Unit - 4

Unit-IV:Unit-IV: Link Layer: Introduction, Services, Error Introduction, Services, Error detection & correction techniques, detection & correction techniques, Multiple Access Protocols, LAN address Multiple Access Protocols, LAN address & ARP, CSMA/CD, PPP details& ARP, CSMA/CD, PPP details, Multimedia Multimedia networking and RTSP protocol, RTP details.networking and RTSP protocol, RTP details.

WELCOMEWELCOME

Unit - 4

Some terms:Some terms: hosts and routers are

nodes communication channels

that connect adjacent nodes along communication path are links wired links wireless links LANs

layer-2 packet is a frame, encapsulates datagram

“link”

data-link layer has responsibility of has responsibility of transferring datagram from one node transferring datagram from one node to to adjacentadjacent node over a link node over a link

MessageDatagramSegmentPacketFrame

Unit - 4

tourist = datagram transport segment = communication link transportation mode = link layer protocol travel agent = routing algorithm

tourist = datagram transport segment = communication link transportation mode = link layer protocol travel agent = routing algorithm

Data Link Layer ServicesData Link Layer Services

DLL is responsible for moving frames from one hop (node) to the next.Other responsibilities of DLL are:Framing: DLL provides the stream of bits received from n/w layer into manageable data unit called frames.

Physical Addressing: DLL adds a header to the frame to define the sender or receiver of the frame.

Flow Control: Rate at which receiver is absorbing data is less than produced by the sender. DLL imposes the FC to prevent overwhelming the receiver.

Error Control: DLL adds reliability to PLL by adding mechanism to detect and transmit damaged frames or lost frames. Error control is achieved through a trailer added to end of the fame

DLL is responsible for moving frames from one hop (node) to the next.Other responsibilities of DLL are:Framing: DLL provides the stream of bits received from n/w layer into manageable data unit called frames.

Physical Addressing: DLL adds a header to the frame to define the sender or receiver of the frame.

Flow Control: Rate at which receiver is absorbing data is less than produced by the sender. DLL imposes the FC to prevent overwhelming the receiver.

Error Control: DLL adds reliability to PLL by adding mechanism to detect and transmit damaged frames or lost frames. Error control is achieved through a trailer added to end of the fame

Access Control: When two or more devices are connected to the same link, DLL protocols are necessary to determine which device has control over the link at any given time.

Access Control: When two or more devices are connected to the same link, DLL protocols are necessary to determine which device has control over the link at any given time.

Hop to hop delivery

Unit - 4

Link Layer ServicesLink Layer Services

Framing, link access: encapsulate datagram into frame, adding header, trailer channel access if shared medium “MACMAC” addresses used in frame headers to identify

source, dest • different from IP address!different from IP address!

Reliable delivery between adjacent adjacent nodes we learned how to do this already (chapter 3)! seldom used on low bit error link (fiber, some twisted pair) wireless links: high error rates

• Q: why both link-level and end-end reliability?

48 bit address48 bit address

Unit - 4

Flow Control: pacing between adjacent sending and receiving nodes

Error Detection: errors caused by signal attenuation, noise. receiver detects presence of errors:

• signals sender for retransmission or drops frame

Error Correction: receiver identifies and corrects bit error(s) without

resorting to retransmission

Half-duplex and full-duplex with half duplex, nodes at both ends of link can transmit,

but not at same time

Unit - 4

Adaptors CommunicatingAdaptors Communicating

link layer implemented in “adaptor” (aka NICNIC) Ethernet card, PCMCI card,

802.11 card sending side:

encapsulates datagram in a frame

adds error checking bits, rdt, flow control, etc.

receiving side looks for errors, rdt, flow

control, etc extracts datagram, passes to

rcving node

adapter is semi-autonomoussemi-autonomous link & physical layers

sendingnode

frame

rcvingnode

datagram

frame

adapter adapter

link layer protocol

Unit - 4

Error DetectionError DetectionEDC=EDC= EError DDetection and CCorrection bits (redundancy)DD = Data protected by error checking, may include header fields

• Error detection not Error detection not 100%100% reliable! reliable!• protocol may miss some errors, but rarely• larger EDC field yields better detection and correction

