Chapter 9 - Computer Networking a top-down Approach 7th

Computer
Networking: A Top
Down Approach
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All material copyright 1996-2016
J.F Kurose and K.W. Ross, All Rights Reserved
7th edition
Jim Kurose, Keith Ross
Pearson/Addison Wesley
April 2016
Chapter 9
Multimedia
Networking
9-1Multimedia Networking

Multimedia networking: outline
9.1 multimedia networking applications
9.2 streaming stored video
9.3 voice-over-IP
9.4 protocols for real-time conversational
applications
9.5 network support for multimedia

Multimedia: audio
 analog audio signal
sampled at constant rate
• telephone: 8,000
samples/sec
• CD music: 44,100
samples/sec
 each sample quantized,
i.e., rounded
• e.g., 28=256 possible
quantized values
• each quantized
value represented by
bits, e.g., 8 bits for
256 values
time
audiosignalamplitude
analog
signal
quantized
value of
analog value
quantization
error
sampling rate
(N sample/sec)

Multimedia: audio
 example: 8,000
samples/sec, 256
quantized values: 64,000
bps
 receiver converts bits back
to analog signal:
• some quality reduction
example rates
 CD: 1.411 Mbps
 MP3: 96, 128, 160 kbps
 Internet telephony: 5.3
kbps and up
time
audiosignalamplitude
analog
signal
quantized
value of
analog value
quantization
error
sampling rate
(N sample/sec)

 video: sequence of
images displayed at
constant rate
• e.g., 24 images/sec
 digital image: array of
pixels
• each pixel represented
by bits
 coding: use redundancy
within and between
images to decrease #
bits used to encode
image
• spatial (within image)
• temporal (from one
image to next)
Multimedia: video
……………………...…
spatial coding example: instead
of sending N values of same
color (all purple), send only two
values: color value (purple) and
number of repeated values (N)
……………………...…
frame i
frame i+1
temporal coding example:
instead of sending
complete frame at i+1,
send only differences from
frame i

Multimedia: video
……………………...…
spatial coding example: instead
of sending N values of same
color (all purple), send only two
values: color value (purple) and
number of repeated values (N)
……………………...…
frame i
frame i+1
temporal coding example:
instead of sending
complete frame at i+1,
send only differences from
frame i
 CBR: (constant bit rate):
video encoding rate fixed
 VBR: (variable bit rate):
video encoding rate
changes as amount of
spatial, temporal coding
changes
 examples:
• MPEG 1 (CD-ROM) 1.5
Mbps
• MPEG2 (DVD) 3-6
Mbps
• MPEG4 (often used in
Internet, < 1 Mbps)

Multimedia networking: 3 application
types
 streaming, stored audio, video
• streaming: can begin playout before downloading
entire file
• stored (at server): can transmit faster than
audio/video will be rendered (implies
storing/buffering at client)
• e.g., YouTube, Netflix, Hulu
 conversational voice/video over IP
• interactive nature of human-to-human
conversation limits delay tolerance
• e.g., Skype
 streaming live audio, video
• e.g., live sporting event (futbol)

9.3 voice-over-IP
applications

Streaming stored video:
1. video
recorded
(e.g., 30
frames/sec)
2. video
sent
streaming: at this time, client
playing out early part of video,
while server still sending later
part of video
network delay
(fixed in this
example)
time
3. video received,
played out at client
(30 frames/sec)

challenges
 continuous playout constraint: once client
playout begins, playback must match original
timing
• … but network delays are variable (jitter),
so will need client-side buffer to match
playout requirements
 other challenges:
• client interactivity: pause, fast-forward,
rewind, jump through video
• video packets may be lost, retransmitted

constant bit
rate video
transmission
time
variable
network
delay
client video
reception
constant bit
rate video
playout at client
client playout
delay
buffered
video
 client-side buffering and playout delay:
compensate for network-added delay, delay
jitter
revisited

