Quality of Service For Voip

Quality of Service for VoIP
Dr. Danny Tsang Department of Electronic & Computer Engineering Hong Kong University of Science and Technology
Quality of Service
Multimedia, Quality of Service: What is it?

Multimedia applications: network audio and video (continuous media)
QoS
network provides application with level of
performance needed for application to function.

Quality of Service 2
Quality of Service: Goals

Principles Classify multimedia applications Identify the network services the apps need Making the best of best effort service Mechanisms for providing QoS QoS issues for VoIP Service Considerations Audio compression standards Jitter mitigation Loss recovery Echo cancellation Silence Suppression
Architectures for QoS TOS, Diffserv, Intserv, RSVP, MPLS, 802.1p/q

Voice activity detection (VAD) Comfort noise generation (CNG)
Quality of Service: Outline

Delay & loss in packet-switched networks
Multimedia Networking Applications

Real-time Multimedia: VoIP study QoS issues for VoIP Beyond Best Effort Scheduling and Policing Mechanisms Integrated Services and Differentiated Services RSVP MPLS Virtual LANs (802.1p/q)
Quality of Service
How do loss and delay occur?

packets queue in router buffers
packet arrival rate to link exceeds output link capacity packets queue, wait for turn
packet being transmitted (delay)
A B
packets queueing (delay) free (available) buffers: arriving packets dropped (loss) if no free buffers
Quality of Service
Four sources of packet delay

1. nodal processing:

2. queueing

check bit errors determine output link
time waiting at output link for transmission depends on congestion level of router
A B
transmission
propagation
nodal processing
queueing
Delay in packet-switched networks

3. Transmission delay: R=link bandwidth (bps) L=packet length (bits) time to send bits into link = L/R 4. Propagation delay: d = length of physical link s = propagation speed in medium (~2x108 m/sec) propagation delay = d/s
A B
transmission
Note: s and R are very different quantities!

propagation
nodal processing
queueing
Quality of Service
Nodal delay
d nodal d proc d queue d trans d prop
dproc = processing delay
typically a few microsecs or less
dqueue = queuing delay
depends on congestion
= L/R, significant for low-speed links
dtrans = transmission delay
dprop = propagation delay
a few microsecs to hundreds of msecs
Quality of Service
Queueing delay (revisited)

R=link bandwidth (bps) L=packet length (bits) a=average packet arrival
rate
traffic intensity = La/R

La/R ~ 0: average queueing delay small La/R -> 1: delays become large La/R > 1: more work arriving than can be serviced,
average delay infinite!
Quality of Service
Real Internet delays and routes

What do real Internet delay & loss look like? Traceroute program: provides delay measurement from
source to router along end-end Internet path towards destination. For all i:

sends three packets that will reach router i on path towards destination router i will return packets to sender sender times interval between transmission and reply.
3 probes 3 probes
3 probes
Quality of Service
10
Real Internet delays and routes

traceroute: gaia.cs.umass.edu to www.eurecom.fr
Three delay measements from gaia.cs.umass.edu to cs-gw.cs.umass.edu
1 cs-gw (128.119.240.254) 1 ms 1 ms 2 ms 2 border1-rt-fa5-1-0.gw.umass.edu (128.119.3.145) 1 ms 1 ms 2 ms 3 cht-vbns.gw.umass.edu (128.119.3.130) 6 ms 5 ms 5 ms 4 jn1-at1-0-0-19.wor.vbns.net (204.147.132.129) 16 ms 11 ms 13 ms 5 jn1-so7-0-0-0.wae.vbns.net (204.147.136.136) 21 ms 18 ms 18 ms 6 abilene-vbns.abilene.ucaid.edu (198.32.11.9) 22 ms 18 ms 22 ms 7 nycm-wash.abilene.ucaid.edu (198.32.8.46) 22 ms 22 ms 22 ms trans-oceanic 8 62.40.103.253 (62.40.103.253) 104 ms 109 ms 106 ms link 9 de2-1.de1.de.geant.net (62.40.96.129) 109 ms 102 ms 104 ms 10 de.fr1.fr.geant.net (62.40.96.50) 113 ms 121 ms 114 ms 11 renater-gw.fr1.fr.geant.net (62.40.103.54) 112 ms 114 ms 112 ms 12 nio-n2.cssi.renater.fr (193.51.206.13) 111 ms 114 ms 116 ms 13 nice.cssi.renater.fr (195.220.98.102) 123 ms 125 ms 124 ms 14 r3t2-nice.cssi.renater.fr (195.220.98.110) 126 ms 126 ms 124 ms 15 eurecom-valbonne.r3t2.ft.net (193.48.50.54) 135 ms 128 ms 133 ms 16 194.214.211.25 (194.214.211.25) 126 ms 128 ms 126 ms 17 * * * * means no reponse (probe lost, router not replying) 18 * * * 19 fantasia.eurecom.fr (193.55.113.142) 132 ms 128 ms 136 ms
Packet loss
queue (aka buffer) preceding link in buffer has
finite capacity when packet arrives to full queue, packet is dropped (aka lost) lost packet may be retransmitted by previous node, by source end system, or not retransmitted at all
Quality of Service
12


Quality of Service
13
MM Networking Applications
Classes of MM applications: 1) Streaming stored audio and video 2) Streaming live audio and video 3) Real-time interactive audio and video Fundamental characteristics: Typically delay sensitive

end-to-end delay delay jitter
But loss tolerant:
Jitter is the variability of packet delays within the same packet stream
infrequent losses cause minor glitches Antithesis of data, which are loss intolerant but delay tolerant.
Quality of Service
14
Streaming Stored 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
Quality of Service
15
Streaming Stored Multimedia: What is it?
1. video recorded
2. video sent
network delay
3. video received, played out at client time
streaming: at this time, client
playing out early part of video, while server still sending later part of video
Streaming Stored Multimedia: Interactivity
pause, rewind, FF, push slider bar 10 sec initial delay OK 1-2 sec until command effect OK RTSP often used (more later) timing constraint for still-to-be transmitted data: in time for playout
VCR-like functionality: client can
Streaming Live Multimedia

