Multimedia

Networking 
Questions

Raj Jain
Washington University in Saint Louis
Saint Louis, MO 63130
[email protected]
Audio/Video recordings of this lecture are available on-line at:
http://www.cse.wustl.edu/~jain/cse473-20/
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 Overview Questions


 Multimedia Networking Applications
 Skype
 Real-Time Transport Protocol (RTP)
 Session Initiation Protocol (SIP)

Note: This class lecture is based on Chapter 9 of the textbook (Kurose

and Ross) and the figures provided by the authors.

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 Overview Multimedia Applications

1. Audio Digitization Questions

2. Playout Buffers

3. Streaming Using UDP
4. Streaming Using HTTP

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 Audio Digitization
Sampling: Analog audio signal sampled at constant rate
Questions
Telephone:
8,000 samples/sec quantization
error
quantized value
of  Does digitization
CD music: analog value
and compression
44,100 samples/sec

audio signal amplitude

analog of audio reduce its
Quantization: Each sample signal
quality?
8 bits: 28=256 values
16 bit: 216 values
8 k samples/s each 8 bit time

⇒ 64 kbps sampling rate

(N sample/sec)
Compression: Compress to 5-16 kbps, e.g., using differences
Lower bits/sample, samples/sec, or higher compression ⇒ Lower Quality
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 Multimedia Networking Applications

Streaming Stored Multimedia

Stored Media: Fast rewind, pause, fast forward
Questions
Streaming: simultaneous play out and download
Continuous play out: Delay jitter smoothed by playout buffer 
Streaming Live Multimedia: IPTV and Internet Radio
No fast-forward
High data rate to large number of users
⇒ multicast or P2P,
Delay jitter controlled by caching,
Real-Time Interactive Multimedia: Internet Telephone, Video
Conferencing
Delay<400 ms.

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 Playout Buffers
constant bit Questions
rate video client video constant bit
transmission reception rate video
playout at client 
variable
network

buffered
video
delay

client playout time

delay
Playout delay compensates for network delay, delay jitter
Delay > Playout Delay ⇒ Packet late ⇒ Same as a lost packet
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 Client-side Buffering
1. Initial fill of buffer until playout begins at tp
Questions
2. Fill rate x(t) varies and playout rate r is constant
3. x < r: Buffer eventually empties causing freezing of video  In Point 5, what
4. x > r: buffer will not empty, Flow control to avoid overflow does it mean by
"Larger delay until
5. Tradeoff: Large initial playout delay ⇒ Buffer starvation less
user begins
likely but larger delay until user begins watching
watching"?
buffer fill level,
Q(t)
variable fill playout rate,
rate, x(t) e.g., CBR r

client application
client
video server buffer, size B
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 Streaming Using UDP
Server sends at rate appropriate for client
Questions
Often: Send rate = Encoding rate = Constant
Transmission rate can be oblivious to congestion levels  Why might firewalls block
UDP streaming?
Short playout delay (2-5 seconds) to remove network jitter  Why cant UDP
Application level error recovery prefetch/buffer? In general
given that video streaming
UDP may not go through firewalls ~ 80% of internet traffic
why does it seem so
unoptimized?
 Why do firewalls block
UDP and not TCP? Do most
personal computers’
firewalls just leave that
port open or how would
the typical person’s firewall
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9-8
 Streaming Using HTTP
Multimedia file retrieved via HTTP GET
Questions
Send at maximum possible rate under TCP
Variable  Just to verify, if
rate, x(t) playout begins at
Video TCP send
t=8, for the very 1st
TCP receive Application
file buffer buffer playout buffer packet, is it arriving
Server Client in time? Or, does it
have to begin at t=9
Fill rate fluctuates due to TCP congestion control,
retransmissions (in-order delivery) (also for the very
1st packet)?
Larger playout delay to smooth TCP delivery rate
HTTP/TCP passes more easily through firewalls
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9-9
Review
Multimedia Applications

1. Audio is sampled, digitized, and compressed Questions

2. Initial playout delay helps overcome the jitter in  Does the HTTP
delay implementation of
3. UDP results in lower jitter but may not go through stored streaming
firewall always use DASH, or
are there other
4. HTTP uses TCP and so the delay variation can be implementations
large using HTTP/TCP as
well?