Client-side buffering, playout
variable fill
rate, x(t)
client application
buffer, size B
playout rate,
e.g., CBR r
buffer fill level,
Q(t)
video server
client

variable fill
rate, x(t)
client application
buffer, size B
playout rate,
e.g., CBR r
buffer fill level,
Q(t)
video server
client
1. Initial fill of buffer until playout begins at tp
2. playout begins at tp,
3. buffer fill level varies over time as fill rate x(t)
varies and playout rate r is constant

playout buffering: average fill rate (x), playout
rate (r):
 x < r: buffer eventually empties (causing freezing of
video playout until buffer again fills)
 x > r: buffer will not empty, provided initial playout
delay is large enough to absorb variability in x(t)
• initial playout delay tradeoff: buffer starvation less
likely with larger delay, but larger delay until user
begins watching
variable fill
rate, x(t)
client application
buffer, size B
playout rate,
e.g., CBR r
buffer fill level,
Q(t)
video server

Streaming multimedia: UDP
 server sends at rate appropriate for client
• often: send rate = encoding rate = constant
rate
• transmission rate can be oblivious to
congestion levels
 short playout delay (2-5 seconds) to remove
network jitter
 error recovery: application-level, time
permitting
 RTP [RFC 2326]: multimedia payload types
 UDP may not go through firewalls

Streaming multimedia: HTTP
 multimedia file retrieved via HTTP GET
 send at maximum possible rate under TCP
 fill rate fluctuates due to TCP congestion
control, retransmissions (in-order delivery)
 larger playout delay: smooth TCP delivery
rate
 HTTP/TCP passes more easily through
firewalls
variable
rate, x(t)
TCP send
buffer
video
file
TCP receive
buffer
application
playout buffer
server client

9.3 voice-over-IP
applications

Voice-over-IP (VoIP)
 VoIP end-end-delay requirement: needed to
maintain “conversational” aspect
• higher delays noticeable, impair interactivity
• < 150 msec: good
• > 400 msec bad
• includes application-level (packetization, playout),
network delays
 session initialization: how does callee
advertise IP address, port number, encoding
algorithms?
 value-added services: call forwarding,
screening, recording
 emergency services: 911

VoIP characteristics
 speaker’s audio: alternating talk spurts, silent
periods.
• 64 kbps during talk spurt
• pkts generated only during talk spurts
• 20 msec chunks at 8 Kbytes/sec: 160 bytes of
data
 application-layer header added to each chunk
 chunk+header encapsulated into UDP or TCP
segment
 application sends segment into socket every
20 msec during talkspurt

VoIP: packet loss, delay
 network loss: IP datagram lost due to network
congestion (router buffer overflow)
 delay loss: IP datagram arrives too late for
playout at receiver
• delays: processing, queueing in network; end-
system (sender, receiver) delays
• typical maximum tolerable delay: 400 ms
 loss tolerance: depending on voice encoding,
loss concealment, packet loss rates between
1% and 10% can be tolerated

constant bit
rate
transmission
time
variable
network
delay
(jitter)
client
reception
constant bit
rate playout
at client
client playout
delay
buffered
data
Delay jitter
 end-to-end delays of two consecutive
packets: difference can be more or less than
20 msec (transmission time difference)

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 in choosing q:
• large q: less packet loss
• small q: better interactive experience

packets
time
packets
generated
packets
received
loss
r
p p'
playout schedule
p' - r
playout schedule
p - 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’
VoIP: fixed playout delay

Adaptive playout delay (1)
 goal: low playout delay, low late loss rate
 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
 adaptively estimate packet delay: (EWMA -
exponentially weighted moving average, recall TCP
RTT estimate):di = (1-a)di-1 + a (ri – ti)
delay estimate
after ith packet
small constant,
e.g. 0.1
time received - time sent
(timestamp)
measured delay of ith packet

 also useful to estimate average deviation of delay, vi
 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:
 remaining packets in talkspurt are played out
periodically
vi = (1-b)vi-1 + b |ri – ti – di|
playout-timei = ti + di + Kvi

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.