Examples: Internet radio talk show Live sporting event Streaming playback buffer playback can lag tens of seconds after transmission still have timing constraint Interactivity fast forward impossible rewind, pause possible!
Interactive, Real-Time Multimedia
video conference, distributed interactive worlds end-end delay requirements: audio: < 150 msec good, < 400 msec OK
includes application-level (packetization) and network delays higher delays noticeable, impair interactivity
applications: IP telephony,
session initialization
how does callee advertise its IP address, port number, encoding algorithms? Quality of Service
19
Multimedia Over Todays Internet

TCP/UDP/IP: best-effort service
no guarantees on delay, loss
But you said multimedia apps requires ? QoS and level of performance to be ? ? effective! ?
Todays Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss
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 Requires new, complex software in hosts & routers Laissez-faire no major changes more bandwidth when needed content distribution, application-layer multicast
Differentiated services philosophy: Fewer changes to Internet infrastructure, yet provide 1st and 2nd class service.
application layer
Whats your opinion?

A few words about audio compression

Analog signal sampled
at constant rate

Example: 8,000
telephone: 8,000 samples/sec CD music: 44,100 samples/sec
Each sample quantized,
samples/sec, 256 quantized values --> 64,000 bps Receiver converts it back to analog signal:
i.e., rounded
some quality reduction
e.g., 28=256 possible quantized values
Each quantized value
represented by bits
8 bits for 256 values
Example rates CD: 1.411 Mbps MP3: 96, 128, 160 kbps Internet telephony: 5.3 - 13 kbps
A few words about video compression

Video is sequence of
images displayed at constant rate
e.g. 24 images/sec
Digital image is array of
pixels Each pixel represented by bits Redundancy
Examples: MPEG 1 (CD-ROM) 1.5 Mbps MPEG2 (DVD) 3-6 Mbps MPEG4 (often used in Internet, < 1 Mbps) Research: Layered (scalable) video
spatial temporal
adapt layers to available bandwidth
Quality of Service
23


Quality of Service
24
Real-time interactive applications

PC-2-PC phone instant messaging services are providing this PC-2-phone
Going to now look at a PC-2-PC VoIP example in detail
Dialpad Net2phone videoconference with Webcams
Quality of Service
25
Interactive Multimedia: VoIP

Introduce VoIP by way of an example
speakers 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.
Quality of Service
26
VoIP: Real-Time Delivery

Real-time application data must be delivered with the same time relationship as it was created (but with some delay) Two aspects of real-time delivery (for protocols): Order: data should be played in the same order as it was created Time: the receiver must know when to play the packets, in order to reproduce the same signal as was input Tracking packet order by using a sequence number
and a time stamp for timing
Quality of Service
27
Key Challenges in VoIP

QoS requirements, customer expectations Low delay Low jitter Voice consistent best effort IP networks can offer NO latency guarantee NO bandwidth guarantee CAN ONLY guarantee CHEAP!
Quality of Service
28
Service Considerations in VoIP

Delay Jitter Packet loss Packet mis-order Available bandwidth Network design Endpoint audio characteristics (sound card, microphone, earpiece, etc.)
Transcoding Echo Silence suppression Duplex Codec selection Router and data-switch configuration Wan protocols QoS/CoS policy Encryption Decryption
Quality of Service
29
Delay
Time cost to traverse the networks
Voice calls are real-time, full-duplex
communications, end-to-end delay of packets can have severe impacts on usability of the VoIP applications. Contributed by
Propagation delay
Processing delay or Handling delay (more serious)
caused by the characteristics of the speed of light traveling via a fiber-optic-based or copper-based medium the devices that handle voice information switch, routers, firewall, etc. Time cost to generate a voice packet the DSP generates a frame every 10 milliseconds. Two of these frames are then placed within one voice packet; the packet delay is therefore 20 milliseconds
End-to-end delay components in VoIP
Quality of Service
31
Delay may cause

Voice traffic is real-time traffic, a long
delay may cause

speech
unrecognizable Talk over

Each person starts to talk because the delay prevents them from realizing that the other person has already started talking
An acceptable delay is less than 200
milliseconds
Quality of Service
32
Usability of a voice circuit as a function of end-to-end delay[4]
Citizens Band (CB) Radio Service is a private two-way voice communication service for use in personal and business activities of the general public. Its communications range is from one to five miles.
Quality of Service
33
ITU delay recommendations (G.114)

One Way Delay 0-150ms 150-400ms Description Acceptable for most user applications Acceptable provided that administrators are aware of the transmission time impact on the transmission quality of user applications. Unacceptable for general network planning purposes; however, it is recognized that in some exceptional cases this limit will be exceeded.
400ms+
Quality of Service
34
Jitter
Packets have varying transmission times Variation between when a voice packet is expected to be
received and when it actually is received (variations in delay ) Occur when packets get held up in queues because of congestion within the internet Causing a discontinuity in the real-time voice stream Approaches to compensate
Jitter buffer buffer packets with a specified period to smooth packet flow
To increase the size of the jitter buffer However, jitter buffer increases overall delay
Adjust queuing methods
Quality of Service
35
VoIP: Packet Loss and Delay

network loss:

IP datagram lost due to network congestion (router buffer overflow) Due to network outages, network re-routing, etc. delays: processing, queueing in network; end-system (sender, receiver) delays typical maximum tolerable delay: 400 ms
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.
A rule of thumb: When the packet loss rate exceeds
20%, the audio quality of VoIP is degraded beyond Quality of Service usefulness (RFC3714)
36
Transcoding
Convert a voice signal from analog to
digital or digital to analog Calls may experience multiple transcoding when routed with multiple voice coders Result in quality degradation
Quality of Service
37
Network Design to Support Voice

Qos / Cos
Path reservation RSVP Scheduling Priority Queuing, Weighted Fair Queuing, Class-based Queuing
Other elements
Virtual Private Networks Network Address Translation (NAT)
Quality of Service
38
Bandwidth Tradeoffs
Encode and compress
data for transport across the Internet Compression techniques developed for
Coding techniques for
Reduction in the required bandwidth Preserving voice quality
Comparison Standard Bandwidth (Compressed voice rate) Complexity (CPU usage) Voice quality Digitizing delay MOS, Mean Opinion Score
telephony and voice packet are standardized by the ITU-T in its Gseries recommendations
ITU-T Recommendation P.800 Excellent 5 Good 4 Fair 3 Poor 2 Bad 1 Toll quality, 4.0 or higher

