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School of Computing Science Simon Fraser University CMPT 820: Multimedia Systems Introduction Instructor: Dr. Mohamed Hefeeda. Course Objectives. Understand fundamentals of networked multimedia systems Know current research issues in multimedia Develop research skills . Course Info. - PowerPoint PPT Presentation
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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.11
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: morePriority 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 moreWeighted 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 MechanismsLeaky 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, bper-flowrate, R
D = b/Rmax
arrivingtraffic
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IETF Differentiated ServicesConcerns 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 Conditioningmay 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 controlFTP 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 server50
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 OK53
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'sterminal rings
Alice
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 messageINVITE 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
34
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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