Download Voice over Mobile IP

Ch 7. Multimedia Networking
Myungchul Kim
[email protected]
Multimedia and 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.
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Sensitive to end-to-end delay and delay variation
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Streaming stored audio/video
Streaming live audio/video
Real-time interactive audio/video
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Multimedia networking applications
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Examples of multimedia applications
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Streaming stored audio and video
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Streaming live audio and video
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Stored media
Streaming: RealPlayer, QuickTime, Media
Continous playout
Internet radio and IPTV
IP multicasting
Application-layer multicast
Real-time interactive audio and video
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Internet telephony (150 msec)
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Hurdles for multimedia in Today’s Internet
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Best-effor service
How should the Internet evolve to support multimedia
better?
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Hard guarantee vs soft guarnatee
Reservation approach
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Laissez-faire approach
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Protocol
Modification of scheduling policies in the router queues
Description of the application traffic
Available bandwidth in the network
Overprovision bandwidth and switching capacity
Content distribution networks (CDN)
Multicast overlay networks
Differentiated service (Diffserv)
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Audio compression in the Internet
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8,000 samples per second
256 quantization with 8 bits
64Kbps
Pulse code modulation (PCM)
GSM, G.729, G.723.3, MPEG 1 player 3 (MP3)
Video compression in the Internet
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MPEG1, 2, 4
H.261
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Streaming Stored Audio and Video
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Medio player
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Decompression
Jitter removal
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Real-time Streaming Protocol (RTSP)
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Making the best of the best-effort service
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Packet loss
End-to-end delay
Packet jitter
Removing jitter at the receiver for audio
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Sequence number
Timestamp
Delaying playout at the receiver
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Recovering from packet loss
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Content Distribution Networks
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Dimensioning best-effort networks to provide Quality of
Service
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Bandwidth provisioning
Network dimensioning
Models of traffic demand between network end points
Well-defined performance requirements
Workload model
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Protocols for Real-time Interactive
Applications
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RTP
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UDP
RTP header: the type of audio encoding, a sequence number,
and a timestamp
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RTP control protocol (RTCP)
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Using IP multicast
Reports about statistics
Reception report
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SSRC of the RTP streams
The fraction of packets lost
The last sequence number received
The interarrival jitter
Sender report
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The SSRC of the RTP streams
The timestamp and wall clock time of the most recently generated
RTP packet
The number of packets sent
The number of bytes sent
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Session Initiation Protocol (SIP)
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Protocol does
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Establishing calls between a caller and a callee over an IP network
For the caller to determine the current IP address of the callee
Call management
Key characteristics
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Out-of-band protocol
ASCII-readable
All messages to be acknowledged
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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
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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
m=audio 4875
ACK
port 5060
Bob’s 200 OK message
indicates his port number,
IP address, preferred
encoding (GSM)
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SIP messages can be
sent over TCP or UDP;
here sent over RTP/UDP.
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m Law audio
port 38060
GSM
Alice’s SIP invite
message indicates her
port number, IP address,
encoding she prefers to
receive (PCM ulaw)
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port 48753
default
is 5060.
time
time
SIP port number
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Example
Caller [email protected]
with places a
call to [email protected]
SIP registrar
upenn.edu
SIP
registrar
eurecom.fr
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(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 proxy
umass.edu
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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.
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H.323
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Providing multiple classes of service
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Divide traffic into classes and provide different levels of service
to the different classes of traffic.
Differentiated service is provided among aggregates of traffic.
Type-of-service (ToS) in the IPv4
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Scenario 1: a 1 Mbps audio application and an FTP
transfer
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FIFO
Give strict priority to audio packets at R1
Each packet must be marked as belonging to one of these two
classes of traffic, e.g., ToS in IPv4
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Scenario 2: a 1 Mbps audio application and a highpriority FTP transfer
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Packet classification allows a router to distinguish among
packets belonging to different classes of traffic.
A policy decision
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Scenario 3: A misbehaving audio application and an
FTP transfer
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Scheduling and policing mechanisms
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Link-scheduling mechanisms
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First-In-First-Out (FIFO)
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Priority Queueing
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Round robin and weighted fair queueing (WFQ)
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Policing: The Leaky Bucket: regulate the injecting rate of
packets into the networks
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Average rate
Peak rate
Burst size
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Diffserv
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Edge function: packet classification and traffic conditioning: the
diffentiated service field of the packet header
Core function: forwarding, per-hop behavior, aggregation
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Diffserv traffic classfication and conditioning
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Per-hop behaviors
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Differences in performance among classes
Differences in performance observable and measureable
Expedited forwarding, assured forwarding
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Providing quality of service guarantees
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Resouce reservation, call admission, call setup
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Traffic characterization and specification of the desired QoS
Signaling for call setup
Pre-element call admission
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Guaranteed QoS: Intserv and RSVP
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Individualized QoS guarantees
Reservations for bandwidth in multicast trees
Receiver-oriented
Provisioning? Using the policing and scheduling
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