Delay & loss in packet-switched networks Multimedia Networking Applications Real-time Multimedia: VoIP study QoS issues for VoIP

Audio compression standards Jitter mitigation Loss recovery Echo control Silence Suppression
Voice activity detection (VAD), comfort noise generation (CNG)
Beyond Best Effort

Scheduling and Policing Mechanisms Integrated Services and Differentiated Services RSVP
MPLS
Virtual LANs (802.1p/q)
Codec Selection
Depending on The bandwidth availability Acceptable voice quality
Standard G.711 G.729 G.723.1 Coding Type PCM CS-ACELP ACELP Bit Rate (Kbps) 64 8 6.3 Quality rate 4.3 4.0 3.8
MP-MLQ
5.3
Audio codecs
G.711 Raw telephone audio PCM, Pulse-Code Modulation Highest bandwidth consumption (64 kbps )
SPEEX

open-source (www.speex.org) excellent bandwidth (2 to 44 kbps) excellent quality high CPU usage
GSM G.729 cell phone audio compression low bandwidth usage, low CPU usage, good quality very low bandwidth, low cpu usage broad support from handset manufacturers medium quality Sensitive to packet loss (resilient to bit errors instead of packet loss)
iLBC
Speech codec developed for robust voice communication over
IP (www.ilbcfreeware.org)) Designed for narrow band speech, with a sampling rate of 8kHz Treat each packet independently from all other packets, ideal for packet communications Graceful speech quality degradation with increasing severity of IP packet loss and /or delay Bitrate 13.33 kbps (399 bits, packetized in 50 bytes) for the frame size of 30 ms and 15.2 kbps (303 bits, packetized in 38 bytes) for the frame size of 20 ms Basic quality higher then G.729A, high robustness to packet loss Computational complexity in a range of G.729A Royalty Free Codec
iLBC MOS behavior with packet loss
Quality of Service
44


Beyond Best Effort

MPLS
Delay Jitter
constant bit rate transmission client reception
variable network delay (jitter)
constant bit rate playout at client
buffered data
client playout delay
time
Consider the end-to-end delays of two consecutive
packets: difference can be more or less than 20 msec
Quality of Service
46
VoIP: 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
Quality of Service
47
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
packets generated packets received
loss
playout schedule p' - r playout schedule p-r
time
r p p'
Quality of Service
48
Adaptive Playout Delay, I

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.
t i timestamp of the ith packet ri the time packet i is received by receiver p i the time packet i is played at receiver ri t i network delay for ith packet d i estimate of average network delay after receiving ith packet
Dynamic estimate of average delay at receiver:
d i (1 u )d i 1 u( ri ti )
where u is a fixed constant (e.g., u = .01).
Adaptive playout delay II

Also useful to estimate the average deviation of the delay, vi :
vi (1 u )vi 1 u | ri ti d i |
The estimates di and vi are calculated for every received packet, although they are only used at the beginning of a talk spurt. For first packet in talk spurt, playout time is:
pi ti d i Kvi
where K is a positive constant. Remaining packets in talkspurt are played out periodically
Quality of Service
50
Adaptive Playout, III

Q: How does receiver determine whether packet is first in a talkspurt? 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.
Quality of Service
51
Delay & loss in packet-switched networks Multimedia Networking Applications Real-time Multimedia: VoIP study
QoS issues for VoIP

Jitter mitigation Loss recovery Echo control Voice activity detection (VAD), comfort noise generation (CNG)
Beyond Best Effort Scheduling and Policing Mechanisms Integrated Services and Differentiated Services RSVP MPLS Virtual LANs (802.1p/q)
Recovery from packet loss (1)

forward error correction (FEC): simple scheme for every group of n chunks create a redundant chunk by exclusive OR-ing the n original chunks send out n+1 chunks, increasing the bandwidth by factor 1/n. can reconstruct the original n chunks if there is at most one lost chunk from the n+1 chunks
Playout delay needs to
be fixed to the 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 resolution audio stream as the redundant information for example, nominal stream PCM at 64 kbps and redundant stream GSM at 13 kbps.
Whenever there is non-consecutive loss, the
receiver can conceal the loss. Can also append (n-1)st and (n-2)nd low-bit rate chunk
Interleaving chunks are broken up into smaller units for example, 4 5 msec units per chunk Packet contains small units from different chunks
if packet is lost, still have
most of every chunk has no redundancy overhead but adds to playout delay
Quality of Service
55
Loss concealment
There are various techniques for loss concealment
(i.e., hiding losses), such as those used in the Robust Audio Tool (RAT):
UCLs Robust Audio Tool (RAT) page: http://wwwmice.cs.ucl.ac.uk/multimedia/software/rat/
Quality of Service
56
A brief reflection: bag of tricks

use UDP to avoid TCP congestion control (delays)
for time-sensitive traffic
client-side adaptive playout delay: to compensate
for delay 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, interleaving retransmissions, time permitting conceal errors: repeat nearby data


Beyond Best Effort

MPLS
Echo in VoIP
A common impairment of digital telephone connections is
Echo is when a delayed and distorted copy of a voice signal is
Echo.
reflected back to its sender interrupting communication. Only speaker hears, receiver does not. A beneficial echo is called Sidetone. It is an effect and applied technique allowing the speaker to hear their own voice in the handsets speaker, convincing themselves their voice is being heard. The benefits of Sidetone are felt when a speaker hears their own voice at a low volume within between 5 and 25 milliseconds (ms). Call quality begins to be is impaired when a speaker hears their speech repeated after approximately 30 ms and more. Types of Echo

Electrical Echo, also known as line, hybrid or network echo Acoustic Echo
Quality of Service
59
Electrical Echo
Electrical Echo arises due to impedance mismatches at hybrids in
Public Switched Telephone Network (PSTN). Near Echo for A:

