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Review Computer Communication II
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
1
Part I – Multimedia networking
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Taxonomy of multimedia applications
Streaming stored audio and video
making the best out of best effort service
protocols for real-time interactive applications
RTP,RTCP
• providing multiple classes of service
• providing QoS guarantees
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
2
MM Networking Applications
Classes of MM applications:
1) stored streaming
2) live streaming
3) interactive, real-time
Jitter is the variability
of packet delays within
the same packet stream
Karlstads Universitet
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Fundamental characteristics:
• typically delay sensitive
– end-to-end delay
– delay jitter
• loss tolerant: infrequent losses
cause minor glitches
• antithesis of data, which are
loss intolerant but delay
tolerant.
Datakommunikation II
Katarina Asplund
3
Multimedia Over Today’s Internet
TCP/UDP/IP: “best-effort service”
• no guarantees on delay, loss
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But you said multimedia apps requires
QoS and level of performance to be
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Today’s Internet multimedia applications
use application-level techniques to mitigate
(as best possible) effects of delay, loss
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
4
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, applicationlayer multicast
– application layer
Differentiated services philosophy:
• fewer changes to Internet
infrastructure, yet provide 1st and
2nd class service
What’s your opinion?
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
5
Streaming Stored Multimedia
application-level streaming
techniques for making the
best out of best effort
service:
– client-side buffering
– use of UDP versus TCP
– multiple encodings of
multimedia
• Use web server or streaming
server
Karlstads Universitet
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Media Player
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jitter removal
decompression
error concealment
graphical user interface
w/ controls for interactivity
(RTSP)
Datakommunikation II
Katarina Asplund
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 remove network jitter
• error recover: 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
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
7
Real-time applications
• 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, losses
concealed, packet loss rates between 1% and 10% can be
tolerated.
• Delay jitter
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Mechanisms
• Jitter
– Seq#, timestamps, delaying playout
– Playout delay: fixed, adaptive
• Loss
– Forward error correction (FEC)
– Interleaving
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
9
Content distribution networks (CDNs)
Content replication
• challenging to stream large files (e.g.,
video) from single origin server in real
time
• solution: replicate content at hundreds of
servers throughout Internet
– content downloaded to CDN servers
ahead of time
– placing content “close” to user avoids
impairments (loss, delay) of sending
content over long paths
– CDN server typically in edge/access
network
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origin server
in North America
CDN distribution node
CDN server
in S. America
Datakommunikation II
CDN server
in Europe
CDN server
in Asia
Katarina Asplund
10
Summary: Internet Multimedia: bag of tricks
• use UDP to avoid TCP congestion control (delays) for timesensitive traffic
• client-side adaptive playout delay: to compensate for delay
• server side matches stream bandwidth to available client-toserver path bandwidth
– chose among pre-encoded stream rates
– dynamic server encoding rate
• error recovery (on top of UDP)
– FEC, interleaving, error concealment
– retransmissions, time permitting
• CDN: bring content closer to clients
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
11
Real-Time Protocol (RTP)
• RTP runs in end systems
• RTP specifies packet
structure for packets
• RTP packets encapsulated
carrying audio, video data
in UDP segments
• RFC 3550
• interoperability: if two
Internet phone
• RTP packet provides
applications run RTP, then
– payload type
they may be able to work
identification
together
– packet sequence
numbering
– time stamping
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
12
RTP and QoS
• RTP does not provide any mechanism to ensure
timely data delivery or other QoS guarantees.
• RTP encapsulation is only seen at end systems
(not) by intermediate routers.
– routers providing best-effort service, making no
special effort to ensure that RTP packets arrive
at destination in timely matter.
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
13
Real-Time Control Protocol (RTCP)
• works in conjunction with RTP. • feedback can be used to
control performance
• each participant in RTP session
– sender may modify its
periodically transmits RTCP
transmissions based on
control packets to all other
feedback
participants.
• each RTCP packet contains
sender and/or receiver reports
– report statistics useful to
application: # packets sent,
# packets lost, interarrival
jitter, etc.
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
14
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 vs regular service)
• granularity: differential
service among multiple
classes, not among
0111
individual connections
• history: ToS bits
Karlstads Universitet
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Principles for QoS guarantees
• Packet marking to distinguish between classes +
new router policy to treat packets differently
• Protect classes from each other (isolation)
• Use resources as efficient as possible
• Call admission (hard guarantees)
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Katarina Asplund
16
Scheduling and Policing mechanisms
• Scheduling
– FIFO
– Priority scheduling
– Weighted fair Queuing (WFQ)
• Policing
– Token bucket to regulate sending rate
Karlstads Universitet
Datavetenskap
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Katarina Asplund
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IETF 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
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
18
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 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?
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
19
Part II - signaling
• Telecom signaling
– SS7
• Voice over IP
– Overview
– SIP
• SCTP
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
20
SIP Services
• Setting up a call, SIP provides
mechanisms ..
– for caller to let callee know she
wants to establish a call
– so caller, callee can agree on
media type, encoding
– to end call
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• determine current IP address of
callee:
– maps mnemonic identifier to
current IP address
– SIP proxy, SIP registrar
• call management:
– add new media streams during
call
– change encoding during call
– invite others
– transfer, hold calls
Datakommunikation II
Katarina Asplund
21
Setting up a call to 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
Bob's
terminal rings
200 OK
.210.89
c=IN IP4 193.64
RTP/AVP 3
m=audio 48753
ACK
port 5060
m Law audio
port 38060
GSM
time
Karlstads Universitet
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indicates her port number, IP
address, encoding she prefers
to receive (PCM ulaw)
Bob’s 200 OK message
indicates his port number, IP
address, preferred encoding
(GSM)

port 5060
port 5060
 Alice’s SIP invite message
port 48753
SIP messages can be sent
over TCP or UDP; here sent over
RTP/UDP.

default
5060.
SIP port number is
time
Datakommunikation II
Katarina Asplund
22
What is SCTP?
• A (relatively) new general purpose transport
protocol
• Main motivation: TCP and UDP are inadequate for
telephony signaling transport
• Provides a reliable message-oriented transport
service
• + a number of functions beneficial for telephony
signaling
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
23
TCP Limitations
• Enforces total ordering of data
– May cause head-of-line blocking
• Is byte-oriented, i.e. does not support framing of
message boundaries
• Has no support for multi-homing
– Difficult to build link or path-level redundancy
• Is vulnerable to denial of service attacks
– Security is often a top priority for phone appl.
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
24
UDP Limitations
• Provides unreliable transport service
• Packets can be lost, duplicated or arrive out-oforder
• Has no congestion control mechanism
• Possible for the application to build its own
mechanism for the above, but…
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
25
Comparing SCTP to TCP - similarities
• Both are connection-oriented, i.e. exchange
messages at startup and closing down
• Both provides reliability through retransmissions,
using either timeouts or fast retransmit
• Both provides for orderly delivery of data (but
SCTP also allows for no or partial ordering)
• Both use the same congestion control mechanism
– Slow start, congestion avoidance (AIMD)
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
26
Comparing SCTP to TCP - differences
• Startup procedure
– Better protection against SYN flooding
• Message abstraction
– Easier buffering and framing for receiving application
• Multi-streaming
– Protection against head-of-line blocking
• Multi-homing
– Better robustness in the presence of network failure
Karlstads Universitet
Datavetenskap
Datakommunikation II
Katarina Asplund
27