Transcript slides - network systems lab @ sfu

School of Computing Science
Simon Fraser University
CMPT 771/471: Internet Architecture and
Protocols
Multimedia Networking and Quality of Service
Instructor: Dr. Mohamed Hefeeda
1
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.
2
Chapter 7: Goals
Principles
 Classify multimedia applications
 Identify the network services the apps need
 Making the best of best effort service
 Mechanisms for providing QoS
Protocols and Architectures
 Specific protocols for best-effort
 Architectures for QoS
3
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
Jitter is the variability
of packet delays within
the same packet stream
Fundamental
characteristics:
 Typically delay sensitive


end-to-end delay
delay jitter
 But loss tolerant:
infrequent losses cause
minor glitches
 In contrast to data,
which are loss intolerant
but delay tolerant.
4
A few words about audio compression
 Analog signal sampled
at constant rate


telephone: 8,000
samples/sec
CD music: 44,100
samples/sec
 Each sample quantized,
i.e., rounded

e.g., 28=256 possible
quantized values
 Each quantized value
represented by bits

8 bits for 256 values
 Example: 8,000
samples/sec, 256
quantized values -->
64,000 bps
 Receiver converts it
back to analog signal:

some quality reduction
Example rates
 CD: 1.411 Mbps
 MP3: 96, 128, 160 kbps
 Internet telephony:
5.3 - 13 kbps
5
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


spatial
temporal
Examples:
 MPEG 1 (CD-ROM) 1.5
Mbps
 MPEG2 (DVD) 3-6 Mbps
 MPEG4 (often used in
Internet, < 1 Mbps)
Research:
 Layered (scalable) video

adapt layers to available
bandwidth
6
Streaming Multimedia: UDP or TCP?
UDP
 server sends at rate appropriate for client (oblivious to
network congestion !)
 often send rate = encoding rate = constant rate
 then, fill rate = constant rate - packet loss
 short playout delay (2-5 seconds) to compensate for network
delay jitter
 error recovery: time permitting
TCP
 send at maximum possible rate under TCP
 fill rate fluctuates due to TCP congestion control
 larger playout delay: smooth TCP delivery rate
 HTTP/TCP passes more easily through firewalls
7
User Control of Streaming Media: RTSP
HTTP
 Does not target multimedia
content
 No commands for fast
forward, etc.
RTSP: RFC 2326
 Client-server application
layer protocol.
 For user to control display:
rewind, fast forward,
pause, resume,
repositioning, etc…
 RTSP is out-of-band
protocol (similar to FTP)
What it doesn’t do:
 does not define how
audio/video is encapsulated
for streaming over network
 does not restrict how
streamed media is
transported; it can be
transported over UDP or
TCP
 does not specify how the
media player buffers
audio/video
8
RTSP Example
:
 metafile communicated to web browser
 browser launches player
 player sets up an RTSP control connection, data
connection to streaming server
9
Real-time interactive applications
 PC-2-PC phone
 instant messaging services are
providing this
 PC-2-phone
Dialpad
 Net2phone
 videoconference with Webcams

10
Interactive Multimedia: Internet Phone
 speaker’s audio: alternating talk spurts, silent
periods.

64 kbps during talk spurt
 pkts generated only during talk spurts

20 msec chunks at 8 Kbytes/sec: 160 bytes data
 application-layer header added to each chunk
 Chunk+header encapsulated into UDP segment
 application sends UDP segment into socket every 20
msec during talk spurt
 Need to handle packet losses and delay jitter
11
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
12
Recovery from packet loss (2)
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
13
Recovery from packet loss (3)
Interleaving
 chunks are broken
up into smaller units
 for example, four 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
14
Handling Delay Jitter
 Receiver uses a playout buffer
 Fixed buffer
 Adaptive buffer: depends on network conditions
packets
loss
packets
generated
packets
received
playout schedule
p' - r
playout schedule
p-r
time
r
p’
p
15
Adaptive Playout Buffer (1)
 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.
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:
di  (1  u)di 1  u(ri  ti )
where u is a fixed constant (e.g., u = 0.01).
16
Adaptive Playout Buffer (2)
Also useful to estimate the average deviation of the delay, vi :
vi  (1  u)vi 1  u | ri  ti  di |
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  di  Kvi
where K is a positive constant.
Remaining packets in talk spurt are played out periodically
17
Real-Time Protocol (RTP): FRC 1889
 RTP specifies packet structure
for audio and video data



payload type identification
packet sequence numbering
time stamping
 RTP runs in the end systems
 RTP packets are encapsulated in
UDP segments
 RTP does not provide any
mechanism to ensure QoS