VoiP: recovery from packet loss
(1)
Challenge: recover from packet loss given small
tolerable delay between original transmission
and playout
 each ACK/NAK takes ~ one RTT
 alternative: Forward Error Correction (FEC)
• send enough bits to allow recovery without
retransmission (recall two-dimensional parity in Ch.
5)
simple FEC
 for every group of n chunks, create redundant chunk by
exclusive OR-ing n original chunks
 send n+1 chunks, increasing bandwidth by factor 1/n
 can reconstruct original n chunks if at most one lost
chunk from n+1 chunks, with playout delay 9-27Multimedia Networking

another FEC scheme:
 “piggyback lower
quality stream”
 send lower resolution
audio stream as
redundant information
 e.g., nominal
stream PCM at 64 kbps
and redundant stream
GSM at 13 kbps
 non-consecutive loss: receiver can conceal loss
 generalization: can also append (n-1)st and (n-2)nd low-bit ra
chunk
(2)

interleaving to conceal
loss:
 audio chunks divided into
smaller units, e.g. four 5
msec units per 20 msec
audio chunk
 packet contains small
 if packet lost, still have
most of every original
chunk
 no redundancy overhead,
but increases playout
delay
(3)

supernode
overlay
network
Voice-over-IP: Skype
 proprietary application-
layer protocol (inferred
via reverse
engineering)
• encrypted msgs
 P2P components:
Skype clients (SC)
 clients: Skype peers
connect directly to
each other for VoIP
call
 super nodes (SN):
Skype peers with
special functions
 overlay network: among
SNs to locate SCs
 login server
Skype
login server supernode (SN)

P2P voice-over-IP: Skype
Skype client
operation:1. joins Skype network by
contacting SN (IP
address cached) using
TCP2. logs-in (username,
password) to
centralized Skype login
server3. obtains IP address for
callee from SN, SN
overlay
or client buddy list
4. initiate call directly to
callee
Skype
login server

 problem: both Alice, Bob
are behind “NATs”
• NAT prevents outside
peer from initiating
connection to insider 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
initially initiated to his SN
Skype: peers as relays

9.3 voice-over-IP
applications: RTP, SIP

Real-Time Protocol (RTP)
 RTP specifies
packet structure for
packets carrying
audio, video data
 RFC 3550
 RTP packet
provides
• payload type
identification
• packet sequence
numbering
• time stamping
 RTP runs in end
systems
 RTP packets
encapsulated in UDP
segments
 interoperability: if two
VoIP applications run
RTP, they may be
able to work together

RTP runs on top of UDP
RTP libraries provide transport-layer interface
that extends UDP:
• port numbers, IP addresses
• payload type identification
• packet sequence numbering
• time-stamping

RTP example
example: sending 64
kbps PCM-encoded
voice over RTP
 application collects
encoded data in
chunks, e.g., every 20
msec = 160 bytes in a
chunk
 audio chunk + RTP
header form RTP
packet, which is
encapsulated in UDP
segment
 RTP header
indicates type of
audio encoding in
each packet
• sender can change
encoding during
conference
 RTP header also
contains sequence
numbers,
timestamps

RTP and QoS
 RTP does not provide any mechanism to
ensure timely data delivery or other QoS
guarantees
 RTP encapsulation only seen at end systems
(not by intermediate routers)
• routers provide best-effort service, making
no special effort to ensure that RTP
packets arrive at destination in timely
matter

RTP header
payload type (7 bits): indicates type of encoding currently
being
used. If sender changes encoding during call, sender
informs receiver via payload type field
Payload type 0: PCM mu-law, 64 kbps
Payload type 3: GSM, 13 kbps
Payload type 7: LPC, 2.4 kbps
Payload type 26: Motion JPEG
Payload type 31: H.261
Payload type 33: MPEG2 video
sequence # (16 bits): increment by one for each RTP packet
sent
payload
type
sequence
number
type
time stamp Synchronization
Source ID
Miscellaneous
fields

 timestamp field (32 bits long): sampling instant
of first byte in this RTP data packet
• for audio, timestamp clock increments by one for
each sampling period (e.g., each 125 usecs for 8
KHz sampling clock)
• if application generates chunks of 160 encoded
samples, timestamp increases by 160 for each RTP
packet when source is active. Timestamp clock
continues to increase at constant rate when source
is inactive.
 SSRC field (32 bits long): identifies source of RTP
stream. Each stream in RTP session has distinct
RTP header
payload
type
sequence
number
type
time stamp Synchronization
Source ID
Miscellaneous
fields