Unit - 4

Parity CheckingParity Checking

Single Bit Parity:Detect single bit errors

Two Dimensional Bit Parity:Detect and correct single bit errors

0 0

Unit - 4

Internet checksumInternet checksum

Goal: detect “errors” (e.g., flipped bits) in transmitted segment (note: used at transport layer only)

Unit - 4

Checksumming: Cyclic Redundancy CheckChecksumming: Cyclic Redundancy Check

view data bits, D, as a binary number choose r+1 bit pattern (generator), G goal: choose r CRC bits, R, such that

<D,R> exactly divisible by G (modulo 2) receiver knows G, divides <D,R> by G. If non-zero

remainder: error detected! can detect all burst errors less than r+1 bits

widely used in practice (ATM, HDCL)

Unit - 4

CRC ExampleCRC Example

Want:

D.2r XORXOR R = nG

equivalently:

D.2r = nG XORXOR R

equivalently:

if we divide D.2r by G, want remainder R

R = remainder[ ]D.2r

G

Unit - 4

Multiple Access Links and ProtocolsMultiple Access Links and ProtocolsTwo types of “links”: point-to-point

PPP for dial-up access point-to-point link between Ethernet switch and

host broadcast (shared wire or medium)

traditional Ethernet upstream HFC 802.11 wireless LAN

Transmission freq is differentdifferent from receiving frequency.

Unit - 4

Slotted ALOHASlotted ALOHA

Assumptions all frames same size time is divided into

equal size slots, time to transmit 1 frame

nodes start to transmit frames only at beginning of slots

nodes are synchronized

if 2 or more nodes transmit in slot, all nodes detect collision

Operation when node obtains fresh

frame, it transmits in next slot

no collision, node can send new frame in next slot

if collision, node retransmits frame in each subsequent slot with prob. p until success

Unit - 4

Slotted ALOHASlotted ALOHA

Pros single active node can

continuously transmit at full rate of channel

highly decentralized: only slots in nodes need to be in sync

simple

Cons collisions, wasting slots idle slots nodes may be able to

detect collision in less than time to transmit packet

clock synchronizationclock synchronization

Unit - 4

Slotted Aloha efficiencySlotted Aloha efficiency

Suppose N nodes with many frames to send, each transmits in slot with probability p

prob that node 1 has success in a slot = p(1-p)N-1

prob that any node has a success = Np(1-p)N-1

For max efficiency with N nodes, find p* that maximizes Np(1-p)N-1

For many nodes, take limit of Np*(1-p*)N-1 as N goes to infinity, gives 1/e = .37

Efficiency is the long-run fraction of successful slots when there are many nodes, each with many frames to send

At best: channelused for useful transmissions 37% of time!

Unit - 4

Pure (unslotted) ALOHAPure (unslotted) ALOHA unslotted Aloha: simpler, no synchronization when frame first arrives

transmit immediately collision probability increases:

frame sent at t0 collides with other frames sent in [t0-1,t0+1]

Unit - 4

CSMACSMA(Carrier Sense Multiple Access)(Carrier Sense Multiple Access)

CSMA:

listen before transmit:If channel sensed idle: transmit entire frame

If channel sensed busy, defer transmission Human analogy: don’t interrupt others!

Unit - 4

CSMA(Carrier Sense Multiple Access)

Medium (channel) is shared by all stations.

And

Only one station at a time can use it.

All stations receive a frame frame sent by a station (broadcast)

The real destination keeps it while others DROP it.

Minimum frame length / Transmission rateMinimum frame length / Transmission rate

is proportional tois proportional to

collision domain / propagation speedcollision domain / propagation speed

Exponential back off policyExponential back off policy

Unit - 4

CSMA collisionsCSMA collisions

collisions can still occur:propagation delay means two nodes may not heareach other’s transmission

collision:entire packet transmission time wasted

spatial layout of nodes

note:role of distance & propagation delay in determining collision probability

Unit - 4

CSMA/CD CSMA/CD (Collision Detection)(Collision Detection)

CSMA/CD: carrier sensing, deferral as in CSMA collisions detected within short time colliding transmissions aborted, reducing channel

wastage collision detection:

easy in wired LANs: measure signal strengths, compare transmitted, received signals

difficult in wireless LANsdifficult in wireless LANs: receiver shut off while transmitting

human analogy: the polite conversationalist

Unit - 4

““Taking Turns” MAC protocolsTaking Turns” MAC protocols

Polling: master node “invites”

slave nodes to transmit in turn

concerns: polling overhead latency single point of failure

(master)

Token passing: control token passed from

one node to next sequentially. token message concerns:

token overhead latency single point of failure (token)

Unit - 4

Summary of MAC protocolsSummary of MAC protocols What do you do with a shared media?What do you do with a shared media?