RTP encapsulation is only seen
at the end systems
18
RTP Header
Payload Type (7 bits): Indicates type of encoding currently being
used: e.g.,
•Payload type 0: PCM mu-law, 64 kbps
•Payload type 33, MPEG2 video
Sequence Number (16 bits): Increments by one for each RTP packet
sent, and may be used to detect packet
Timestamp field (32 bytes long). Reflects the sampling instant of the
first byte in the RTP data packet.
SSRC field (32 bits long). Identifies the source of the RTP stream.
Each stream in a RTP session should have a distinct SSRC.
19
Real-Time Control Protocol (RTCP)
 Works in conjunction with RTP
 Each participant in RTP session
periodically transmits RTCP control
packets to all other participants
 Each RTCP packet contains sender and/or
receiver reports



statistics, e.g., number of packets sent, number
of packets lost, interarrival jitter, etc.
used to control app performance
Sender may modify its transmissions based on
feedback
20
RTCP Packets
Receiver report packets:
 fraction of packets
lost, last sequence
number, average
interarrival jitter.
Sender report packets:
 SSRC of the RTP
stream, the current
time, the number of
packets sent, and the
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.
21
RTCP: Synchronization of Streams
 RTCP can synchronize
different media streams
within a RTP session.
 Consider videoconferencing
app for which each sender
generates one RTP stream
for video and one for audio.
 Timestamps in RTP packets
tied to the video and audio
sampling clocks
 not tied to the wallclock time
 Each RTCP sender-report
packet contains (for the
most recently generated
packet in the associated
RTP stream):


timestamp of the RTP
packet
wall-clock time for when
packet was created.
 Receivers can use this
association to synchronize
the playout of audio and
video.
22
Session Initiation Protocol (SIP)
 Comes from IETF
SIP long-term vision
 All telephone calls and video conference calls take
place over the Internet
 People are identified by names or e-mail
addresses, rather than by phone numbers
 You can reach the callee, no matter where the
callee roams, no matter what IP device the callee
is currently using.
23
SIP Services
 Setting up a call
 Provides mechanisms for
caller to let callee know
she wants to establish a
call
 Provides mechanisms so
that caller and callee can
agree on media type and
encoding.
 Provides mechanisms 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 and hold calls
24
Setting up a call to a known IP address
Bob
Alice
167.180.112.24
INVITE bob
@193.64.2
10.89
c=IN IP4 16
7.180.112.2
4
m=audio 38
060 RTP/A
VP 0
193.64.210.89
port 5060
port 5060
Bob's
terminal rings
200 OK
.210.89
c=IN IP4 193.64
RTP/AVP 3
3
75
m=audio 48
ACK
port 5060
• Alice’s SIP invite
message indicates her
port number & IP address.
Indicates encoding that
Alice prefers to receive
(PCM ulaw)
• Bob’s 200 OK message
indicates his port number,
IP address & preferred
encoding (GSM)
m Law audio
port 38060
GSM
time
port 48753
time
• SIP messages can be
sent over TCP or UDP;
here sent over RTP/UDP.
•Default SIP port number
is 5060.
25
Name translation and 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, PDA, 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)
Service provided by SIP
servers:
 SIP registrar server
 SIP proxy server
26
SIP Registrar
 When Bob starts SIP client, client sends SIP
REGISTER message to Bob’s registrar server
(similar function needed by Instant Messaging)
Register Message:
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/UDP 193.64.210.89
From: sip:[email protected]
To: sip:[email protected]
Expires: 3600
27
SIP Proxy
 Alice sends invite message to her proxy server
 contains address sip:[email protected]
 Proxy responsible for routing SIP messages to
callee

possibly through multiple proxies.
 Callee sends response back through the same set
of proxies.
 Proxy returns SIP response message to Alice

contains Bob’s IP address
 Note: proxy is analogous to local DNS server
28
Example
Caller [email protected] places
a call to [email protected]
(1) Jim sends INVITE
message to umass SIP
proxy. (2) Proxy forwards
request to upenn
registrar server.
(3) upenn server returns
redirect response,
indicating that it should
try [email protected]
SIP registrar
upenn.edu
SIP
registrar
eurecom.fr
2
SIP proxy
umass.edu
1
3
4
5
7
8
6
9
SIP client
217.123.56.89
SIP client
197.87.54.21
(4) umass proxy sends INVITE to eurecom registrar. (5) eurecom
registrar forwards INVITE to 197.87.54.21, which is running keith’s SIP
client. (6-8) SIP response sent back (9) media sent directly
between clients.
Note: also a SIP ack message, which is not shown.
29
Comparison with H.323
 H.323 is another signaling
protocol for real-time,
interactive
 H.323 is a complete,
vertically integrated suite
of protocols for multimedia
conferencing: signaling,
registration, admission
control, transport and
codecs.
 SIP is a single component.
Works with RTP, but does
not mandate it. Can be
combined with other
protocols and services.
 H.323 comes from the ITU
(telephony).
 SIP comes from IETF:
Borrows much of its
concepts from HTTP. SIP
has a Web flavor, whereas
H.323 has a telephony
flavor.
 SIP uses the KISS
principle: Keep it simple
stupid.
30
Improving QoS in IP Networks
Thus far: “making the best of best effort”
Future: next generation Internet with QoS guarantees