Leakage at AH1 + Reflection at AH2. Leakage at BH2 + Reflection at BH1.
Far Echo for A:

Quality of Service
60
Acoustic Echo
Acoustic Echo occurs in both PSTN and digital networks. Acoustic echo results from the following sources:
Handset use and design (see the above figure): a loudspeaker and a microphone are placed such that the microphone picks up the signal radiated by the loudspeaker and its reflections at the borders of the enclosure. Voice encoding and decoding devices (codecs): an unsuppressed acoustic or electrical echo made worse by digital encoding.
Quality of Service
61
Echo suppressors

Eliminating Echo
An echo suppressor detects human speech coming from one end of a connection, and suppresses all signals going the other way. An echo suppressor is toggled by a voice recognition circuit. They can typically trip within 5 ms to block a reflected signal. This technique results in a half-duplex channel. This half duplex operation is not noticeable in voice communication, but can adversely effect data communications. Echo suppressors are found on the PSTN installed by Inter-Exchange Carriers (IXCs). An echo canceller is a computer-based device that samples a call and profiles it. An echo will violate the profile. When an echo is detected it simulates the echo, estimates its magnitude, and then subtracts it from the audio signal it is sampling. Echo cancellers are found in modern digital networks at their junction with other networks.
Echo cancellers
Quality of Service
62
Echo cancellers
Have varying amounts of memory Compare received voice with current
patterns, cancel if match Fail if delay is larger than that the echo canceller memory can afford, which is called echo trail Built into the low bit-rate CODECs and are operated on DSP
Quality of Service
63
Echo Canceller Evolution
Echo cancellers are the superior solution for controlling echo in
networks. They are less expensive than the analog technology of echo suppressors, and they eliminate in-band signaling requirements. They are also digital. Echo cancellers are typically based on off-the-self Digital Signal Processors (DSPs) that are inexpensive and easy to program.
Echo Canceller in Packet Voice Gateway: Key Requirements

Cost / channel

Power consumption / channel
DSP solutions: MHz; Memory; Device choice Custom processors: Die size; Technology choice Lower the better
Quality of Service
65
ITU-T Standards on Echo Control

G.164 Echo suppressors G.165 Echo cancellers
G.167 Acoustic echo controllers
G.168 Digital network echo cancellers
Line echo cancellers are more than just adaptive filters.
The G.168 specification does not specify any one design or even recommend a particular tail length. Compliance with G.168 test requirements is necessary but not sufficient.
Only performance requirements laid down by ITU.
Adaptation control, non-linear processor, comfort noise injection, V.25 tone detection, etc. are non-trivial in terms of design and /or resource requirements.
ITU strongly recommends additional objective and subjective testing.



Beyond Best Effort

MPLS
Speech Characteristics
Conversational speech is a sequence of contiguous segments of
silence and speech. VAD algorithms take recourse to some form of speech pattern classification to differentiate between voice and silence periods.
Silence Suppression
Packets of silence are suppressed to save
bandwidth Voice Activity Detection (VAD)
Monitor the received signal for voice activity Comfort Noise Generation (CNG).
locally generated white noise make call appear normally connected to both parties when no packets received in silence period
Quality of Service
69
Speech Communication System with DTX

Voice Activity Detector (VAD) - classify speech frame as
active or inactive Silence suppression software monitors the received signal for voice activity. Comfort Noise Generation (CNG) - analysis of background noise parameters (encoder) and synthesis of comfort noise (decoder) When no activity is detected for the configured period of time the software informs the Packet Voice Protocol. The encoder output is stopped for bandwidth savings Relay this information to the remote end for comfort noise generation Discontinuous Transmission (DTX) - update background noise parameters Some VADs can cause voice clipping and can result in poor voice quality.
Speech Encoder
Speech Payload Speech Decoder
CNG Encoder Input CN Payload No Tx DTX Algorithm
Communication Channel CNG Decoder
Output
VAD Algorithm
Encoder
Decoder
Speech Communication System with DTX

Desirable aspects of VAD algorithms

A Good Decision Rule: A physical property of speech
that can be exploited to give consistent judgment in classifying segments of the signal into silent or voiced segments. Adaptability to Changing Background Noise: Adapting to non-stationary background noise improves robustness, especially in wireless telephony where the user is mobile. Low Computational Complexity: Internet telephony is a real-time application. Therefore the complexity of VAD algorithm must be low to suit real-time applications (not more than one packet time). Toll quality voice reproduction. Saving in bandwidth to be maximized.
VAD Standards
The latest ITU-T VAD standard is Rec. G.729
Annex B, developed for fixed telephony and multimedia communications. More recently the ETSI has standardized two VADs (options 1 and 2) for the adaptive multirate (AMR) codec developed for thirdgeneration mobile communication systems.
Quality of Service
73
ITU-T G.711 Appendix II

Comfort noise payload definition
approved at ITU-T February 2000 SG16

tested (MOS) using G.729B VAD/DTX and
example algorithm (described in Appendix) equivalent to G.711 without comfort noise flexible for use with any codec guidelines for use in Appendix
Quality of Service
74
QoS issues for VoIP

Quality of Service
75
Improving QOS in IP Networks

Thus far: making the best of best effort Future: next generation Internet with QoS guarantees RSVP: signaling for resource reservations Differentiated Services: differential guarantees Integrated Services: firm guarantees simple model for sharing and congestion studies:
Quality of Service
76
Principles for QOS Guarantees

Example: 1MbpsI P phone, FTP share 1.5 Mbps link. bursts of FTP can congest router, cause audio loss want to give priority to audio over FTP
Principle 1 packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly
Principles for QOS Guarantees (more)

what if applications misbehave (audio sends higher
than declared rate)
policing: force source adherence to bandwidth allocations
marking and policing at network edge: similar to ATM UNI (User Network Interface)
Principle 2 provide protection (isolation) for one class from others


Allocating
fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesnt use
its allocation
Principle 3 While providing isolation, it is desirable to use resources as efficiently as possible

Basic fact of life: can not support traffic demands

beyond link capacity
Principle 4 Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs
Summary of QoS Principles
Lets next look at mechanisms for achieving this .