Ref: Section 9.1 and 9.2, Review Questions R1-R8, Problems P1-P5
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9-10
 Overview Voice Over IP

1. VoIP Packet Losses Questions

2. VoIP with Fixed Playout Delay
3. VoIP with Adaptive Playout Delay
4. Recovering From Packet Loss
5. Skype

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 Voice-over-IP (VoIP)
End-end-delay Requirement: needed to maintain
“conversational” aspect
Questions
Higher delays noticeable, impair interactivity  20ms at 8 Kbytes / sec
< 150 ms: good should be 163.84 bytes,
but the slide says it is 160
> 400 ms: bad bytes. Is it okay if we just
Includes application-level (packetization, playout), network calculate data size using
delays the lowercase k, not the
capital K?
Alternating talk spurts, silent periods.  Think you said that 160B
64 kbps during talk spurt of data is wrong (in the
2nd to last bullet point)
Packets generated only during talk spurts because K = 1024, not
20 ms chunks at 8 kbytes/sec: 160 bytes of data 1000. So is it 163.84B of
data? (8*1024*20/1000)
Application sends a segment every 20 ms during talk spurt
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9-12
 VoIP Packet Losses
Network Loss: IP datagram lost due to network congestion
(router buffer overflow)
Questions
Delay Loss: IP datagram arrives too late for playout  Loss Tolerance: Packet
typical maximum tolerable delay: 400 ms loss rates between 1%
and 10% can be
Loss Tolerance: Packet loss rates up to 10% can be concealed concealed. Just curious
about how the range is
determined, what if rate
is below 1%?
 For loss tolerance, what
exactly does 'concealed'
mean? Is it just no
sound or is some sound
transmitted to cover it
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up?
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 VoIP with Fixed Playout Delay
Example: Packets sent every 20 ms during talk spurt.
 First packet received at time r Questions
 If playout begins at p, 4th packet will arrive too late
 what if the green
 If playout begins at p’, all packets can be played on time
and red tip are on
Packets
the same
Sent Received Played position, is the
loss
packet loss? or
can it receive the
packet and start
playing at the
time
same time?
r
p p'
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 Homework 9: Playout Delay
8

7
Questions
6
Packets

Packets  For the homework do we

5
Generated Packets need to draw out and
4 Received
show the line related to
3
the playout delay on the
2 graph or are the answers
1 to the question sufficient
for the turnin?
1 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17
 Just to be sure, packets 4,
Time
 [4 points] Consider the packet generation and reception sequence shown above. The first packet 5, and 6 arrive at the same
is generated at t=1 and is received at t=8. time "13" right?
 A. If Playout delay is zero and playout begins at t=8, which of the packets will not arrive in  For A, if playout delay is 0,
time? Show in a table.
then does the 'packets
 B. What is the minimum playout delay at the receiver that result in all of the first eight packets
arriving in time for their playout? generated' line start from
t=8 instead of t=1?
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 Adaptive Playout Delay
Estimate network delay, adjust playout delay at beginning of
each talk spurt
Questions
Silent periods compressed and elongated 
Chunks still played out every 20 ms during talk spurt
Adaptively estimate packet delay: Similar to TCP RTT estimate

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 Adaptive Playout Delay
ti =Sending time
ri= Receiving time Questions
Measured delay sample = ri-ti
di= Average network delay 
di= (1-a)di-1+a(ri-ti)
vi= Variation of the delay
vi= (1-a)vi-1+a|ri-ti-di|
pi= Playout time
pi = ti + di + Kvi
Here K is a constant, say 4.
Sequence numbers and timestamps used to determine talk spurts and
silence