Channel Partitioning, by time, frequency or codeChannel Partitioning, by time, frequency or code• Time Division, Frequency Division

Random partitioning (dynamic), Random partitioning (dynamic), • ALOHA, S-ALOHA, CSMA, CSMA/CD• carrier sensing: easy in some technologies (wire),

hard in others (wireless)• CSMA/CD used in Ethernet• CSMA/CA used in 802.11

Taking TurnsTaking Turns• polling from a central site, token passing

Unit - 4

What Is a MACMAC Address?The MAC address is a unique value associated with a network adapter. MAC addresses are also known as hardware addresseshardware addresses or physical addressesphysical addresses. They uniquely identify an adapter on a LAN.They uniquely identify an adapter on a LAN.

MAC addresses are 12-digit hexadecimal numbers(48 bits in length).

MAC addresses are usually written in one of the following two formats:MM:MM:MM:SS:SS:SS ororMM-MM-MM-SS-SS-SS

The first half of a MAC addressThe first half of a MAC address contains the ID number of the adapter manufacturer. (vendor)The second half of a MAC addressThe second half of a MAC address represents the serial number assigned to the adapter by the manufacturer.

C:>ipconfigC:>ipconfig

$ifconfig$ifconfig

Unit - 4

MAC AddressesMAC Addresses and ARP and ARP

32-bit IP address: network-layer address used to get datagram to destination IP subnet

48-bit MAC address: (or LAN or physical or Ethernet)

used to get datagram from one interface to another physically-connected interface (same network)

48 bit MAC address (for most LANs) burned in the adapter ROM

Logical AddressPhysical Address

Unit - 4

Learn aboutCommands

arpnetstatroute ipconfig

etc…

Unit - 4

LAN Addresses and ARPLAN Addresses and ARP

Each adapter on LAN has unique LAN address

Broadcast address =FF-FF-FF-FF-FF-FF

= adapter

1A-2F-BB-76-09-AD

58-23-D7-FA-20-B0

0C-C4-11-6F-E3-98

71-65-F7-2B-08-53

LAN(wired orwireless)

Hexadecimal

Unit - 4

MAC address allocation administered by IEEE manufacturer buys portion of MAC address space

(to assure uniqueness) Analogy:

(a) MAC address: like Social Security Number

(b) IP address: like postal address MAC flat address ➜ portability

can move LAN card from one LAN to another

IP hierarchical address NOT portable depends on IP subnet to which node is attached

LAN Addresses

Unit - 4

ARPARPAddress Resolution ProtocolAddress Resolution Protocol

Each IP node (Host, Router) on LAN has ARP table

ARP Table: IP/MAC address mappings for some LAN nodes

< IP address; MAC address; TTL> TTL (Time To Live): time

after which address mapping will be forgotten (typically 20 min)

Question: how to determineMAC address of Bknowing B’s IP address?

1A-2F-BB-76-09-AD

58-23-D7-FA-20-B0

0C-C4-11-6F-E3-98

71-65-F7-2B-08-53

LAN

237.196.7.23

237.196.7.78

237.196.7.14

237.196.7.88

Unit - 4

A wants to send datagram to B, and B’s MAC address not in A’s ARP table.