Evolution of IP-Services
Initially All users were treated equally: no privileges Eventually Different types of usage need different treatment inside the network Evolving concepts (IntServ, DiffServ) Priority by type of services Priority by reservation
Per flow (RSVP) Per aggregate of flows (DiffServ)
Quality of Service
82
IP Service Classes
Best Effort No Signaling
Assured By Type of Service Prioritized By Type of Service
Guaranteed Per Flow Reservation
Assured By Aggregate Reservation
Pure packet-switching Most scalable
Circuit-switching Least scalable
Quality of Service
83
QoS management mechanisms of VoIP

Resource Provisioning Traffic Engineering Admission Control Control Plane
Connection Management Resource Reservation
Buffer Management/ Scheduling Packet Classification
Shaping
Transport Plane Policing
Loss Recovery/ Error Concealment
Quality of Service
84
QoS issues for VoIP

Quality of Service
85
Scheduling And Policing Mechanisms

scheduling: choose next packet to send on link FIFO (first in first out) scheduling: send in order of
arrival to queue

real-world example? discard policy: if packet arrives to full queue: who to discard? Tail drop: drop arriving packet priority: drop/remove on priority basis random: drop/remove randomly
Quality of Service
86
Scheduling Policies: more

Priority scheduling: transmit highest priority queued packet multiple classes, with different priorities
class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc.. Real world example?
Quality of Service
87
Scheduling Policies: still more

round robin scheduling: multiple classes cyclically scan class queues, serving one from each class (if available) real world example?
Quality of Service
88
Scheduling Policies: still more

Weighted Fair Queuing: generalized Round Robin each class gets weighted amount of service in each cycle real-world example?
Quality of Service
89
Policing Mechanisms
Goal: limit traffic to not exceed declared parameters
Three common-used criteria:
(Long term) Average Rate: how many pkts can be sent

per unit time (in the long run)
Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 15000
ppm peak rate
crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average!
(Max.) Burst Size: max. number of pkts sent

consecutively (with no intervening idle)
Quality of Service
90
Policing Mechanisms
Token Bucket: limit input to specified Burst Size
and Average Rate.
bucket can hold b tokens tokens generated at rate
full
r token/sec unless bucket
over interval of length t: number of packets admitted less than or equal to (r t + b).
Policing Mechanisms (more)

token bucket, WFQ combine to provide guaranteed
upper bound on delay, i.e., QoS guarantee!

token rate, r
bucket size, b
arriving
traffic
WFQ
per-flow rate, R
D = b/R max
Quality of Service
92

Delay & loss in packet-switched networks Multimedia Networking Applications Real-time Multimedia: VoIP study QoS issues for VoIP Beyond Best Effort
Scheduling and Policing Mechanisms

Integrated Services and Differentiated Services RSVP
MPLS
Quality of Service
93
IETF Integrated Services

architecture for providing QOS guarantees in IP
networks for individual application sessions resource reservation: routers maintain state info (a la VC) of allocated resources, QoS reqs admit/deny new call setup requests:
Question: can newly arriving flow be admitted with performance guarantees while not violated QoS guarantees made to already admitted flows?
Quality of Service
94
Intserv: QoS guarantee scenario

Resource reservation call setup, signaling (RSVP) traffic, QoS declaration per-element admission control
request/ reply
QoS-sensitive scheduling (e.g., WFQ)

Call Admission
Arriving session must :
declare its QOS requirement
defines the QOS being requested characterize traffic it will send into network T-spec: defines traffic characteristics signaling protocol: needed to carry R-spec and Tspec to routers (where reservation is required) RSVP
R-spec:
Quality of Service
96
Intserv QoS: Service models [rfc2211, rfc 2212]

Guaranteed service:
worst case traffic arrival:
Controlled load service:

"a quality of service closely
leaky-bucket-policed source simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988]
arriving traffic token rate, r bucket size, b
approximating the QoS that same flow would receive from an unloaded network element."
WFQ
per-flow rate, R
D = b/R max
IETF Differentiated Services

Concerns with Intserv: Scalability: signaling, maintaining per-flow router state difficult with large number of flows Flexible Service Models: Intserv has only two classes. Also want qualitative service classes

behaves like a wire relative service distinction: Platinum, Gold, Silver
Diffserv approach: simple functions in network core, relatively complex functions at edge routers (or hosts) Dot define define service classes, provide functional components to build service classes
Diffserv Architecture
Edge router:
per-flow traffic management marks packets as in-profile
r marking b
and out-profile
Core router:
per class traffic management buffering and scheduling based
on marking at edge preference given to in-profile packets Assured Forwarding
Quality of Service
99
Edge-router Packet Marking

profile: pre-negotiated rate A, bucket size B packet marking at edge based on per-flow profile
Rate A
B
User packets
Possible usage of marking:

class-based marking: packets of different classes marked
differently intra-class marking: conforming portion of flow marked differently than non-conforming one
Quality of Service
100
Classification and Conditioning

Packet is marked in the Type of Service (TOS) in
IPv4, and Traffic Class in IPv6 6 bits used for Differentiated Service Code Point (DSCP) and determine PHB that the packet will receive 2 bits are currently unused
Quality of Service
101
Classification and Conditioning

may be desirable to limit traffic injection rate of some class: user declares traffic profile (e.g., rate, burst size) traffic metered, shaped if non-conforming
Quality of Service
102
Forwarding (PHB)
PHB result in a different observable (measurable)
forwarding performance behavior PHB does not specify what mechanisms to use to ensure required PHB performance behavior Examples:

Class A gets x% of outgoing link bandwidth over time intervals of a specified length Class A packets leave first before packets from class B
Quality of Service
103
Forwarding (PHB)
PHBs being developed:
Expedited Forwarding: pkt departure rate of a
class equals or exceeds specified rate
logical link with a minimum guaranteed rate
Assured Forwarding: 4 classes of traffic each guaranteed minimum amount of bandwidth each with three drop preference partitions
Quality of Service
104
QoS issues for VoIP