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 Recovering From Packet Loss
Forward Error Correction
Send n+1 packets in place of n packets
Questions
Send a lower-resolution stream in addition
Play out the old syllable  From the book: "The
obvious disadvantage of
Busty Loss ⇒ Interleave audio/video frames interleaving is that it
increases latency. This
limits its use for
conversational applications
such as VoIP, although it
can perform well for
streaming stored audio."
Why does interleaving
increase latency?
 What is the error
correction packet in the
diagram?
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 Voice-over-IP: Skype
Proprietary
Skype clients (SC)
application-layer protocol Questions
(inferred via reverse
engineering)  You said these super nodes
Encrypted messages have "special functions".
Skype Are there any other "special
P2P: Media does not go login server Super node (SN) functions" the super nodes
through a central server are responsible for other
Clients: Skype peers connect Super node than keeping track of where
overlay the clients are?
directly to each other for VoIP network  Super nodes would be
call dedicated servers to hold the
Super Nodes (SN): Skype calls? You mentioned that
you didn't know if was
peers with special functions people or owned by Skype,
Overlay Network: Among but they are acting as servers
SNs to locate clients to "host" the VoIP?

Login server
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9-19
 P2P voice-over-IP: Skype
Skype client operation:
1. Joins Skype network by Questions
contacting SN (IP address  Can you explain the
relationship between NATs and
cached) using TCP firewalls?
2. Logs-in: Username, Skype
login server
 From the book: "Alice’s and
Bob’s super peers then instruct
password to centralized Alice and Bob respectively to
initiate a session with he relay.
Skype login server As shown in Figure 9.7, Alice
then sends voice packets to the
3. Obtains IP address for relay over the Alice-to-relay
connection (which was
callee from SN, SN overlay initiated by Alice), and the
relay then forwards these
4. Initiate call directly to packets over the relay-to-Bob
callee via SN connection (which was
initiated by Bob);"
 The book seems to indicate
that when connection is setup
via relay the two hosts never
actually connect directly. The
lecture seemed to say the SN
where used so that the two
could establish a direct
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9-20
 Skype: Super Nodes as Relays
Problem: both Alice, Bob are
behind “NATs” Questions
NAT prevents outside peer from
initiating connection to inside
peer 
Inside peer can initiate
connection to outside
Relay solution: Alice, Bob
maintain open connection to their
SNs
Alice signals her SN to connect
to Bob
Alice’s SN connects to Bob’s
SN
Bob’s SN connects to Bob over
open connection Bob initiated

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 Review
Voice over IP

1. Talk spurts are delayed to overcome jitter Questions

2. Playout delay is estimated adaptively using mean and
standard deviation  How does mobile
wireless VoIP
3. Forward error correction and interleaving is used to overcome
work? Will we
losses and burst errors
learn about this in
4. Skype uses super nodes to help connect peers. A login server the Wireless
is used for authentication. course?
5. Skype nodes maintain an outgoing connection with the super
nodes. These connections are used for incoming VoIP packets.

Ref: Section 9.3, Review Question R9-R11, Problems P6-P14

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9-22
 Overview Protocols for Real-Time Applications

1. Real-Time Transport Protocol (RTP) Questions

2. Session Initiation Protocol (SIP)

3. H.323 Protocols

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 Real-Time Transport Protocol (RTP)

Common sublayer between applications and UDP

Questions
Provides sequence numbers, timestamps, and other facilities
Supports both unicast and multicast 