What do we need to do to get QoS guarantees?
 simple model for sharing and congestion studies:
31
Principles for QoS Guarantees
 Example: 1Mbps IP 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
32
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
33
Principles for QoS Guarantees (more)
 Allocating fixed (non-sharable) bandwidth to flow:
inefficient use of bandwidth if flows doesn’t use
its allocation
Principle 3
While providing isolation, it is desirable to use
resources as efficiently as possible
34
Principles for QoS Guarantees (more)
 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
35
Summary of QoS Principles
Let’s next look at mechanisms for achieving this ….
36
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

discard policy: if packet arrives to full queue: what to
discard?
• Tail drop: drop arriving packet
• priority: drop/remove on priority basis
• random: drop/remove randomly
37
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
38
Scheduling Policies: still more
Weighted Fair Queuing:
 generalized Round Robin
 each class gets weighted amount of service in each
cycle
39
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 and 6000 packets per min (ppm) have same average!
 Peak Rate: e.g.,
 Avg rate: 6000 ppm
 Peak rate: 1500 ppm
 (Max.) Burst Size: max. number of pkts sent
consecutively (with no intervening idle period)
40
Policing Mechanisms
Leaky 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).
41
Policing Mechanisms (more)
 Leaky bucket + WFQ  provide guaranteed upper bound
on delay, i.e., QoS guarantee! How?


WFQ: guaranteed share of bandwidth
Leaky bucket: limit max number of packets in queue (burst)
Ri  R wi /  w j
d
max
i
 bi / Ri
42
IETF Integrated Services (IntServ)
 architecture for providing QoS guarantees in IP
networks for individual application sessions
 resource reservation: routers maintain state info
of allocated resources, QoS req’s
 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?
43
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)
44
Call Admission
Arriving session must:
 declare its QoS requirement
R-spec: 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

45
IntServ QoS: Service models [rfc2211, rfc 2212]
Guaranteed service:
 worst case traffic arrival: leaky-bucket-policed source
 simple (mathematically provable) bound on delay [Parekh
1993, Cruz 1988]
arriving
traffic
token rate, r
bucket size, b
WFQ
per-flow
rate, R
D = b/R
max
46
IETF Differentiated Services (DiffServ)
Concerns with IntServ:
 Scalability: signaling, maintaining per-flow router
state difficult with large number of flows

Example: OC-48 (2.5 Gbps) link serving 64 Kbps audio
streams  39,000 flows! Each requires state maintenance.
 Flexible Service Models: Intserv has only two classes.
Also want “qualitative” service classes

relative service distinction: Platinum, Gold, Silver
DiffServ approach:
 simple functions in network core, relatively complex
functions at edge routers (or hosts)
 Don’t define service classes, provide functional
components to build service classes
47
DiffServ Architecture
Edge router:
r
 per-flow traffic management
 Classifies (marks) pkts
 different classes
 within a class: in-profile
b
marking
scheduling
..
.
and out-profile
Core router:
 per class traffic management
 buffering and scheduling based
on marking at edge
 preference given to in-profile
packets
48
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
49
Edge-router: 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 Per-Hop Behavior (PHB)
that the packet will receive
 2 bits are currently unused
50
Edge-router: 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
51
Core-router: 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
52
Core-router: Forwarding (PHB)
PHBs being developed:
 Expedited Forwarding (EF): pkt departure rate of a
class equals or exceeds specified rate



logical link with a minimum guaranteed rate
May require edge routers to limit EF traffic rate
Could be implemented using strict priority scheduling or
WFQ with higher weight for EF traffic
 Assured Forwarding: multiple traffic classes,
treated differently


amount of bandwidth allocated, or drop priorities
Can be implemented using WFQ+leaky bucket or RED
(Random Early Detection) with different threshold values.
• See Sections. 6.4.2 and 6.5.3 in [PD07]
53
Intserv and Diffserv: Discussion
 For over 20 years, many attempts failed to introduce
QoS into packet-switched networks. Why?
 Both Intserv and Diffserv need collaboration between
all ISPs, otherwise you can not provide guarantee 
hard
 QoS needs accounting, extra processing (shaping,
marking, policing,…)  complex and costly
equipment/management  charge customers more
 Most of the time there would be no perceived
difference between Best-effort and
Intserv/Diffserv services, at moderate load

So why would customers pay more??
54
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 (minimum
intervention)
 no major changes
 more bandwidth when
needed
 content distribution,
application-layer multicast

application layer
Differentiated services
philosophy:
 Fewer changes to Internet
infrastructure, yet provide
1st and 2nd class service.
What’s your opinion?
55
Multimedia Networking: Summary
 multimedia applications and requirements
 making the best of today’s best effort
service
 scheduling and policing mechanisms
 next generation Internet: Intserv, RSVP,
Diffserv
56