Quality of Service
105
Signaling in the Internet

connectionless (stateless) forwarding by IP routers
best effort service
no network signaling protocols in initial IP design
New requirement: reserve resources along end-to-end
path (end system, routers) for QoS for multimedia applications RSVP: Resource Reservation Protocol [RFC 2205]
allow users to communicate requirements to network in robust and efficient way. i.e., signaling !
earlier Internet Signaling protocol: ST-II [RFC 1819]

RSVP Design Goals

1.
2. 3. 4.
5.
6.
accommodate heterogeneous receivers (different bandwidth along paths) accommodate different applications with different resource requirements make multicast a first class service, with adaptation to multicast group membership leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes control protocol overhead to grow (at worst) linear in # receivers modular design for heterogeneous underlying technologies
RSVP: does not

specify how resources are to be reserved
rather: a mechanism for communicating needs thats the job of routing protocols signaling decoupled from routing separation of control (signaling) and data (forwarding) planes
determine routes packets will take

interact with forwarding of packets
Quality of Service
108
RSVP: overview of operation

senders, receiver join a multicast group done outside of RSVP senders need not join group sender-to-network signaling path message: make sender presence known to routers path teardown: delete senders path state from routers receiver-to-network signaling reservation message: reserve resources from sender(s) to receiver reservation teardown: remove receiver reservations network-to-end-system signaling path error reservation error
Path msgs: RSVP sender-to-network signaling

path message contents:
unicast destination, or multicast group flowspec: bandwidth requirements spec. filter flag: if yes, record identities of upstream senders (to allow packets filtering by source) previous hop: upstream router/host ID refresh time: time until this info times out path message: communicates sender info, and reversepath-to-sender routing info later upstream forwarding of receiver reservations
address:
RSVP: simple audio conference

H1, H2, H3, H4, H5 both senders and receivers multicast group m1 no filtering: packets from any sender forwarded audio rate:
only one multicast routing tree possible

H2 R1 H1 H5
H3 R2 R3
H4
RSVP: building up path state

H1, , H5 all send path messages on
(address=m1, Tspec=b, filter-spec=no-filter,refresh=100)
m1:
Suppose H1 sends first path message

m1: in L1 out L2 L6 L6 m1: in out L5 L7 m1: in L7 out L3 L4
H2
L2
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5

next, H5 sends path message, creating more state
in routers
L6 L1 m1: in out L1 L2 L6 L5 L6 m1: in out L5 L6 L7
m1:
in L7 out L3 L4
H2
L2
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5

H2, H3, H5 send path msgs, completing path state
tables
L1 L2 L6 m1: in out L1 L2 L6 L5 L6 L7 m1: in out L5 L6 L7
m1:
in L3 L4 L7 out L3 L4 L7
H2
L2
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5
reservation msgs: receiver-to-network signaling

reservation message contents: desired bandwidth: filter type: no filter: any packets address to multicast group can use reservation fixed filter: only packets from specific set of senders can use reservation dynamic filter: senders whos p[ackets can be forwarded across link will change (by receiver choce) over time. filter spec reservations flow upstream from receiver-to-senders,
reserving resources, creating additional, receiverrelated state at routers
Quality of Service
115
RSVP: receiver reservation example 1

H1 wants to receive audio from all other senders H1 reservation msg flows uptree to sources H1 only reserves enough bandwidth for 1 audio stream reservation is of type no filter any sender can use reserved bandwidth
H2
L2
H3
L3 L1
R1
L6
R2
L5
L7
R3
L4
H4
H1
H5
RSVP: receiver reservation example 1

H1 reservation msgs flows uptree to sources routers, hosts reserve bandwidth b needed on
downstream links towards H1

L6 L6 m1: in L5 out L5
m1: in L1 L2 out L1(b) L2
m1: L7 L6 L6(b) L7
in L3 out L3
L4 L4
L7 L7(b)
H2
L2
b b L1
R1
b L6
H1
R2
L5
b L7
R3
L3 b L4
H3
H4
H5
RSVP: receiver reservation example 1 (more)

next, H2 makes no-filter reservation for bandwidth H2 forwards to R1, R1 forwards to H1 and R2 (?) R2 takes no action, since
L6 m1: in L1 L2 out L1(b) L2(b) L6 m1: in L5 out L5 L7 L6 L6(b) L7 L3 b L4 b
b already reserved on L6
m1: in L3 out L3 L4 L4 L7 L7(b)
H2
b L2
b b b L1
H3
R1
b L6
H1
R2
L5
b L7
R3
H4
H5
RSVP: receiver reservation: issues

What if multiple senders (e.g., H3, H4, H5) over link (e.g., L6)? arbitrary interleaving of packets L6 flow policed by leaky bucket: if H3+H4+H5 sending rate exceeds b, packet loss will occur
L6 m1: in L1 L2 out L1(b) L2(b) L6 m1: in L5 out L5 L7 L6 L6(b) L7 L3 b L4 b m1: in L3 out L3 L4 L4 L7 L7(b)
H2
b L2
b b b L1
H3
R1
b L6
H1
R2
L5
b L7
R3
H4
H5
RSVP: example 2
H1, H4 are only senders send path messages as before, indicating filtered reservation Routers store upstream senders for each upstream link H2 will want to receive from H4 (only)
H2
L2
H3
L3 L1
R1
L6
R2
L7
R3
L4
H4
H1
Quality of Service
120
RSVP: example 2
H1, H4 are only senders send path messages as before, indicating filtered reservation
in
L1, L6 L2(H1-via-H1 out L6(H1-via-H1 L1(H4-via-R2
in ; H4-via-R2 ) ) )
L4, L7 ; H1-via-R3 ) ) )
L3(H4-via-H4 out L4(H1-via-R2 L7(H4-via-H4
H2
L2
H3 R2
L1 L3 L7 L6, L7 ) )
R1
L6 in
R3
L4
H4
H1
L6(H4-via-R3 out L7(H1-via-R1
RSVP: example 2
receiver H2 sends reservation message for source H4
at bandwidth b
propagated upstream towards H4, reserving b

in ;H4-via-R2 (b)) ) ) L4, L7 )
in
L3(H4-via-H4 ; H1-via-R2 out L4(H1-via-R2 ) L7(H4-via-H4 (b))
H2
L2
b L1
H3 R1
b L6 in
R2
b L7
R3
L3 b L4
H4
H1
L6, L7
L6(H4-via-R3 (b)) out L7(H1-via-R1 )