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 RTP Packet Format
Marker Payload Sequence Time- Synchronization Miscellaneous
Type Number stamp Source Identifier Fields
1b 7b 16b 32b 32b Questions
Marker indicates that the packet contains special data required
by some applications  From the book: "The
payload type field in the
SSRC = Synchronization Source = Stream # RTP packet is 7 bits
long"
Payload Coding Rate
 Slides say 8. Which is
Type typo?
0 PCM mu-law 64 kbps  Does RTP offer
3 GSM 13 kbps something like h.264 or
h.265? I noticed that
7 LPC 2.4 kbps h.261 is specified which
26 Motion JPEG is a fairly old
compression/encoding
31 H.261
technology from the h.2X
33 MPEG2 video family.
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 Session Initiation Protocol (SIP)

Application level signaling protocol for voice and Questions

video conferencing over Internet
Allows creating, modifying, terminating sessions with 
one or more participants
Carries session descriptions (media types) for user
capabilities negotiation
Supports user location, call setup, call transfers
Supports mobility by proxying and redirection

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 SIP (Cont)
SIP Uniform Resource Identifiers (URIs):
Similar to email URLs Questions
sip:[email protected]
sip:+1-614-292-3989:[email protected]?subject=lecture  Is sip part of the URI in these
slides or is that just a label?
SIP can use UDP or TCP  If so, is this common
practice or can the URI be
SIP messages are sent to SIP servers: anything?
So SIP is similar to skype?
Registrar: Clients register and tell their location to it 
 Does the proxy have any
Location: Given name, returns possible addresses for a user. authentication? I have read it
that it uses tags and nonces.
Like Directory service or DNS. Does the proxy is constantly
authenticating while both
Redirect: Returns current address to requesters parties are communicating
Proxy: Intermediary. Acts like a server to internal client and through RTP?

like a client to external server

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9-27
 Locating using SIP
Allows locating a callee at different locations
Callee registers different locations with Registrar Questions
The seemed to indicate that if a proxy found
that an URI had moved the proxy would
SIP Messages: Ack, Bye, Invite, Register, Redirection, ... directly contact the new location. The example
in the slides shows the proxy responding to the
host and the host sending the new request.
Excerpt from the book:
"(1) Jim sends an INVITE message to
the umass SIP proxy. (2) The proxy does a
DNS lookup on the SIP registrar upenn.edu
(not shown in diagram) and then forwards the
message to the registrar server. (3) Because
[email protected] is no longer registered at the
Redirect upenn registrar, the upenn registrar sends a
redirect response, indicating that it should try
[email protected]. (4) The umass proxy sends an
X Server Jain@cis Jain@acm INVITE message to the NYU SIP registrar. (5)
The NYU registrar knows the IP address of
[email protected] and forwards the INVITE
Invite Jain@wustl message to the host 197.87.54.21, which is
running Keith’s SIP client. (6–8) An SIP
response is sent back through
Moved to Jain@acm registrars/proxies to the SIP client on
217.123.56.89. (9) Media is sent directly

Invite Jain@acm
between the two clients. (There is also an SIP
acknowledgment message, which is not
shown.)"
Ack Jain@acm What is this box with dashed line Jain@cis? Is
it just a server that does not have SIP server?
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 SIP Proxy

Questions
SIP Invite
SIP Trying SIP Invite For last two actions
SIP 180 Ringing SIP 180 Ringing "SIP Bye" and "SIP
OK":
SIP 200 OK SIP 200 OK The slides shows
SIP Ack SIP Ack "Bye" from right to left
& "OK" from left to
Conversation using RTP right. Can it be
reverse? i.e., "Bye"
SIP Bye SIP Bye from left to right &
SIP OK SIP OK "OK" from right to
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 H.323 Protocols
Multimedia over LANs, V1 (June 96), V2(Feb 98)
Provides component descriptions, signaling Questions
procedures, call control, system control, audio/video So are the protocols
codecs, data protocols listed under "Video"
Video Audio Control and Management Data and "Audio" header
only video and audio
G.711, G.722, H.225.0 H.225.0 H.245 T.124
H.261 RTCP protocols (not
G.723.1, G.728, RAS Signaling Control
H.263 network), and then the
G.729
protocols under
RTP X.224 Class 0 T.125 "Control and
UDP TCP T.123 Management" and
Network (IP)
"Data" network
Datalink (IEEE 802.3)
protocols?
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9-30
 Review
Protocols for Real-Time Applications

