Multimedia Networking Quality of Services Hongli Luo, IPFW

Multimedia Networking Quality of Services

Hongli Luo, IPFW

Topics

Multimedia networking applications Stored, live, interactive multimedia applications

How should Internet better support QoS Internet multimedia streaming

Web server or streaming server TCP or UDP

Packet loss, delay and jitter How to alleviate it

Multimedia and Quality of Service: What is it?

multimedia applications: network audio and video(“continuous media”)

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

QoS

MM Networking Applications

Fundamental characteristics:

typically delay sensitive end-to-end delay delay jitter

loss tolerant: infrequent losses cause minor glitches

Elastic applications Web, e-mail, FTP, telnet loss intolerant but delay

tolerant.

Classes of MM applications:

1) stored streaming2) live streaming3) interactive, real-time

Jitter is the variability of packet delays within the same packet stream

Streaming Stored Multimedia

Stored streaming: Applications: CNN, Microsoft Video, YouTube media stored at source transmitted to clientstreaming: client playout begins before all data has

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

for playout Continuous playout Streaming multimedia clients: RealPlayer, QuickTime,

Windows Media

Streaming Stored Multimedia: What is it?

1. videorecorded

2. videosent

3. video received,played out at client

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

networkdelay

time

Streaming Stored Multimedia: Interactivity

VCR-like functionality: client can pause, rewind, FF, push slider bar 10 sec initial delay OK 1-2 sec until command effect

OK

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

Streaming Live Multimedia

Examples: Internet radio talk show live sporting event IPTVStreaming (as with streaming stored multimedia) playback buffer playback can lag tens of seconds after

transmission still have timing constraintInteractivity fast forward impossible rewind, pause possible!

Streaming Live Multimedia

Live, broadcast like applications often have many clients who are receiving the same audio/video program

How to efficiently deliver the contents Multiple separate server-to-client unicast streams IP multicast Application-layer multicast – multicast overlay

networks Content Distribution Networks (CDNs) Peer-to-peer networks

Real-Time Interactive Multimedia

end-end delay requirements: audio: < 150 msec good, < 400 msec OK

• includes application-level (packetization) and network delays• higher delays noticeable, impair interactivity

session initialization how does callee advertise its IP address, port number, encoding

algorithms?

applications: IP telephony, video conference, distributed interactive worlds

Requirements for Data Transport

Delay Some small delay at the beginning is acceptable E.g., start-up delays of a few seconds are okay

Jitter Variability of packet delay within the same packet

stream Client cannot tolerate high variation if the buffer

starves

Loss Small amount of missing data does not disrupt

playback Retransmitting a lost packet might take too long

anyway

Multimedia Over Today’s InternetTCP/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!

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Challenges to the Current Internet (1)

TCP/UDP/IP suite provides best-effort, no guarantees on expectation or variance of packet delay

Streaming applications delay of 5 to 10 seconds is typical and has been acceptable, but performance deteriorate if links are congested (transoceanic)

Most router implementations use only First-Come-First-Serve (FCFS) packet processing and transmission scheduling

Challenges to the Current Internet (2)

To mitigate impact of “best-effort” protocols, we can: Use UDP to avoid TCP and its slow-start phase… Buffer content at client and delay playback to

remedy jitter Prefetch data during playback when client storage

and extra bandwidth are available Adapt compression level to available bandwidth Different treatment of packets at the router

• Different classes of packets

How should the Internet evolve to better support multimedia?

Integrated services philosophy: fundamental changes in Internet so that apps can

reserve end-to-end bandwidth Applications receive a guarantee on its end-to-end

performance Hard guarantee – received QoS with certainty Soft guarantee – received QoS with high

probability requires new, complex software in hosts & routers

Application makes reservation Modify scheduling policies at the router Description of traffic Admission control

How should the Internet evolve to better support multimedia?

