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

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School of Computing ScienceSimon Fraser University

CMPT 820: Multimedia Systems

Introduction

Instructor: Dr. Mohamed Hefeeda

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Course Objectives

Understand fundamentals of networked multimedia systems

Know current research issues in multimedia

Develop research skills

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Course Info

Course web page http://nsl.cs.sfu.ca/teaching/10/820/ References

[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

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Course Info: Grading

Class participation and Assignments: 50% Few assignments and quizzes Present one chapter/paper (important) Read all Mandatory Reading and participate in

discussion Final Project: 50%

New Research Idea (publishable A+) 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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Course Info: Topics

Introduction Overview of the big picture QoS Requirements for Multimedia Systems

QoS in the Network Principles DiffServ and IntServ

Multimedia Protocols RTP, RTSP, RTCP, SIP, …

Image Representation and Compression Sampling, quantization, DCT, compression

Color Models RGB, CMY, YIQ

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Course Info: Topics

Video Coding Compression methods MPEG compression Scalable video coding

Error Control for Video Coding and Transmission Tools for error resilient video coding Error concealment

Internet Characteristics and Impact on Multimedia Channel modeling Internet measurement study

Multimedia Streaming Fundamentals On-demand streaming and live broadcast

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Course Info: Topics

Network-adaptive media transport Rate-distortion optimized streaming

Wireless Multimedia WLANs and QoS

Cross-layer design

QoS Support in mobile operating systems

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Introduction

Motivations

Definitions

QoS Specifications & Requirements

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Definitions and Motivations

“Multimedia” is an overused term 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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Definitions and Motivations

Why should we study/research multimedia topics?

Huge interest and opportunities 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 …

Some video statistics

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

Y-axis in Petabytes (1000 Terabytes) per month.

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2006 2007 2008 2009 2010 2011 20120

2000

4000

6000

8000

10000

12000

14000

Internet Video to PCInternet Video to TVNon-Internet Consumer Video

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Some video statistics

YouTube: fastest growing Internet server in history 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 Had more than 200,000 concurrent clients And at peak time there were 150,000 Waiting

AOL streamed 8 live concerts online in 2006 There were 180,000 clients at peak time

Plus … All major web sites have multimedia

clips/demos/news/broadcasts/…

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Definitions and Motivations

Given all of this, are users satisfied? Not Really!

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!

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Multimedia:The Big Picture [SN04]

QoS in Networked Multimedia Systems

Quality of Service = “well-defined and controllable behavior of a system according to quantitatively measurable parameters”

There are multiple entities in a networked multimedia system User Network Local system (memory, processor, file system,

…)

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QoS in Networked Multimedia Systems

Different parameters belong to different entities QoS Layers

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QoS Layers

User

Application

System

Local Devices Network

Perceptual(e.g., window size, security)

Media Quality(e.g., frame rate, adaptation rules)

Traffic(e.g., bit rate, loss, delay, jitter)

Processing(e.g., CPU scheduling, memory, hard drive)

QoS Layers

QoS Specification Languages 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 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 200018

QoS Layers QoS enforcement methods

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) 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 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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QoS Support in IP Networks

Principles

IntServ

DiffServ

Multimedia Protocols

Reading: Ch. 7 in [KR08]

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QoS in IP Networks: Two Models

Guaranteed QoS Need to reserve resources

Statistical (or Differential) QoS Multiple traffic classes with different priorities

In both models, network devices (routers) should be able to perform certain functions (in addition to forwarding data packets)

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Principles for QoS Guarantees Let us explore these functions using a simple

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

packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly

Principle 1

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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:

provide protection (isolation) for one class from othersPrinciple 2

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Principles for QoS Guarantees (more)

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

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Principles for QoS Guarantees (more)

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

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Summary of QoS Principles

Let’s next look at mechanisms for achieving this ….

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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: who to

discard?• Tail drop: drop arriving packet• priority: drop/remove on priority basis• random: drop/remove randomly

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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..

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Scheduling Policies: still more

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

each cycle

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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)

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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).