1. RTP is used to transmit multimedia over UDP Questions

2. SIP is a signaling (control) protocol to establish 
multimedia connections
3. H.323 is a framework for a group of protocols used
for multimedia

Ref: Section 9.4, Review questions R12-R13, Problems P15-P16

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9-31
 Overview Networking Support for Multimedia

1. QoS Components Questions

2. Traffic Shaping

3. Token Bucket Shaper
4. Traffic Policing
5. Differentiated Services

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QoS Components
1 Signaling:What I need
2 Admission control 4 5
Shaping Policing Classification Questions
3


Scheduling 6
8 Routing

Buffer Mgmt
9 Traffic Monitoring 7
Drop Policies
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QoS Components (Cont)
1. Signaling: Users need to tell/negotiate their QoS requirements with
the network
2. Admission Control: Network can deny requests that it can not meet Questions
3. Shaping: Traffic is smoothed out so that it is easier to handle From the book: "When
4. Policing: Ensuring that the users are sending at the rate they agreed demand increases are
to forecasted, the ISPs deploy
5. Marking/Classification: Packets are classified based on the source, additional bandwidth and
destination, TCP ports (application) switching capacity to continue
to ensure satisfactory delay
6. Scheduling: Different flows get appropriate treatment and packet-loss performance"
7. Drop Policies: Low priority packets are dropped. Do they change the protocols
to ones that are more fit for
8. Routing: Packets are sent over paths that can meet the QoS high demand but worse for
9. Traffic Management: Sources may be asked to reduce their rates to normal cases? If not why do
meet the loss rate and delay guarantees they not always operate at full
capacity?

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 Traffic Shaping
Altering the traffic characteristics of a given flow is
called traffic shaping Questions
The source must shape its traffic prior to sending it to 
network so it does not violate traffic contract

Shaper
Arriving traffic Leaving traffic with
with undesired desired characteristics
characteristics

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 Token Bucket Shaper
Questions
Tokens arrive periodically at
Is the token system
Average Rate independent of packet size?
For example a 512 byte
packet would take one
Token Bucket Size K token and a 512 kB packet
would take one token? or
could the bucket send
several smaller packets with
one token?
Server Shaped Traffic
Just bucket size always
Incoming Traffic regulate the bucket size or
does it depend on the set
up?
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 Traffic Policing
Users violating the traffic contract can jeopardise the Questions
QoS of other connections does policing and shaping happen at the same
time?