Laissez-faire no major changes more bandwidth when needed content distribution,

Replicate stored content and put the replicated contents at the edges of the Internet

application-layer multicast Multicast Overlay networks can be deployed Application layer

Differentiated services philosophy: fewer changes to Internet infrastructure, yet provide 1st

and 2nd class service

A few words about audio compression analog signal sampled

at constant rate telephone: 8,000

samples/sec CD music: 44,100

samples/sec

each sample quantized, i.e., rounded e.g., 28=256 possible

quantized values

each quantized value represented by bits 8 bits for 256 values

example: 8,000 samples/sec, 256 quantized values --> 64,000 bps

receiver converts bits back to analog signal: some quality reduction

Example rates CD: 1.411 Mbps MP3: 96, 128, 160

kbps Internet telephony:

5.3 kbps and up

A few words about video compression

video: sequence of images displayed at constant rate e.g. 24 images/sec

digital image: array of pixels each pixel represented

by bits

redundancy spatial (within image) temporal (from one

image to next)

Examples: MPEG 1 (CD-ROM) 1.5

Mbps MPEG2 (DVD) 3-6 Mbps MPEG4 (often used in

Internet, < 1 Mbps)Research: layered (scalable)

video adapt layers to available

bandwidth


Client-server system Server stores the audio and video files Clients request files, play them as they download,

and perform VCR-like functions (e.g., rewind and pause)

Playing data at the right time Server divides the data into segments labels each segment with timestamp or frame id so the client knows when to play the data

Avoiding starvation at the client The data must arrive quickly enough otherwise the client cannot keep playing


application-level streaming techniques for making the best out of best effort service: client-side buffering use of UDP versus TCP multiple encodings of

multimedia

jitter removal decompression error concealment graphical user interface

w/ controls for interactivity

Media Player

Streaming stored multimedia Through a Web server

• Deliver the video/audio to the client over HTTP Through a streaming server

• Using HTTP or some other protocol HTTP popular – pass through firewall while

proprietary protocols are blocked

Internet multimedia: simplest approach – through a Web Server

audio or video stored in file files transferred as HTTP object

TCP connection between client and server

received in entirety at client then passed to player

Client invokes the media player Content-type indicates the encoding Browser launches the media player Media player then renders the file

Internet multimedia: through a Web Server

User clicks on a hyperlink for an audio/video file browser GETs metafile describing the object browser launches player, passing metafile player sets up its own connection to the Web server server streams audio/video to player

Streaming from a streaming server

Web server returns a meta file describing the object via HTTP Player requests the data using a different protocol allows for non-HTTP protocol between server, media player To control playback, protocol is needed to exchange

playback control information – RTSP protocol

Streaming Multimedia: Client Buffering

How to deliver media from the streaming server to the media player

Media is sent over UDP at a constant rate equal to the drain rate As soon as the client receives media, it decompresses it and

plays it back The media player delays playout for 2 to 5 seconds in order to

eliminate network-induced jitter. Media is sent over TCP

Fill rate x(t) fluctuates with time – due to TCP congestion control and window flow control

X(t) may be less that d for long periods because of congestion Empty client buffer - starvation/underflow Introduces undesirable pause during the playback X(t) very much depends on the size of the client buffer

• Large client buffer – TCP make use of the available bandwidth to prfetch, less likely starvation

• Small client buffer – more client starvation

constant bit rate videotransmission

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time

variablenetwork

delay

client videoreception

constant bit rate video playout at client

client playoutdelay

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


client-side buffering, Store the data as it arrives from the server Play data for the user in a continuous fashion

playout delay Client typically waits a few seconds to start playing Allow some data to build up in the buffer compensate for network-added delay, delay jitter

Streaming Multimedia: UDP or TCP?

TCP send at maximum possible rate under TCP fill rate fluctuates due to TCP congestion control

Slow down after a packet loss, May cause starvation at the client

HTTP/TCP passes more easily through firewalls TCP’s Reliable delivery not needed for

multimedia Retransmission of lost packets even though

retransmission may not be useful Late packet can be thrown away

Protocol overhead TCP header of 20 bytes in every packet


UDP server sends at rate appropriate for client

(oblivious to network congestion !) often send rate = encoding rate = constant rate then, fill rate = constant rate - packet loss No automatic adaptation of sending rate

short playout delay (2-5 seconds) to remove network jitter

error recover: time permitting No automatic retransmission of lost packets Retransmit if packet deadline not past

Smaller packet header


UDP Do not respond to congestion

Not sharing bandwidth fairly Possible to cause congestion collapse Put flow control, congestion control into application

UDP leaves many things to the application When to transmit data How to encapsulate the data Whether to retransmit lost data Whether to adapt the sending rate Whether to adapt the quality of the audio/video encoding


UDP Even when using UDP, applications should

respond to congestion end-to-end. Need to promote “TCP-friendly” behavior.