A broadcasts ARP query packet, containing B's IP address Dest MAC address = FF-

FF-FF-FF-FF-FF all machines on LAN

receive ARP query B receives ARP packet,

replies to A with its (B's) MAC address frame sent to A’s MAC

address (unicast)

A caches (saves) IP-to-MAC address pair in its ARP table until information becomes old (times out) soft state: information

that times out (goes away) unless refreshed

ARP is “plug-and-play”: nodes create their ARP

tables without intervention from net administrator

ARPARPAddress Resolution Protocol

Unit - 4

Routing to another LANRouting to another LAN

walkthrough: send datagram from A to B via R assume A know’s B IP address

Two ARP tables in router R, one for each IP network (LAN)

In routing table at source Host, find router 111.111.111.110 In ARP table at source, find MAC address E6-E9-00-17-BB-4B, etc

A

RB

Unit - 4

A creates datagram with source A, destination B A uses ARP to get R’s MAC address for 111.111.111.110A uses ARP to get R’s MAC address for 111.111.111.110 A creates link-layer frame with R's MAC address as dest,

frame contains A-to-B IP datagram A’s adapter sends frame R’s adapter receives frame R removes IP datagram from Ethernet frame, sees its R removes IP datagram from Ethernet frame, sees its

destined to Bdestined to B R uses ARP to get B’s MAC address R creates frame containing A-to-B IP datagram sends to B

A

RB

Unit - 4

EthernetEthernet“dominant” wired LAN technology: cheap $20 for 100Mbs! first widely used LAN technology Simpler, cheaper than token LANs and ATM Kept up with speed race: 10 Mbps – 10

Gbps

Metcalfe’s Ethernetsketch

Read his inspiring Interview At the end of chapter

Unit - 4

Ethernet

The Preambleconsists of seven bytes all of the form 10101010, and is used by the receiver to allow it to establish bit

synchronisation (there is no clocking information on the Ether when nothing is being sent).

The Start frame delimiteris a single byte, 10101011, which is a frame flag, indicating the start of a frame.

The MAC addressesare always 48 bits long

The Length/EtherType field is the only one which differs between 802.3 and Ethernet II. In 802.3 it indicates the number of bytes of data in the frames payload, and can be anything from 0 to 1500

bytes.

Frames must be at least 64 bytes long, not including the preamble, so, if the data field is shorter than 46 bytes, it must be compensated by the Pad field.Pad field.

The reason for specifying a minimum length lies with the collision-detect mechanism. In CSMA/CD a station must never be allowed to believe it has transmitted a frame successfully if that frame has, in fact, experienced a collision.

Unit - 4

Ethernet uses Ethernet uses CSMA/CDCSMA/CD

No slots adapter doesn’t transmit if

it senses that some other adapter is transmitting, that is, carrier sense

transmitting adapter aborts when it senses that another adapter is transmitting, that is, collision detection

Before attempting a retransmission, adapter waits a random time, that is, random access

Unit - 4

CSMA/CD efficiencyCSMA/CD efficiency

Tprop = max prop between 2 nodes in LAN ttrans = time to transmit max-size frame

Efficiency goes to 1 as tprop goes to 0 Goes to 1 as ttrans goes to infinity Much better than ALOHA, but still decentralized,

simple, and cheap

transprop tt /51

1efficiency

Unit - 4

10BaseT and 100BaseT10BaseT and 100BaseT 10/100 Mbps rate; latter called “fast ethernet” T stands for Twisted Pair Nodes connect to a hub: “star topology”; 100

m max distance between nodes and hub

twisted pair

hub

Unit - 4

HubsHubsHubs are essentially physical-layer repeaters:

bits coming from one link go out all other links at the same rate no frame buffering no CSMA/CD at hub: adapters detect collisions provides net management functionality

twisted pair

hub

Unit - 4

Manchester encodingManchester encoding

Used in 10BaseT Each bit has a transition Allows clocks in sending and receiving nodes to

synchronize to each other no need for a centralized, global clock among nodes!

Hey, this is physical-layer stuff!

RLLRLL??

encodingencoding??

Unit - 4

Gbit EthernetGbit Ethernet

uses standard Ethernet frame format allows for point-to-point links and shared broadcast

channels in shared mode, CSMA/CD is used; short distances

between nodes required for efficiency uses hubs, called here “Buffered Distributors” Full-Duplex at 1 Gbps for point-to-point links 10 Gbps now !