The network must protect well behaving users against Can you give an example of users violating the
traffic contract?
such traffic violations Is the flow shaped and policed at the same time
by the token bucket?
Policing functions are deployed at the edge (entry) of From the book: "The call admission process
described above requires that a call be able to
the network Conforming traffic reserve sufficient resources at each and every
network router on its source-to-destination path
admitted into network to ensure that its end-to-end QoS requirement is
met. Each router must determine the local
resources required by the session, consider the
Policer amounts of its resources that are already
committed to other ongoing sessions, and
Arriving traffic determine whether it has sufficient resources to
satisfy the per-hop QoS requirement of the
session at this router without violating local
Non-conforming QoS guarantees made to an already- admitted
session." How is routing table (path)
traffic (dropped) recalculation handled. Is path recalculated and
QOS guarantees rechecked or if a router has
packets with guarantees is it forbidden from
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 Peak Rate Policing with Leaky
Bucket
Enforces sustained rate and maximum
burst size Incoming Questions
Packets
Requires only one counter When do counters get
decremented? when packet
counter is decremented, to a leaves?
minimum of zero, at the avg rate Rejected
Is token bucket = leaky bucket?
counter is incremented by one, to a Are the rejected packets resent?
maximum of a limiting value, for each Can you explain how the token
packet arrival bucket can police the peak rate?
Does the conforming traffic get a
An arriving packet is non-conforming if token, and the non-conforming
counter is at its limit Accepted ones don't?
For the counter, is it
"decremented" and "incremented"
at the same time? This is
confusing.
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 Fair Queueing
Questions
S For Fair Queueing:
Would it be correct to
say that only one flow is
served at a time, and
Bit-level round robin but packet level scheduling within that flow
faster/smaller packets are
Count the packet size and determine which packet scheduled before the
would finish first. Serve that packet. others? Trying to
understand the second
Each flow gets the same number of bits/sec bullet point.
Fair queuing: Does
bits/sec include size of
header fields or only
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 Weighted Fair Queueing (WFQ)
Questions
S With weighted fair queuing,
do we transfer all three
queues at different rates at
the same time? Or do we
Fair queueing with different weight for each queue transfer queue with the
highest weights first
Flow 1 gets x bit/sec
What are the different
Flow 2 gets y bit/sec usages for fair queuing and
weighted fair queuing?
Flow n gets z bit/sec If x = 0.5, I am currently
Here, x, y, z are weights interpreting the slide "Flow
1 gets x bit/sec" to be
"Flow 1 gets 0.5 bit/sec" is
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 Maximum Delay with WFQ and Policing
Max Delay dmax = bi/(R wi/Σ wj)
Here, Questions
bi=Burst size of ith flow
To clarify, sigma W_j
R=Service Rate means the sum of every
Wi=Weight of ith flow packet's weight?
Do higher weights have
arriving token rate, r higher priority because a
traffic lower weight corresponds
bucket size, b to a larger maximum
delay? If not, can you
per-flow explain the weight to
rate, R maximum delay
WFQ relationship once more?
is average rate: r * R *
D = b/R (W_i / SUM(W))?
max
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 Differentiated Services
Ver Hdr Len Precedence ToS Unused Tot Len Questions
4b 4b 3b 4b 1b 16b
1) Is it correct to say that
something like WFQ +
DSCP ECN leaky bucket is an
6b 2b example of an
IPv4: 3-bit precedence + 4-bit ToS implementation for DS?
OSPF and integrated IS-IS can compute paths for each ToS 2) Is MPLS an example
of DS or QoS
Many vendors use IP precedence bits but the service varies
⇒ Need a standard ⇒ Differentiated Services What does "Tot" stand
for in "Tot Len"?
Edge routers can mark the packets ⇒ Set ToS field
Core routers use ToS field to provide "Per-Hop-Behavior"
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 Per-hop Behaviors
In PHB Out Questions

How does this work in

Externally Observable Forwarding Behavior practice? Wouldn't
x% of link bandwidth everyone want to send
Minimum x% and fair share of excess bandwidth out their packets with
very high per hop
Priority relative to other PHBs behavior x values, if
their routers are setting
these fields?

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 Expedited Forwarding
Also known as “Premium Service” Questions
Virtual leased line
If we are using 2 bits, shouldn't
Guaranteed minimum service rate there be 4 drop preferences?
Which 6 bits are these? Slide 42
Policed: Arrival rate < Minimum Service Rate shows groups of 3 and 4 bits and 1
unused bit.
Not affected by other data PHBs Can you explain more about
⇒ Highest data priority (if priority queueing) "policed: arrival rate < minimum
service rate"? If arrival rate <
minimum service rate, how can
Code point: 101 110 they guarantee the minimum
service rate if the data isn't coming
fast enough?
Where does the code "101 110" fill
amongst those fields on Slide Page
42?
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 Assured Forwarding
Questions