RTSP/RTP programming
assignment
 build a server that encapsulates stored video
frames into RTP packets
• grab video frame, add RTP headers, create UDP
segments, send segments to UDP socket
• include seq numbers and time stamps
• client RTP provided for you
 also write client side of RTSP
• issue play/pause commands
• server RTSP provided for you

Real-Time Control Protocol
(RTCP)
 works in conjunction
with RTP
 each participant in
RTP session
periodically sends
RTCP control packets
to all other
participants
 each RTCP packet
contains sender and/or
receiver reports
• report statistics useful
to application: #
packets sent, # packets
lost, interarrival jitter
 feedback used to
control performance
• sender may modify its
transmissions based on
feedback

RTCP: multiple multicast senders
 each RTP session: typically a single multicast address;
all RTP /RTCP packets belonging to session use
multicast address
 RTP, RTCP packets distinguished from each other via
distinct port numbers
 to limit traffic, each participant reduces RTCP traffic as
number of conference participants increases
RTCP
RTP
RTCP
RTCP
sender
receivers

RTCP: packet types
receiver report packets:
 fraction of packets lost,
last sequence number,
average interarrival jitter
sender report packets:
 SSRC of RTP stream,
current time, number of
packets sent, number of
bytes sent
source description
packets:
 e-mail address of
sender, sender's name,
SSRC of associated
RTP stream
 provide mapping
between the SSRC and
the user/host name

RTCP: stream synchronization
 RTCP can synchronize
different media streams
within a RTP session
 e.g., videoconferencing
app: each sender
generates one RTP
stream for video, one
for audio.
 timestamps in RTP
packets tied to the
video, audio sampling
clocks
• not tied to wall-clock
time
 each RTCP sender-
report packet contains
(for most recently
generated packet in
associated RTP
stream):
• timestamp of RTP
packet
• wall-clock time for
when packet was
created
 receivers uses
association to
synchronize playout of
audio, video

RTCP: bandwidth scaling
RTCP attempts to limit
its traffic to 5% of
session bandwidth
example : one sender,
sending video at 2
Mbps
 RTCP attempts to limit
RTCP traffic to 100
Kbps
 RTCP gives 75% of
rate to receivers;
remaining 25% to
sender
 75 kbps is equally shared
among receivers:
• with R receivers, each
receiver gets to send RTCP
traffic at 75/R kbps.
 sender gets to send RTCP
traffic at 25 kbps.
 participant determines
RTCP packet transmission
period by calculating avg
RTCP packet size (across
entire session) and
dividing by allocated rate

SIP: Session Initiation Protocol [RFC
3261]
long-term vision:
 all telephone calls, video conference calls
take place over Internet
 people identified by names or e-mail
addresses, rather than by phone numbers
 can reach callee (if callee so desires), no
matter where callee roams, no matter what IP
device callee is currently using

SIP services
 SIP provides
mechanisms for call
setup:
• for caller to let
callee know she
wants to establish
a call
• so caller, callee
can agree on
media type,
encoding
• to end call
 determine current IP
address of callee:
• maps mnemonic
identifier to current
IP address
 call management:
• add new media
streams during call
• change encoding
during call
• invite others
• transfer, hold calls

Example: setting up call to known IP
address
 Alice’s SIP invite
message indicates her
port number, IP address,
encoding she prefers to
receive (PCM mlaw)
 Bob’s 200 OK message
indicates his port number,
IP address, preferred
encoding (GSM)
 SIP messages can be
sent over TCP or UDP;
here sent over RTP/UDP
 default SIP port number
is 5060
time time
Bob's
terminal rings
Alice
167.180.112.24
Bob
193.64.210.89
port 5060
port 38060
m Law audio
GSM
port 48753
INVITE bob@193.64.210.89c=IN IP4 167.180.112.24m=audio 38060 RTP/AVP 0
port 5060
200 OK
c=IN IP4 193.64.210.89
m=audio 48753 RTP/AVP 3
ACK
port 5060