Unit - 4

Interconnecting with hubsInterconnecting with hubs Backbone hub interconnects LAN segments Extends max distance between nodes But individual segment collision domains become one large

collision domain Can’t interconnect 10BaseT & 100BaseT

hub

hubhub

hub

Unit - 4

SwitchSwitch

Link layer devicestores and forwards Ethernet framesexamines frame header and selectively

forwards frame based on MAC dest addresswhen frame is to be forwarded on segment,

uses CSMA/CD to access segment transparent

hosts are unaware of presence of switchesplug-and-play, self-learning

switches do not need to be configured

Unit - 4

ForwardingForwarding

• How do determine onto which LAN segment to How do determine onto which LAN segment to forward frame?forward frame?• Looks like a routing problem...

hub

hubhub

switch1

2 3

Unit - 4

Self learningSelf learning

A switch has a switch table entry in switch table:

(MAC Address, Interface, Time Stamp) stale entries in table dropped (TTL can be 60 min)

switch learns which hosts can be reached through which interfaces when frame received, switch “learns” location of

sender: incoming LAN segment records sender/location pair in switch table

Unit - 4

Filtering/ForwardingFiltering/Forwarding

When switch receives a frame:

index switch table using MAC dest addressif entry found for destination

then{ if dest on segment from which frame arrived

then drop the frame else forward the frame on interface indicated } else flood

forward on all but the interface on which the frame arrived

Unit - 4

Switch exampleSwitch example

Suppose C sends frame to D

Switch receives frame from from C notes in bridge table that C is on interface 1 because D is not in table, switch forwards frame into interfaces

2 and 3

frame received by D

hub

hub hub

switch

A

B CD

EF

G H

I

address interface

ABEG

1123

12 3

Unit - 4

Suppose D replies back with frame to C.

Switch receives frame from from D notes in bridge table that D is on interface 2 because C is in table, switch forwards frame only to interface 1

frame received by C

hub

hub hub

switch

A

B CD

EF

G H

I

address interface

ABEGC

11231

Switch exampleSwitch example

Unit - 4

Switch: traffic isolationSwitch: traffic isolation switch installation breaks subnet into LAN segments switch filters packets:

same-LAN-segment frames not usually forwarded onto other LAN segments

segments become separate collision domains

hub hub hub

switch

collision domain collision domain

collision domain

Unit - 4

Institutional networkInstitutional network

hub

hubhub

switch

to externalnetwork

router

IP subnet

mail server

web server

Unit - 4

WANWAN LANLAN

n e t l a b n e t l a b I N F R A S T R U C T U R E I N F R A S T R U C T U R E

Atire WireSpan 5000 Modem V.35

Atire WireSpan 5000 SERVER ROOM

DATA CENTER

NetLinkG.703/G.7042701 PATTONModemBSNL TOWER

SHEGAONSSGMCE

BSNLKHAMGAON

OFC (FIBER OPTIC CABLE)Distance 20 KM

RASCISCO 5300

CISCO 7500

CISCO 2620ROUTER

42U Switch & Server Racks SSGMCE

CAMPUS

1 2 3 4 5 61 2 3 4 5 6

1]1] Traffic Monitoring Server2]2] DNS (Secondary NS2)

Server3]3] Proxy & Dial-in Server4]4] WWW & POP Server5]5] DNS (Primary NS1) & Mail6]6] DHCP+FTP Server

T&PEPABXDIAL INSERVICE

GMSENGLIS

HSCHOO

L

Unit - 4

n e t l a b n e t l a b I N F R A S T R U C T U R E I N F R A S T R U C T U R E

DWL-2100APD-Link

Atire WireSpan 5000 Modem V.35

Atire WireSpan 5000

EXTCMECHSAP-LAB

SGIRCNEW ADMIN BUILDING

ELECT

SM1LADIESHOSTEL

MBADEPARTMENT

SVBOYSHOSTEL

GMSENGSCHOOL

NetLinkG.703/G.7042701 PATTONModem

RASCISCO 5300

CISCO 7500

BSNL TOWERSHEGAONSSGMCE

BSNL KHAMGAON

T&PEPABXDIAL INSERVICE

DWL-2100APD-Link

Wi-FiFacility

6 CORE FIBER

UTP

Layer II/III switch

OFCSWITCH/ SERVER RACK

SERVER ROOMDATA CENTER

CENTRALLIBRARYBUILDING2000 MTRS FIBER 2000 MTRS FIBER

COVERS EACH CORNER OFCOVERS EACH CORNER OFSSGMCESSGMCE

N

E

Unit - 4

MBA DEPT12U-Rack

192.168.254.10

ELECTRONICS DEPARTMENT

12U-Rack

3300TM (24)