Rate-based Adaptation adjust rate to avoid congestion

reduce rate before packet loss or at the detection of packet loss.

TCP-Friendly

Throughput of a TCP connection

P: the packet size p: the lost probability of a packet

Limit flows to TCP-style BW Don’t know RTT exactly

)/(2.1 RTTpP

YouTube: HTTP, TCP, and Flash

Flash videos All uploaded videos converted to Flash format Nearly every browser has a Flash plug-in avoids need for users to install players

HTTP/TCP Implemented in every browser Easily gets through most firewalls

Keep It Simple, Stupid Simplicity more important than video quality

Real-time interactive applications PC-2-PC phone

Skype PC-2-phone

Dialpad, Net2phone, Skype videoconference with webcams

Skype, Polycom Video game QoS requirements:

Strict delay constraints Much harder than streaming applications Receiver can not introduce much playback

delay

Going to now look at a PC-2-PC Internet phone example in detail

Interactive Multimedia: Internet Phone

Introduce Internet Phone by way of an example

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 data

application-layer header added to each chunk.

chunk+header encapsulated into UDP segment.

application sends UDP segment into socket every 20 msec during talkspurt

Internet Phone: Packet Loss and Delay

network loss: IP datagram lost due to network congestion (router buffer overflow)

delay loss: IP datagram arrives too late for playout at receiver

loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10% can be tolerated.

Most Internet phone applications that run over UDP do not retransmit lost packets.

Recovering From Packet Loss

Loss is defined in a broader sense Does a packet arrive in time for playback? A packet that arrives late is as good as lost Retransmission is not useful if deadline passed

Selective retransmission Sometimes retransmission is acceptable E.g., if client has not already started playing data Data can be retransmitted within time constraint

Could do Forward Error Correction (FEC) Send redundant info so receiver can reconstruct

Internet Phone: Packet Loss and Delay

Packet delay delays: processing, queueing in network;

end-system (sender, receiver) delays For highly interactive audio applications,

e.g., Internet phone• End-to-end delays smaller than 150 ms

are not perceived by a human listener• Delays between 150 and 400 ms

acceptable but not ideal• typical maximum tolerable delay: 400 ms

Removing Jitter at the Receiver for Audio

Provide synchronous playout of voice chunks in the presence of random network jitter

To alleviate the jitter Sequence number is added to each chunk Timestamp – sender stamps each chunk with the

time at which the chunk was generated Delaying playout

• Playout delay must be long enough so that most of the packets are received before the scheduled playout times

• Fixed playout delay• Adaptive playout delay

Internet 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 in choosing q:

large q: less packet loss• When large variations in end-to-end delay

small q: better interactive experience• Delay and variations in delay are small• .e.g., < 150 ms

constant bit ratetransmission

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variablenetwork

delay(jitter)

clientreception

constant bit rate playout at client

client playoutdelay

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Delay Jitter

consider end-to-end delays of two consecutive packets: difference can be more or less than 20 msec (transmission time difference)

Fixed Playout Delay

packets

time

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’

Adaptive Playout Delay Goal: minimize playout delay, keeping late

loss rate low 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.

Recovery from packet loss (1)

Forward Error Correction (FEC): simple scheme

for every group of n chunks create redundant chunk by exclusive OR-ing n original chunks

send out n+1 chunks, increasing bandwidth by factor 1/n.

can reconstruct original n chunks if at most one lost chunk from n+1 chunks

playout delay: 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


2nd FEC scheme “piggyback lower quality stream” send lower resolutionaudio stream as redundant information e.g., nominal stream PCM at 64 kbpsand redundant streamGSM at 13 kbps.

whenever there is non-consecutive loss, receiver can conceal the loss. can also append (n-1)st and (n-2)nd low-bit ratechunk


Interleaving chunks divided into smaller units for example, four 5 msec units

per chunk packet contains small units from

different chunks

if packet lost, still have most of every chunk

no redundancy overhead, but increases playout delay

Documents

Multimedia Networking Quality of Services Hongli Luo, IPFW