Setting up a call (more)
 codec negotiation:
• suppose Bob doesn’t
have PCM mlaw
encoder
• Bob will instead reply
with 606 Not
Acceptable Reply,
listing his encoders.
Alice can then send
new INVITE message,
advertising different
encoder
 rejecting a call
• Bob can reject with
replies “busy,”
“gone,” “payment
required,”
“forbidden”
 media can be sent
over RTP or some
other protocol

Example of SIP message
INVITE sip:bob@domain.com SIP/2.0
Via: SIP/2.0/UDP 167.180.112.24
From: sip:alice@hereway.com
To: sip:bob@domain.com
Call-ID: a2e3a@pigeon.hereway.com
Content-Type: application/sdp
Content-Length: 885
c=IN IP4 167.180.112.24
m=audio 38060 RTP/AVP 0
Notes:
 HTTP message syntax
 sdp = session description protocol
 Call-ID is unique for every call
 Here we don’t
know Bob’s IP
address
• intermediate SIP
servers needed Alice sends,
receives SIP
messages using
SIP default port
506
 Alice specifies in
header that SIP
client sends,
receives SIP
messages over9-50Multimedia Networking

Name translation, user
location
 caller wants to call
callee, but only has
callee’s name or e-
mail address.
 need to get IP address
of callee’s current
host:
• user moves around
• DHCP protocol
• user has different IP
devices (PC,
smartphone, car
device)
 result can be based
on:
• time of day (work,
home)
• caller (don’t want boss
to call you at home)
• status of callee (calls
sent to voicemail when
callee is already talking
to someone)

SIP registrar
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/UDP 193.64.210.89
From: sip:bob@domain.com
To: sip:bob@domain.com
Expires: 3600
 one function of SIP server: registrar
 when Bob starts SIP client, client sends SIP
REGISTER message to Bob’s registrar server
register message:

SIP proxy
 another function of SIP server: proxy
 Alice sends invite message to her proxy server
• contains address sip:bob@domain.com
• proxy responsible for routing SIP messages to
callee, possibly through multiple proxies
 Bob sends response back through same set of
SIP proxies
 proxy returns Bob’s SIP response message to
Alice
• contains Bob’s IP address
 SIP proxy analogous to local DNS server plus
TCP setup

SIP example: jim@umass.edu calls
keith@poly.edu
1
1. Jim sends INVITE
message to UMass
SIP proxy.
2. UMass proxy forwards request
to Poly registrar server
2 3. Poly server returns redirect response,
indicating that it should try keith@eurecom.fr
3
5. eurecom
registrar
forwards INVITE
to 197.87.54.21,
which is running
keith’s SIP
client
5
4
4. Umass proxy forwards request
to Eurecom registrar server
8
6
7
6-8. SIP response returned to Jim
9
9. Data flows between clients
UMass
SIP proxy
Poly SIP
registrar
Eurecom SIP
registrar
197.87.54.21
128.119.40.186

Comparison with H.323
 H.323: another
signaling protocol for
real-time, interactive
multimedia
 H.323: complete,
vertically integrated
suite of protocols for
multimedia
conferencing: signaling,
registration, admission
control, transport,
codecs
 SIP: single component.
Works with RTP, but
does not mandate it.
Can be combined with
 H.323 comes from the
ITU (telephony)
 SIP comes from IETF:
borrows much of its
concepts from HTTP
• SIP has Web flavor;
H.323 has telephony
flavor
 SIP uses KISS
principle: Keep It
Simple Stupid

9.3 voice-over-IP
applications

Network support for multimedia

Dimensioning best effort
networks
 approach: deploy enough link capacity so that
congestion doesn’t occur, multimedia traffic flows
without delay or loss
• low complexity of network mechanisms (use current
“best effort” network)
• high bandwidth costs
 challenges:
• network dimensioning: how much bandwidth is
“enough?”
• estimating network traffic demand: needed to
determine how much bandwidth is “enough” (for that
much traffic)