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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)

iii

jii

Rbd

wwRR

/

/max

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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:

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IntServ: QoS guarantee scenario

Resource reservation call setup, signaling (RSVP) traffic, QoS declaration per-element admission control

QoS-sensitive scheduling (e.g.,

WFQ)

request/reply

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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

T-spec to routers (where reservation is required) RSVP

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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]

WFQ

token rate, r

bucket size, b

per-flowrate, R

D = b/Rmax

arrivingtraffic

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IETF Differentiated Services

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 require 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

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Edge router: per-flow traffic

management Classifies (marks) pkts

different classes

within a class: 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

DiffServ Architecture

scheduling

...

r

b

marking

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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 A, bucket size B packet marking at edge based on per-flow profile

Possible usage of marking:

User packets

Rate A

B

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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

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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

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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

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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 [Peterson

and Davie 07]

Protocols For Multimedia Applications To manage and stream multimedia data

RTP: Real-Time Protocol

RTSP: Real-Time Streaming Protocol

RTCP: Real-Time Control Protocol

SIP: Session Initiation Protocol

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Real-Time Protocol (RTP): FRC 3550 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

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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 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.

RTP Example

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

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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Real-Time Streaming Protocol (RTSP)

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 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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RTSP: out of band control

FTP uses an “out-of-band” control channel:

file transferred over one TCP connection.

control info (directory changes, file deletion, rename) sent over separate TCP connection

“out-of-band”, “in-band” channels use different port numbers

RTSP messages also sent out-of-band:

RTSP control messages use different port numbers than media stream: out-of-band. port 554

media stream is considered “in-band”.

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RTSP Example

metafile communicated to web browser browser launches player player sets up an RTSP control connection,

data connection to streaming server

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Metafile Example

<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>

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RTSP Operation

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RTSP Exchange Example (simplified) 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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Real-Time Control Protocol (RTCP)

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 feedback54

RTCP - Continued

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

RTCP Packets

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

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Synchronization of Streams

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 Bandwidth Scaling

RTCP attempts to limit its traffic to 5% of session bandwidth.

Example 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

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SIP: Session Initiation Protocol [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

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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

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

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Setting up a call to known IP address Alice’s SIP invite

message 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)

SIP messages can be sent over TCP or UDP; here sent over RTP/UDP.

default SIP port number is 5060.

time time

Bob'stermina l rings

A lice

167.180.112.24

Bob

193.64.210.89

port 38060

Law audio

G SMport 48753

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Setting up a call (more) 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

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Example of SIP message

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 167.180.112.24From: sip:[email protected]: sip:[email protected] Call-ID: [email protected]: application/sdpContent-Length: 885

c=IN IP4 167.180.112.24m=audio 38060 RTP/AVP 0Notes: HTTP message syntax sdp = session description protocol Call-ID is unique for every call.

Here we don’t know Bob’s IP address. Intermediate SIPservers needed.

Alice sends, receives SIP messages using SIP default port 5060

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

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Name translation and user locataion

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

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SIP Registrar

REGISTER sip:domain.com SIP/2.0Via: SIP/2.0/UDP 193.64.210.89 From: sip:[email protected]: sip:[email protected]: 3600

when Bob starts SIP client, client sends SIP REGISTER message to Bob’s registrar server

(similar function needed by Instant Messaging)

Register Message:

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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

proxy analogous to local DNS server

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ExampleCaller [email protected] with places a call to [email protected]

(1) Jim sends INVITEmessage to umass SIPproxy. (2) Proxy forwardsrequest to upenn registrar server. (3) upenn server returnsredirect response,indicating that it should try [email protected]

(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.

SIP client217.123.56.89

SIP client197.87.54.21

SIP proxyum ass.edu

SIP registrarupenn.edu

SIPregistrareurecom .fr

1

2

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5

6

7

8

9

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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, 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.

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Summary: Protocols Several protocols to handle multimedia data RTP: Real-Time Protocol

Packetization, sequence number, time stamp RTSP: Real-Time Streaming Protocol

Establish, Pause, Play, FF, Rewind RTCP: Real-Time Control Protocol

Control and monitor sessions; synchronization SIP: Session Initiation Protocol

Establish and manage VoIP sessions Simpler than the ITU H.323

NONE enforces QoS in the network

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