RSVP: soft-state
senders periodically resend path msgs to refresh (maintain) state
receivers periodically resend resv msgs to refresh (maintain) state

path and resv msgs have TTL field, specifying refresh interval
L1, L6 L2(H1-via-H1 out L6(H1-via-H1 L1(H4-via-R2 in in ;H4-via-R2 (b)) ) ) L4, L7 )
L3(H4-via-H4 ; H1-via-R3 out L4(H1-via-62 ) L7(H4-via-H4 (b))
H2
L2
b L1
H3 R1
b L6 in
R2
b L7
R3
L3 b L4
H4
H1
L6, L7

RSVP: soft-state
suppose H4 (sender) leaves without performing teardown eventually state in routers will timeout and disappear!
in
in ;H4-via-R2 (b)) ) )
L4, L7 )
L3(H4-via-H4 ; H1-via-R3 out L4(H1-via-62 ) L7(H4-via-H4 (b))
H2
L2
b L1
H3 R1
b L6 in
R2
b L7
R3
L3 b L4
H1
gone H4 fishing!
L6, L7

The many uses of reservation/path refresh

recover from an earlier lost refresh message expected time until refresh received must be longer than timeout interval! (short timer interval desired)
Handle receiver/sender that goes away without
teardown
Reservation refreshes will cause new reservations
Sender/receiver state will timeout and disappear
to be made to a receiver from a sender who has joined since receivers last reservation refresh

E.g., in previous example, H1 is only receiver, H3 only sender. Path/reservation messages complete, data flows H4 joins as sender, nothing happens until H3 refreshes reservation, causing R3 to forward reservation to H4, which allocates bandwidth
RSVP: reflections
multicast as a first class service receiver-oriented reservations use of soft-state
Quality of Service
126

Multimedia Networking Applications Real-time Multimedia: VoIP study QoS issues for VoIP Beyond Best Effort Scheduling and Policing Mechanisms Integrated Services and Differentiated Services RSVP MPLS Virtual LANs (802.1p/q)
Quality of Service
127
Virtualization of networks
Virtualization of resources: a powerful abstraction in systems engineering: computing examples: virtual memory, virtual devices Virtual machines: e.g., java IBM VM os from 1960s/70s layering of abstractions: dont sweat the details of the lower layer, only deal with lower layers abstractly
Quality of Service
128
The Internet: virtualizing networks

1974: multiple unconnected nets
ARPAnet data-over-cable networks packet satellite network (Aloha) packet radio network
differing in:
addressing conventions packet formats error recovery routing
ARPAnet
"A Protocol for Packet Network Intercommunication", V. Cerf, R. Kahn, IEEE Transactions on Communications, May, 1974, pp. 637-648.
satellite net
The Internet: virtualizing networks

Gateway: Internetwork layer (IP): addressing: internetwork appears as embed internetwork packets in local packet format or extract a single, uniform entity, despite them underlying local network route (at internetwork level) to heterogeneity next gateway network of networks
gateway
ARPAnet
satellite net
Cerf & Kahns Internetwork Architecture What is virtualized?

two layers of addressing: internetwork and local network new layer (IP) makes everything homogeneous at
internetwork layer underlying local network technology cable satellite 56K telephone modem today: ATM, MPLS invisible at internetwork layer. Looks like a link layer technology to IP!
Quality of Service
131
Multiprotocol label switching (MPLS)

viewed by Internet as logical link connecting IP routers
just like dialup link is really part of separate network (telephone network)
initial goal: speed up IP forwarding by using fixed length
label (instead of IP address) to do forwarding

borrowing ideas from Virtual Circuit (VC) approach but IP datagram still keeps IP address!
PPP or Ethernet header
MPLS header
IP header
remainder of link-layer frame
label
20
Exp S TTL
3 1 5
MPLS capable routers

a.k.a. label-switched router
forwards packets to outgoing interface based only on
label value (dont inspect IP address)
signaling protocol needed to set up forwarding RSVP-TE forwarding possible along paths that IP alone would not allow (e.g., source-specific routing) !! use MPLS for traffic engineering must co-exist with IP-only routers
MPLS forwarding table distinct from IP forwarding tables
Quality of Service
133
MPLS forwarding tables

in label out label dest out interface
10 12 8
A D A
0 0 1
in label
out label dest
out interface
10 12
6 9
A D
1 0
R6
0
1 0 1
R4 R5
R3
0
0 in label outR1 label dest
A
out interface
R2
in label out label dest out interface
0

Delay & loss in packet-switched networks Multimedia Networking Applications
Real-time Multimedia: VoIP study

QoS issues for VoIP Beyond Best Effort Scheduling and Policing Mechanisms Integrated Services and Differentiated Services RSVP MPLS Virtual LANs (802.1p/q)
Quality of Service
135
Simple Ethernet Switch Topology

mail server router switch web server
to external network
IP subnet
switch
switch
switch
Quality of Service
136
Rational for Virtual LANs

Groups of users within an organization are typically separated into
their own IP networks (typically subnets) for network management, performance, security and other policy reasons. Users on LANs should be grouped by their community of interest (sales dept., engineering, accounting), not by their location in the building. However, users within a single community of interest are rarely located in the same part of a building. Ethernet switches are easy, routers are hard. Given all the above, how can we separate users via switches? What are the benefits to users and network administrators?
Quality of Service
137
Ethernet Switch and VLAN Topology

