School of Computing Science
Simon Fraser University
CMPT 820: Multimedia Systems
Introduction
Instructor: Dr. Mohamed Hefeeda
1

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Understand fundamentals of networked multimedia systems

Know current research issues in multimedia

Develop research skills
2

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Course web page http://nsl.cs.sfu.ca/teaching/10/820/
References
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[SC07] Schaar and Chou (editors), Multimedia over IP and
Wireless Networks: Compression, Networking, and
Systems, Elsevier, 2007
[Burg09] Burg, The Science of Digital Media, Prentice
Hall, 2009
[KR08] Kurose and Rose, Computer Networking: A top-
down Approach Featuring the Internet, 4th edition,
Addison Wesley, 2008
[LD04] Li and Drew, Fundamentals of Multimedia, Prentice
Hall, 2004

Complemented by research papers
3

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Class participation and Assignments: 50%
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Few assignments and quizzes
Present one chapter/paper (important)
Read all Mandatory Reading and participate in discussion
Final Project: 50%
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New Research Idea (publishable  A+)
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Implementation and evaluation of an already-published algorithm/technique/system (Good demo  A+)
Quantitative comparisons between two already-published algorithms/techniques/systems.
A survey of a multimedia topic
…
Check wiki page for suggestions
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Introduction
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Overview of the big picture
QoS Requirements for Multimedia Systems
QoS in the Network

Principles
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DiffServ and IntServ
Multimedia Protocols

RTP, RTSP, RTCP, SIP, …
Image Representation and Compression
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Sampling, quantization, DCT, compression
Color Models
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RGB, CMY, YIQ
5

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Video Coding
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Compression methods
MPEG compression
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Scalable video coding
Error Control for Video Coding and Transmission

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Tools for error resilient video coding
Error concealment
Internet Characteristics and Impact on Multimedia
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Channel modeling
Internet measurement study
Multimedia Streaming
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Fundamentals
On-demand streaming and live broadcast
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Network-adaptive media transport
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Rate-distortion optimized streaming
Wireless Multimedia
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WLANs and QoS
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Cross-layer design

QoS Support in mobile operating systems
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Motivations
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Definitions
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QoS Specifications & Requirements
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“Multimedia” is an overused term

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Means different things to different people
Because it touches many disciplines/industries
• Computer Science/Engineering
• Telecommunications Industry
• TV and Radio Broadcasting Industry
• Consumer Electronics Industry
• ….
For users

Multimedia = multiple forms/representation of information (text, audio, video, …)
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Why should we study/research multimedia topics?
Huge interest and opportunities
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High speed Networks
Powerful (cheap) computers (desktops … cell phones)
Abundance of multimedia capturing devices (cameras, speakers, …)
Tremendous demand from users (mm content makes life easier, more productive, and more fun)

Here are some statistics …
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Growth of various video traffic [Cisco 2008]

Video traffic accounted for 32% of Internet traffic in
2008 and is estimated to account for 50% in 2012
14000
12000
10000
8000
6000
4000
2000
Internet Video to PC
Internet Video to TV
Non-Internet Consumer
Video

0
2006 2007 2008 2009 2010 2011 2012
Y-axis in Petabytes (1000 Terabytes) per month.
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YouTube : fastest growing Internet server in history
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Serves about 300—400 million downloads per day
Has 40 million videos
Adds 120,000 new videos (uploads) per day
CBS streamed the NCAA March Madness basketball games in 2007 online

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Had more than 200,000 concurrent clients
And at peak time there were 150,000 Waiting
AOL streamed 8 live concerts online in 2006
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There were 180,000 clients at peak time
Plus …
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All major web sites have multimedia clips/demos/news/broadcasts/…
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Given all of this, are users satisfied?
Not Really!
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We still get tiny windows for video
Low quality
Glitches, rebuffering
Limited scalability (same video clip on PDA and desktop)

Server/network outages (capacity limitations)
Users want high-quality multimedia, anywhere, anytime, on any device!
We (researchers) still strive to achieve this vision in the future!
13

14

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Quality of Service = “well-defined and controllable behavior of a system according to quantitatively measurable parameters”
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There are multiple entities in a networked multimedia system
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User
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Network
Local system (memory, processor, file system, …)
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Different parameters belong to different entities  QoS Layers
16

Perceptual
(e.g., window size, security)
Media Quality
(e.g., frame rate, adaptation rules)
Processing
(e.g., CPU scheduling, memory, hard drive)
Traffic
(e.g., bit rate, loss, delay, jitter)
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QoS Specification Languages
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Mostly application specific
XML based
See: Jin & Nahrstedt, QoS Specification Languages for
Distributed Multimedia Applications: A Survey and
Taxonomy, IEEE MultiMedia, 11(3), July 2004
QoS mapping between layers