ELECTRICAL DEPARTMENT

9U-Rack

D-Link 1024D (24)

42U- RACK

3COM3300MM (24)

3COM 4050 (12)

3COM3300XM (24)

3COM3300MM (24)

3COM3300TM (24)

3COM3300XM (24)

3COM3300XM (24)

3COM3300XM (24)

3300SM (24)

CENTRALLIBRARY9U-Rack

MECHANICAL& GEN ENGG

DEPARTMENT12U-Rack

3300TM (24)

4226 (24)

4226 (24)

4226 (24)

Cisco2950 (12)

DES 1016D (16)

6 Core OFC 1000BASE-SXCAT-5

SERVER ROOM42U-RackCISCO 2620

192.168.254.14

192.168.254.16

192.168.254.14

192.168.254.12

192.168.254.17

192.168.254.18

192.168.254.13

4226

4226

4226

4228T

4226 (24)192.168.254.47

GIRLS HOSTEL12U-Rack

SV BOYS HOSTEL12U-Rack

GMS SHCOOL12U-Rack DETAILEDDETAILED

VIEWVIEW

Unit - 4

DEV QAS PRD

All Rack Mounted Servers

SAP ROUTER

SSGMCE CampusBuilding roof GMS

Building roof

Arial Distance 3 +/- KM

EXTERNAL WORLD

INTERNET

SSGMCESSGMCEWANWAN

SSGMCE INTRANET

LAN-1

N E T L A Bdata center

GMSBuildingComplex

GMS/SSGMCEGMS/SSGMCE INTRANETINTRANET

LAN-2LAN-2

TH

RE

E S

YS

TE

M S

AP

R/3

TH

RE

E S

YS

TE

M S

AP

R/3

LA

ND

SC

AP

EL

AN

DS

CA

PE

Motorola Canopy (Dish)Motorola Canopy (Dish)RF Point to Point Access PointRF Point to Point Access Point

Unit - 4

DEV QAS PRD

All Rack Mounted Servers

SSGMCE CampusBuilding roof

SAP ROUTER

3 KM

INTERNET

SSGMCEWAN

SSGMCE INTRANE

T

ANAND SAGAR

OTHERSCHOOLS

ANAND SAGAR OFFICE

GMSGMS

NEXT PHASEPROJECTPROJECTEXPANSION PLANEXPANSION PLAN

Remote Branches

ALANDI

Remote Branches

PANDHARPUR

Remote Branches

TRAMBK’WAR

Remote Branches

OMKARESHWAR

Unit - 4

Switches vs. RoutersSwitches vs. Routers both store-and-forward devices

routers: network layer devices (examine network layer headers) switches are link layer devices

routers maintain routing tables, implement routing algorithms switches maintain switch tables, implement filtering, learning

algorithms

Unit - 4

Summary comparisonSummary comparison

hubs routers switches

traffi c isolation

no yes yes

plug & play yes no yes

optimal routing

no yes no

cut through

yes no yes

Unit - 4

Point to Point Data Link ControlPoint to Point Data Link Control

one sender, one receiver, one link: easier than one sender, one receiver, one link: easier than broadcast link:broadcast link: no Media Access Control no need for explicit MAC addressing e.g., dialup link, ISDN line

popular point-to-point DLC protocols: PPPPPP (point-to-point protocol) HDLCHDLC: High level data link control (Data link used

to be considered “high layer” in protocol stack!

High-Level Data Link ControlHigh-Level Data Link Control

Unit - 4

PPP Design Requirements [RFC 1557]PPP Design Requirements [RFC 1557]

packet framing: encapsulation of network-layer encapsulation of network-layer datagram in data link framedatagram in data link frame carry network layer data of any network layer

protocol (not just IP) at same time ability to demultiplex upwards

bit transparency: must carry any bit pattern in the data field

error detection (no correction) connection liveness: detect, signal link failure to

network layer network layer address negotiation: endpoint can

learn/configure each other’s network address

Unit - 4

PPP non-requirementsPPP non-requirements

no error correction/recovery no flow control out of order delivery OK no need to support multipoint links (e.g., polling)

Error recovery, flow control, data re-ordering Error recovery, flow control, data re-ordering all relegated to all relegated to higher layershigher layers!!