Providing multiple classes of
service
 thus far: making the best of best effort service
• one-size fits all service model
 alternative: multiple classes of service
• partition traffic into classes
• network treats different classes of traffic differently
(analogy: VIP service versus regular service)
0111
 granularity: differential
service among
multiple classes, not
among individual
connections
 history: ToS bits

Multiple classes of service:
scenario
R1 R2
H1
H2
H3
H4
1.5 Mbps linkR1 output
interface
queue

Scenario 1: mixed HTTP and
VoIP
 example: 1Mbps VoIP, HTTP share 1.5 Mbps
link.
• HTTP bursts can congest router, cause audio loss
• want to give priority to audio over HTTP
packet marking needed for router to distinguish
between different classes; and new router policy to
treat packets accordingly
Principle 1
R1
R2

Principles for QOS guarantees
(more)
 what if applications misbehave (VoIP sends
higher than declared rate)
• policing: force source adherence to bandwidth
allocations
 marking, policing at network edge
provide protection (isolation) for one class from others
Principle 2
R1 R2
1.5 Mbps link
1 Mbps
phone
packet marking and policing

 allocating fixed (non-sharable) bandwidth to
flow: inefficient use of bandwidth if flows
doesn’t use its allocation
while providing isolation, it is desirable to use
resources as efficiently as possible
Principle 3
R1
R2
1.5 Mbps link
1 Mbps
phone
1 Mbps logical link
0.5 Mbps logical link
Principles for QOS guarantees
(more)

Scheduling and policing
mechanisms
 packet scheduling: choose next queued packet
to send on outgoing link
 previously covered in Chapter 4:
• FCFS: first come first served
• simply multi-class priority
• round robin
• weighted fair queueing (WFQ)
queue
(waiting area)
packet
arrivals
packet
departureslink
(server)

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)
• crucial question: what is the interval length: 100
packets per sec or 6000 packets per min have same
average!
 peak rate: e.g., 6000 pkts per min (ppm) avg.;
1500 ppm peak rate
 (max.) burst size: max number of pkts sent
consecutively (with no intervening idle)

Policing mechanisms:
implementation
token bucket: limit input to specified burst size
and average rate
 bucket can hold b tokens
 tokens generated at rate r token/sec unless
bucket full
 over interval of length t: number of packets
admitted less than or equal to (r t + b)

Policing and QoS guarantees
 token bucket, WFQ combine to provide
guaranteed upper bound on delay, i.e., QoS
guarantee!
WFQ
token rate, r
bucket size, b
per-flow
rate, R
D = b/R
max
arriving
traffic
arriving
traffic

Differentiated services
 want “qualitative” service classes
• “behaves like a wire”
• relative service distinction: Platinum, Gold, Silver
 scalability: simple functions in network core,
relatively complex functions at edge routers
(or hosts)
• signaling, maintaining per-flow router state
difficult with large number of flows
 don’t define define service classes, provide
functional components to build service
classes

edge router:
 per-flow traffic
management
 marks packets as in-
profile and out-profile
core router:
 per class traffic management
 buffering and scheduling based
on marking at edge
 preference given to in-profile
packets over out-of-profile
packets
Diffserv architecture
r
b
marking
scheduling
...

Edge-router packet marking
 class-based marking: packets of different classes marked
differently
 intra-class marking: conforming portion of flow marked
differently than non-conforming one
 profile: pre-negotiated rate r, bucket size b
 packet marking at edge based on per-flow
profile
possible use of marking:
user packets
rate r
b

Diffserv packet marking:
details
 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)
• determine PHB that the packet will receive
• 2 bits currently unused
DSCP unused

Classification, 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

Forwarding Per-hop Behavior
(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

Forwarding PHB
PHBs proposed:
 expedited forwarding: packet 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

Per-connection QOS
guarantees
 basic fact of life: can not support traffic
demands beyond link capacity
call admission: flow declares its needs, network may
block call (e.g., busy signal) if it cannot meet needs
Principle 4
R1
R2
1.5 Mbps link
1 Mbps
phone
1 Mbps
phone

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

9.3 voice-over-IP
applications

Chapter 9 - Computer Networking a top-down Approach 7th

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Chapter 9 - Computer Networking a top-down Approach 7th