Map user requirements to Network and Device requirements
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Some (but not all) aspects can be automated
For others, use profiles and rule-of-thumb experience
Several frameworks have been proposed in the literature
See: Nahrstedt et al., Distributed QoS Compilation and
Runtime Instantiation, IWQoS 2000
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QoS enforcement methods
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The most important/challenging aspect
How do we make the network and local devices implement the QoS requirements of MM applications?
We will study (briefly)
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Enforcing QoS in the Network (models/protocols)
Enforcing QoS in the Processor (CPU scheduling for MM)
When we combine them, we get end-to-end QoS
Notice:

This is enforcing application requirements, if the resources are available
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If not enough resources, we have to adapt (or scale) the
MM content (e.g., use smaller resolution, frame rate, drop a layer, etc)
19

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Principles
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IntServ
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DiffServ
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Multimedia Protocols
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Reading: Ch. 7 in [KR08]
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Guaranteed QoS
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Need to reserve resources
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Statistical (or Differential) QoS
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Multiple traffic classes with different priorities
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In both models, network devices (routers) should be able to perform certain functions (in addition to forwarding data packets)
21

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Let us explore these functions using a simple example
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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
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 what if applications misbehave (audio sends higher than declared rate)
 policing: force source adherence to bandwidth allocations marking and policing at network edge:
Principle 2 provide protection (isolation) for one class from others
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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
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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
25

Let’s next look at mechanisms for achieving this ….
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 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: who to discard?
• Tail drop: drop arriving packet
• priority: drop/remove on priority basis
• random: drop/remove randomly
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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..
28

Weighted Fair Queuing :
 generalized Round Robin
 each class gets weighted amount of service in each cycle
29

limit traffic to not exceed declared parameters
Three common-used criteria:
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(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!
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Peak Rate: e.g.,
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Avg rate: 6000 ppm
Peak rate: 1500 ppm
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(Max.) Burst Size: max. number of pkts sent consecutively (with no intervening idle)
30

limit input to specified Burst Size and
Average Rate.
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 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).
31

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Leaky bucket + WFQ  provide guaranteed upper bound on delay, i.e., QoS guarantee! How?
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WFQ: guaranteed share of bandwidth

Leaky bucket: limit max number of packets in queue (burst)
R i d i max

R w i
/
 w j
 b i
/ R i
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 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:
33

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Resource reservation
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 call setup, signaling (RSVP) traffic, QoS declaration per-element admission control
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QoS-sensitive scheduling (e.g.,
WFQ) request/ reply
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 declare its QoS requirement
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R-spec: defines the QoS being requested characterize traffic it will send into network
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T-spec: defines traffic characteristics signaling protocol: needed to carry R-spec and Tspec to routers (where reservation is required)
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RSVP
35

[rfc2211, rfc 2212]
Guaranteed service:
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 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
36

Concerns with IntServ:
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Scalability: signaling, maintaining per-flow router state difficult with large number of flows
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Example: OC-48 (2.5 Gbps) link serving 64 Kbps audio streams  39,000 flows! Each require state maintenance.
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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
37

Edge router:
 per-flow traffic management

Classifies (marks) pkts
 different classes
 within a class: in-profile and out-profile
Core router:

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 per class traffic management buffering and scheduling based on marking at edge preference given to in-profile b r
.
..
38

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 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:
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 class-based marking: packets of different classes marked differently intra-class marking: conforming portion of flow marked differently than non-conforming one
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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
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may be desirable to limit traffic injection rate of some class:
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 user declares traffic profile (e.g., rate, burst size) traffic metered, shaped if non-conforming
41

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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:
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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
42

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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
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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 [Peterson and Davie 07]
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To manage and stream multimedia data
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: Real-Time Protocol
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: Real-Time Streaming Protocol
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: Real-Time Control Protocol
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: Session Initiation Protocol
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RTP specifies packet structure for audio and video data
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 payload type identification packet sequence numbering time stamping
RTP runs in the end systems
RTP packets are encapsulated in
UDP segments
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RTP encapsulation is only seen at the end systems
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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 loss
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.
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 consider 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
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RTP header indicates type of audio encoding in each packet
 sender can change encoding during conference.
RTP header also contains sequence numbers, timestamps.
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RFC 2326 client-server application layer protocol
Used to control a streaming session
 rewind, fast forward, pause, resume, repositioning, etc…
What it doesn’t do:
 doesn’t define how audio/video is encapsulated for streaming over network
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 doesn’t restrict how streamed media is transported
(UDP or TCP possible) doesn’t specify how media player buffers audio/video
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FTP uses an “out-ofband” control channel:
 file transferred over one TCP connection.

