cse 124 networked services fall 2009
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CSE 124 Networked Services
Fall 2009B. S. Manoj, Ph.D
http://cseweb.ucsd.edu/classes/fa09/cse124
10/8/2009 1CSE 124 Networked Services Fall 2009
Some of these slides are adapted from various sources/individuals including but not limited to the slides from the text books by Kurose and Ross. Use of these slides other than for pedagogical purpose for CSE 124, may require explicit permissions from the respective sources.
Multimedia Networking Applications • Network applications can be broadly classified into
– Loss sensitive• Data traffic such as HTTP or FTP traffic• Delay tolerant
– Delay sensitive• Streamed stored audio/video• Streamed live audio/video• Interactive video• Loss tolerant
– Loss and delay sensitive• Time-sensitive stock quotes• Health sensor traffic
10/8/2009 2CSE 124 Networked Services Fall 2009
Streaming Stored Audio/Video• Streaming
– The media transfer scheme where a part of the media file is played out while the remaining parts of the file are being received
– Popular services: stored video sharing servers such s YouTube, Yahoo Videos, CNN etc.
– Uni-directional media communication
• Main features– Stored media files that are pre-recorded and coded– Streaming over the Internet
• Streaming server pushes the content at a regular rate• Streaming client begins play back a few seconds after beginning reception• Two kinds of media players
– Web browser-based and Host based
– Continuous play out• Play out options are many: Fast Forward, Rewind, and Pause• Once play out begins, it should strive to maintain the original recorded timings• Key issue: getting the data over the network in time10/8/2009 3CSE 124 Networked Services Fall 2009
Streaming Live Audio and Video• Media source generates multimedia content in real-
time– e.g., live video or audio transmission– Delay associated with content generation
• Limited play out options: Limited Rewind and Pause
• Uni-directional media communication
• More stringent delay constraints than stored media streaming
10/8/2009 4CSE 124 Networked Services Fall 2009
Real-time Interactive Audio/Video• Mostly bi-directional media communication
• Each end-source generates media content in real-time
• High delay constraints due to interactive nature of communication
• End-to-end delay preferably < 150ms
• e.g, Voice over IP applications such as Skype, Google Talk, Yahoo Messenger, Microsoft Netmeeting
10/8/2009 5CSE 124 Networked Services Fall 2009
Why multimedia services are challenging?
• Internet is designed for delay tolerant data communications– Best-effort traffic support only– Neither guarantee nor timeliness of data delivery
• During high load situations – the delay performance can be worse– High load can be at the server, network links, or the routers
• Main issues– Delay (latency or end-to-end delay)– Jitter (Delay jitter or Delay variation)– Packet loss
10/8/2009 6CSE 124 Networked Services Fall 2009
Delay • Kinds of delays
– Source delay (content generation delay– End-to-end delay – Play out delay
• Source delay– Generating a media content takes certain amount of time
10/8/2009 7CSE 124 Networked Services Fall 2009
Analog voice (4KHz)
Digitization (8KHz, 8 bits per sample)
10101010000000….
8 KBytes per second
10101010000
160 Bytes packet will take about 160 B/8KB/s= 20ms120 B/8KB/s= 15ms
Delay (contd)
• End-to-end delay – Due to the end-to-end network
• Contributed by– Processing time by the intermediate routers– Queuing delay at intermediate routers– Transmission delay due to the source and
intermediate routers– Propagation delay due to the links in the network
10/8/2009 8CSE 124 Networked Services Fall 2009
Introduction 1-9
End-to-end delay: four sources• 1. Router processing:
– Receive– Check bit errors– Buffer– Determine output link
A
B
propagation
transmission
Routerprocessing queueing
2. Queueing Time waiting at
output link/buffer for transmission
Vary drastically depends on congestion level of router
Introduction 1-10
End-to-end delay
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 – copper: ~2x108 m/sec– Wireless: 3x 108 m/sec– Fiber: 3x 108 m/sec
• propagation delay = d/s
A
B
propagation
transmission
Routerprocessing queueing
Introduction 1-11
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!
