate port)

Internet Phone Example

• Packet Loss

• Packet Jitter

Packet Loss Recovery

• Forward Error Correction– Piggybacking redundant information– Utilizes lower bit rate chunks for redundancy– Works well for harsh environments

• Interleaving– Splitting the stream across several packets– Increased latency (delays)

• Packet repetition– Copies good packets to replace lost ones.– Works well for small loss rates

End to End Delays

• The human ear smaller than 150msec are indiscernible to the human ear

• If the delay exceeds 400msec then the packet is lost.

Packet Jitter

• Delay fluctuations between packets.

• Removing Jitter– Use sequence numbers, timestamps and:– Fixed Playout Delay

• Delay before playing packet• Phone system can handle 400 msecs of delay

– Adaptive Playout delay• Adjust delay depending on the flow of data

Real Time Protocol

• Runs on top of UDP (for speed)• Packet header includes:

– Payload type– Sequence number– Timestamp– Sync source identification number– Miscellaneous

• Defines which coding method is used and timing to overcome jitter.

RTCP

• Real Tine Control Protocol

• Used to communicate information from source to destination.– Sync Source ID– Packets lost– Last sequence number– Number of bytes in the stream– Scaling parameters (packet size)

Session Initiation Protocol

• Internet Phone

• Sorta like Telnet for the phone system

Name Translation

• SIP Proxy– Authenticates SIP invitations

• SIP registrar– Keeps track of the users IP address

H.323

• Internet Phone to PSTN (Public Switched Telephone Network)– Defines how endpoints negotiate audio/visual

encodings– Defines how to encapsulate data– Defines how to communicate with

gatekeepers – Defines how VoIP interfaces with ordinary

telephones.

Queuing Schemes

• FIFO (First In First Out)– Every packet in order regardless of class

Queuing Schemes

• Priority Queuing– Based on priority class in header

Queuing Schemes

• Round Robin/Weighted Fair Queuing– Based on class of service

Reservation Protocol

• RSVP– Provides for resource reservation in multicast

trees– Receiver oriented, receiver initiates and

maintains resource reservations

• Handles the problem of different speed receivers all following the same video/audio stream

Tour of USU Network Infrastructure

Borrowed Space in Telephone Switching Site

Surge Protectors

Cross-Connect Blocks

Telephone Equipment

Wiring trays above is typical for telephone locations

Now that’s a Ground!

10BaseT Hubs

10/100 switchwith FDDI

FDDI Concentrator

10/100 switch

Router

Media Converters

Fiber Optics

Close up View

Media Converters

Computer Room in SER Building

Servers, Building Network, and Campus Networking

Internet Feed

Cisco Border Router

Pix Fire WallUSU/UEN

Interface

Primary USU CiscoRouter Rest of USU Campus

UEN Internet

FeedSalt Lake

180 Mbs

Packeteer

Housing

Connections for building network

StudentHousing

Other Campus Network Equipment

Front Side of Equipment Racks

Inside of Fiber Panels

Internet gateway router

Switch with router card & 100BaseFX connections

Misc. Novellservers

VMS clustera.k.a.

cc.usu.edu

barney sneezy

opie

grumpy

sleepy

Cluster Connections

.89 .86 .85 .87 .150

Foundry Load BalancingSwitch

Barney Grumpy Sleepy Sneezy Opie

USU Backbone

129.123.1.1cc.usu.edu

Multimedia Networking