mail server router switch web server
to external network
IP subnet
switch switch
switch
Quality of Service
138
Virtual LANs
Switches are easy, routers are hard. Given
this, how can we separate users via switches? What are the benefits? Virtual LANs provide separate collision and broadcast domains for groups of users. Users are assigned to one or more VLANs automatically or via a management system. VLANs can span multiple switches and sites How do users on different VLANs talk to each other?
Virtual LANs
VLANs are LAN segments (in the classic sense of the
word) that can span multiple ethernet switches. VLANs provide separate collision and broadcast domains for each group of users. Users are assigned to one or more VLANs automatically or via a management system. Potential advantages of VLANs include:
Better isolation between groups of users: however it is incorrect to think that VLANs significantly improve network security. Improved performance: the specific LAN performance requirements of each group can be met more easily. Improved performance: VLANs provide multiple broadcast domains Provides for more sophisticated network administration
IEEE 802.1p
Supplement to MAC Bridges: Traffic Class
Expediting and Dynamic Multicast Filtering, IEEE P802.1p/D6. Extended encapsulation (802.1Q). Method to define relative priority of frames (user_priority). IEEE 802.1p support in LAN switches would provide transmission servicing based on relative priority indicated in each frame (delay indication).
Quality of Service
141
Tagging 802.1Q
6 bytes 6 bytes 4 bytes 2 bytes Up to 1500 bytes 4 bytes
Destination address
Source address
802.1Q Tag
Type field
Data field
CRC
Ethernet V2.0 With 802.1Q Tag
6 bytes
6 bytes
4 bytes
2 bytes
Up to 1496 bytes
4 bytes
Destination address
Source address
802.1Q Tag
Length field
Data field
CRC
IEEE 802.3 With 802.1Q Tag
Octet 1
Octet 2 1 2 3 4
Octet 3 5 6 C FI 7 8
Octet 4
Ethernet-encoded TPID 0x8100
802.1p 3 bits Tag Protocol Identifier (TPID)
VLAN ID 12 bits Tag Control Information (TCI) Format
Configurable .1Q Ethertype Default = 8100

Virtual LANs: 802.1p and 802.1q

Ethernet priority and VLANs are distinct concepts,
but they are intertwined by the technology. VLANs are identified by a 12 bit VLAN Identifier. Frame priority is marked by a 3 bit field, 0 to 7. This is known as Class of Service. Switches can and do, write or re-write, the priority field based on:

Port on the frame was received MAC address of the sending station Protocol IP, IPX, etc. IP Precedence field or DSCP Other IP and/or TCP information Combination of the above
802.1p delivers CoS, not QoS
Quality of Service
144
Quality of Service: Summary

multimedia applications and requirements making the best of todays best effort
service scheduling and policing mechanisms next generation Internet: Intserv, RSVP, Diffserv, MPLS, 802.1p/q
Quality of Service
145
Reference
[1] ITU-T Recommendation G.729 Annex B: a silence compression scheme for use with G.729 optimized for V.70 digital simultaneous voice and data applications, Benyassine, A.; Shlomot, E.; Su, H.-Y.; Massaloux, D.; Lamblin, C.; Petit, J.-P.; Communications Magazine, IEEE Volume 35, Issue 9, Sept. 1997 Page(s):64 73 [2] Algorithmic and implementation aspects of echo cancellation in packet voice networks, Krishna, V.V.; Rayala, J.; Slade, B.; Signals, Systems and Computers, 2002. Conference Record of the Thirty-Sixth Asilomar Conference on Volume 2, 3-6 Nov. 2002 Page(s):1252 - 1257 vol.2 [3] Empirix, Inc., Hammer VoIP test system echo detection and analysis, 2001. [Online]. Available: http://wireless.feld.cvut.cz/mesaqin2002/Echo_Detection.pdf [4] GQ Maguire Jr., 2G1325/2G5564 Practical Voice Over IP (VoIP): SIP and related protocols, http://www.it.kth.se/courses/2G1325/VoIP-CoursepageSpring-2005.html, Spring 2005. S. Floyd and J. Kempf (Editors), IAB Concerns Regarding
Congestion Control for Voice Traffic in the Internet, IETF, RFC 3714, Network Working Group, March 2004. ftp://ftp.rfceditor.org/in-notes/rfc3714.txt
Quality of Service
146

Quality of Service For Voip

Hochgeladen von

Dokumentinformationen

Originaltitel

Copyright

Verfügbare Formate

Dieses Dokument teilen

Dokument teilen oder einbetten

Freigabeoptionen

Stufen Sie dieses Dokument als nützlich ein?

Sind diese Inhalte unangemessen?

Copyright:

Verfügbare Formate

Quality of Service For Voip

Hochgeladen von

Copyright:

Verfügbare Formate

Quality of Service for VoIP

Multimedia, Quality of Service: What is it?

performance needed for application to function.

Quality of Service: Goals

Architectures for QoS TOS, Diffserv, Intserv, RSVP, MPLS, 802.1p/q

Voice activity detection (VAD) Comfort noise generation (CNG)

Quality of Service: Outline

Multimedia Networking Applications

How do loss and delay occur?

packet being transmitted (delay)

Four sources of packet delay

check bit errors determine output link

Delay in packet-switched networks

Note: s and R are very different quantities!

typically a few microsecs or less

dqueue = queuing delay

dtrans = transmission delay

dprop = propagation delay

a few microsecs to hundreds of msecs

Queueing delay (revisited)

traffic intensity = La/R

average delay infinite!

Real Internet delays and routes

Real Internet delays and routes

Quality of Service: Outline

Multimedia Networking Applications

end-to-end delay delay jitter

But loss tolerant:

Streaming Stored Multimedia

transmitted data: in time for playout

Streaming Stored Multimedia: What is it?

3. video received, played out at client time

streaming: at this time, client

Streaming Stored Multimedia: Interactivity

VCR-like functionality: client can

Streaming Live Multimedia

Interactive, Real-Time Multimedia

Multimedia Over Todays Internet

no guarantees on delay, loss

How should the Internet evolve to better support multimedia?

Whats your opinion?

A few words about audio compression

telephone: 8,000 samples/sec CD music: 44,100 samples/sec

Each sample quantized,

some quality reduction

e.g., 28=256 possible quantized values

Each quantized value

8 bits for 256 values

A few words about video compression

images displayed at constant rate

Digital image is array of

pixels Each pixel represented by bits Redundancy

adapt layers to available bandwidth

Quality of Service: Outline

Multimedia Networking Applications

Real-time interactive applications

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

Dialpad Net2phone videoconference with Webcams

Interactive Multimedia: VoIP

64 kbps during talk spurt

pkts generated only during talk spurts

20 msec chunks at 8 Kbytes/sec: 160 bytes data

application sends UDP segment into socket every