End-to-end Delay
• Delay at a router/node
• End-to-end delay
10/8/2009 CSE 124 Networked Services Fall 2009 12
proptransqueueprocrouter ddddd
• dproc = processing delay– typically a few microsecs or
less• dqueue = queuing delay
– depends on congestion• dtrans = transmission delay
– = L/R, significant for low-speed links
• dprop = propagation delay– a few microsecs to hundreds
of msecs– N = number of routers/nodes in
the network– di = delay at router/node i
N
iid
1
Playout delay• The delay added/caused by the receiver-side media
player
• A certain amount of delay in playing out may improve the playout performance
• Challenge is to get the required OS resources to play when desired– High priority for playout processes is essential
• Two types– Fixed playout delay– Adaptive playout delay10/8/2009 13CSE 124 Networked Services Fall 2009
Jitter• The shared network resources such as links and routers
– Results in high variability in end-to-end delay– Sometimes packets can be even out-of-ordered
• Jitter cannot be easily removed– Because the network is best-effort– Its impact can be lessened– Receiver playout management
t
1 2
1t+d
t+20ms
t+20ms+2d
2
3t+40ms
3t+40ms+d
10/8/2009 14CSE 124 Networked Services Fall 2009
Handling Jitter• The impacts of Jitter can be managed together by
– Sequence numbering– Time stamps– Receiver playout delay
• Media source adds sequence numbers to every media packet– Sequence number increments with every packet– Usually unique for a certain duration of the session
• Time stamps include the time instance at which the packets are generated
• Sequence numbers and time stamps help– differentiate packet losses from silence periods10/8/2009 15CSE 124 Networked Services Fall 2009
t
1 2
1t+d
t+20ms
t+20ms+d
2
3t+40ms
3
4 5 6
4 5 6
t+80ms t+100ms t+120ms
t+40ms+d t+80ms+d t+100ms+d t+120ms+d
t
1 2
1t+d
t+20ms
t+20ms+d
2
3t+40ms
3
4 5 6
6
t+60ms t+80ms t+100ms
t+40ms+d t+100ms+d
t
1 2
1t+d
t+20ms
t+20ms+d
2
3t+40ms
3
4 5 6
4 5 6
t+60ms t+80ms t+100ms
t+40ms+d t+60ms+d t+80ms+d t+100ms+d
PacketLoss
Talk spurt
10/8/2009 16CSE 124 Networked Services Fall 2009
Receiver Playout delay• Delay added by receiver media player for every packet
• Two approaches– Fixed playout delay– Adaptive playout delay
• Fixed playout delay– Receiver fixes the playout delay for all packets– Simple to implement– e.g., media receiver plays out every packet exactly q units of
time after receiving it • If packet is received at time t, it is played at time t+q
– Determining a good value for q is a challenge10/8/2009 17CSE 124 Networked Services Fall 2009
CSE 124 Networked Services Fall 2009
Fixed 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’
10/8/2009
Determining Fixed Playout Delay
• There are no strict rules for the choice of fixed playout delay– The delay is sufficient to handle the Jitter
– One good estimate is the play out time can be equal to Mean Delay + Mean Jitter
– Therefore, p = (Mean Delay + Mean Jitter) – r
150 ms 400 ms0 ms
10/8/2009 19CSE 124 Networked Services Fall 2009
Adaptive Playout Delay• In a dynamic network, Jitter can vary highly
– Use of fixed playout delay can result in high packet loss or non optimal play out delay
– Adaptive Playout delay is preferred in such dynamic situations
– Adaptive playout delay, dynamically modifies the playout delay
– Playout delay is modified based on the delay and jitter observations
– Playout delay is estimated for every packet, however, modified only when the talk spurt begins
• Objective: Minimize playout delay, keeping late loss rate low10/8/2009 20CSE 124 Networked Services Fall 2009
Estimating Adaptive Playout Delay
packetith receivingafter delay network average of estimated
acketpith for delay network tr
receiverat played is ipacket timethep
receiverby received is ipacket timether
packetith theof timestampt
i
ii
i
i
i
dynamic estimate of average delay at receiver:
)()1( 1 iiii trudud
where u is a fixed constant (e.g., u = .01).
• One Approach to 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.
10/8/2009 21CSE 124 Networked Services Fall 2009
Estimating Adaptive playout delay also useful to estimate average deviation of delay, vi :
||)1( 1 iiiii dtruvuv
estimates di , vi calculated for every received packet (but used only at start of talk spurt
for first packet in talk spurt, playout time is:
iiii Kvdtp where K is positive constant
remaining packets in talkspurt are played out periodically at time
iijj Kvdtp
10/8/2009 22CSE 124 Networked Services Fall 2009
• TCP-like transport protocols are not suitable for multimedia traffic– They are connection oriented
• high overhead– They offer reliable delivery
• high delay due to potential retransmissions• Larger playout delay: smooth TCP delivery rate• HTTP/TCP passes more easily through firewalls• Transmission rate fluctuates due to TCP congestion control
• UDP-like light connection less protocols are preferred– Low end-to-end delay– short playout delay (2-5 seconds) to remove network jitter
• Due to administrative reasons, TCP still dominates the multimedia video/audio transport
• UDP is prominent for VoIP applications
Transport layer choice for multi-media applications
10/8/2009 23CSE 124 Networked Services Fall 2009
Packet loss
• Packet loss is unavoidable– Recovery from packet loss is an important
objective– Lossy recovery is sufficient for multimedia
• Two popular approaches– Forward Error Correction– Packet Interleaving
10/8/2009 CSE 124 Networked Services Fall 2009 24
Forward Error CorrectionApproach: Packet Redundancy • 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
FEC Approach: Stream redundancy “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
Packet Interleaving method
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
Week-2-Homework
• Reading assignments – File Transfer protocol
• End-of-chapter Problems P10 and P11 from Chapter 7 of Kurose and Ross (page 676)
– Will be placed at the course website
10/8/2009 28CSE 124 Networked Services Fall 2009
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