PHB Group
Four Classes: No particular ordering
Three drop preference per class

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 Assured Forwarding (Cont)
DS nodes SHOULD implement all 4 classes
and MUST accept all 3 drop preferences. Can implement 2 drop Questions
preferences.
Similar to nrt-VBR/ABR/GFR What is the difference
between the classes?
Code Points: The different classes could be
for specific applications? For
example, Netflix would have a
different class than a Zoom or
Drop Prec. Class 1 Class 2 Class 3 Class 4 VoIP call?
Low 010 000 011 000 100 000 101 000 Can you set it so that for
Medium 010 010 011 010 100 010 101 010 example VoIP packets drop
preference increases with time
High 010 100 011 100 100 100 101 100 s.t as they both more likely to
arrive to late for playback they
will simply be dropped to free
 Avoids 11x000 (used for network control) resources.
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 Network Support for Multimedia

Questions

1) How can hard

guarantees ever be
ensured?
Hardware can
always fail.

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 Review Network Support for
Multimedia
Questions
1. QoS is obtained using several components including
shaping, policing, differentiated services 
2. Shaping is done by a token bucket
3. Policing is done using a leaky bucket
4. Differentiated services specifies per-hop behaviors
1. Expedited Forwarding: min service rate
2. Assured Forwarding: 4 classes, 3 drop precedence's

Ref: Section
Washington 9.5, Problems
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 Summary
Questions

Is the goal of SIP

1. Multimedia applications require bounded delay,
registrar is to keep
delay jitter, and minimum throughput
track of user's IP
2. RTP allows sequencing and time stamping addresses while a
3. SIP allows parameter negotiation and location RTP is established
4. QoS requires shaping, policing, scheduling, etc. and data is being
5. Diffserv allows different packets to get different exchanged?
service
Ref: Entire Chapter 9, Review Exercises R1-R12, Problems P2-P4,P9, P11, P16, P19
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 Acronyms
ABR Available Bit Rate
CBR Constant Bit Rate Questions
CD Compact Disk
DNS Domain Name System
DS DiffServe
GFR Guaranteed Frame Rate
HTTP HyperText Transfer Protocol
IEEE Institution of Electrical and Electronics Engineers
IP Internet Protocol
IPTV Internet Protocol Television
IPv4 Internet Protocol Version 4
IS Integrated Services
LAN Local Area Network
NAT Network Address Translator
OSPF Open Shortest Path First
PHB Per-Hop Behavior
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 Acronyms (Cont)
QoS Quality of Service
RAS Registration, Admission, and Status Questions
RTCP Real-Time Transport Protocol Control Protocol
RTP Real-Time Transport Protocol
RTSP Real-Time Streaming Protocol 
RTT Round Trip Time
SC Skype Clients
SIP Session Initiation Protocol
SN Super Node
SSRC Synchronization Source
TCP Transmission Control Protocol
ToS Type of Service
UDP User Datagram Protocol
URI Uniform Resource Identifiers
URL Uniform Resource Locator
VBR Variable Bit Rate
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 Acronyms (Cont)
VoIP Voice over IP
Questions

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 Scan This to Download These Slides

Questions

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http://rajjain.com

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 Related Modules
CSE 567: The Art of Computer Systems Performance Analysis
https://www.youtube.com/playlist?list=PLjGG94etKypJEKjNAa1n_1X0bWWNyZcof
Questions
CSE473S: Introduction to Computer Networks (Fall 2011), 
https://www.youtube.com/playlist?list=PLjGG94etKypJWOSPMh8Azcgy5e_10TiDw

CSE 570: Recent Advances in Networking (Spring 2013)

https://www.youtube.com/playlist?list=PLjGG94etKypLHyBN8mOgwJLHD2FFIMGq5

CSE571S: Network Security (Spring 2011),

https://www.youtube.com/playlist?list=PLjGG94etKypKvzfVtutHcPFJXumyyg93u

Video Podcasts of Prof. Raj Jain's Lectures,

https://www.youtube.com/channel/UCN4-5wzNP9-ruOzQMs-8NUw

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