 control info (directory changes, file deletion, rename) sent over separate TCP connection
“out-of-band”, “inband” channels use different port numbers
RTSP messages also sent out-of-band:
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RTSP control messages use different port numbers than media stream: out-of-band.
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 port 554 media stream is considered “in-band”.
49

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 metafile communicated to web browser browser launches player player sets up an RTSP control connection, data connection to streaming server
50

<title>Twister</title>
<session>
<group language=en lipsync>
<switch>
<track type=audio e="PCMU/8000/1" src = " rtsp ://audio.example.com/twister/audio.en/lofi">
<track type=audio e="DVI4/16000/2" pt="90 DVI4/8000/1" src=" rtsp ://audio.example.com/twister/audio.en/hifi">
</switch>
<track type="video/jpeg" src=" rtsp ://video.example.com/twister/video">
</group>
</session>
51

52

C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0
Transport: rtp/udp; compression; port=3056; mode=PLAY
S: RTSP/1.0 200 OK
Session 4231
C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0
Session: 4231
Range: npt=0-
C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0
Session: 4231
Range: npt=37
C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0
Session: 4231
S: 200 OK
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Also in RFC 3550 (with RTP)


 works in conjunction with RTP
Allows monitoring of data delivery in a manner scalable to large multicast networks
Provides minimal control and identification functionality each participant in RTP session periodically transmits 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, etc.
used to control performance, e.g., sender may modify its transmissions based on feedback
54

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Each RTP session typically uses 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
55

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
56

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
RTCP can synchronize different media streams within an RTP session consider videoconferencing app for which 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
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RTCP attempts to limit its traffic to 5% of session bandwidth.
Example
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Suppose one sender, sending video at 2 Mbps.
Then RTCP attempts to limit its 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
58

[RFC 3261]
SIP long-term vision:


 all telephone calls, video conference calls take place over Internet people are identified by names or e-mail addresses, rather than by phone numbers you can reach callee, no matter where callee roams, no matter what IP device callee is currently using
59

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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

 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
60

Alice
Bob

Alice’s SIP invite message indicates her port number, IP address, encoding she prefers to receive (PCM ulaw) 167.180.112.24
INVITE bob@193.6
4.210.89
m=audio 38060 RT 12.24
P/AVP 0
193.64.210.89
port 5060 Bob's terminal rings
200 OK c=IN IP4 193.6
4.210.89
m=audio 48753
RTP/AVP 3 port 5060

Bob’s 200 OK message indicates his port number,
IP address, preferred encoding (GSM)
ACK port 5060 port 38060 m
Law audio

SIP messages can be sent over TCP or UDP; here sent over RTP/UDP.
GSM port 48753  default SIP port number is 5060.
time time
61

 codec negotiation:
 suppose Bob doesn’t have PCM ulaw 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
62

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 5060

Alice specifies in header that SIP client sends, receives SIP messages over UDP
63


 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
64

 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:bob@domain.com
To: sip:bob@domain.com
Expires: 3600
65






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.
callee sends response back through the same set of proxies.
proxy returns SIP response message to Alice
 contains Bob’s IP address proxy analogous to local DNS server
66

Caller jim@umass.edu with places a call to keith@upenn.edu
SIP registrar upenn.edu
2
SIP registrar eurecom.fr
(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 keith@eurecom.fr
SIP proxy umass.edu
1
8
SIP client
217.123.56.89
3
4
7
9
6
5
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.
67




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, codecs
SIP is a single component.
Works with RTP, but does not mandate it. Can be combined with other protocols, services



H.323 comes from the ITU
(telephony).
SIP comes from IETF:
Borrows much of its concepts from HTTP

SIP has Web flavor, whereas H.323 has telephony flavor.
SIP uses the KISS principle:
Keep it simple stupid.
68







Several protocols to handle multimedia data
: Real-Time Protocol

Packetization, sequence number, time stamp
: Real-Time Streaming Protocol

Establish, Pause, Play, FF, Rewind
: Real-Time Control Protocol

Control and monitor sessions; synchronization
: Session Initiation Protocol

Establish and manage VoIP sessions

Simpler than the ITU H.323
NONE enforces QoS in the network
69