Unit - 4

PPP Data FramePPP Data Frame

Flag: delimiter (framing) Address: does nothing (only one option) Control: does nothing; in the future possible multiple

control fields Protocol: upper layer protocol to which frame

delivered (eg, PPP-LCP, IP, IPCP, etc)

Unit - 4

PPP Data FramePPP Data Frame

info: upper layer data being carried check: cyclic redundancy check for error

detection

Unit - 4

Byte StuffingByte Stuffing “data transparency” requirement: data field must be

allowed to include flag pattern <01111110> Q: is received <01111110> data or flag?

Sender: adds (“stuffs”) extra < 01111110> byte after each < 01111110> data byte

Receiver: two 01111110 bytes in a row: discard first byte,

continue data reception single 01111110: flag byte

Unit - 4

Byte StuffingByte Stuffing

flag bytepatternin datato send

flag byte pattern plusstuffed byte in transmitted data

Unit - 4

PPP Data Control ProtocolPPP Data Control Protocol

Before exchanging network-layer data, data link peers must

configure PPP link (max. frame length, authenticationauthentication)

learn/configure network layer information

for IP: carry IP Control Protocol (IPCP) msgs (protocol field: 8021) to configure/learn IP address

Unit - 4

Unit - 4

MultimediaMultimedia

In past, we listened to audio/video broadcast through a radio or TV transmission.

But now time has changedAudio and Video services are divided into 3 categoriesAudio and Video services are divided into 3 categories

1. Streaming stored audio/video2. Streaming live audio/video and3. Interactive stored audio/video

Streaming –Streaming –Means a user can listen (or watch) the file after downloading has stored.

Unit - 4

Multimedia, Quality of Service: Multimedia, Quality of Service: What is it?What is it?

Multimedia applications: network audio and video(“continuous media”)

network provides application with level of level of performance needed for performance needed for application to function.application to function.

QoS

Unit - 4

MM Networking Applications MM Networking Applications

Fundamental Fundamental characteristics:characteristics:

Typically delay sensitive end-to-end delay delay jitter

But loss tolerant: infrequent losses cause minor glitches

Antithesis of data, which are loss intolerant but delay tolerant.

Classes of MM Classes of MM applications:applications:

1) Streaming stored audio and video

2) Streaming live audio and video

3) Real-time interactive audio and video

JitterJitter is the variability of packet delays within the same packet stream

Unit - 4

Streaming Stored Multimedia Streaming Stored Multimedia

Streaming: media stored at source transmitted to client streamingstreaming: client playout begins

before all data has arrived

timing constraint for still-to-be transmitted data: in time for playout

Unit - 4

StreamingStreaming Stored Multimedia: What is it? Stored Multimedia: What is it?

1. videorecorded

2. videosent

3. video received,played out at client

Cum

ula

tive

data

streaming:streaming: at this time, client playing out early part of video, while server still sending laterpart of video

networkdelay

time

Unit - 4

StreamingStreaming Stored Multimedia: Interactivity Stored Multimedia: Interactivity

VCR-like functionality: client can pause, rewind, FF, push slider bar 10 sec initial delay OK 1-2 sec until command effect OK RTSP often used (more later)

timing constraint for still-to-be transmitted data: in time for playout

Unit - 4

Streaming Live MultimediaStreaming Live Multimedia

Examples: Internet radio talk show Live sporting event

Streaming playback buffer playback can lag tens of seconds after transmission still have timing constraint

Interactivity fast forward impossible rewind, pause possible!

Unit - 4

Multimedia Over Today’s InternetMultimedia Over Today’s Internet

TCP/UDP/IP: “best-effort service” no guarantees on delay, loss

Today’s Internet multimedia applications use application-level techniques to mitigate

(as best possible) effects of delay, loss

But you said multimedia apps requiresQoS and level of performance to be

effective!

?? ???

?

? ??

?

?

Unit - 4

about about audio compressionaudio compression

Analog signal sampled at constant rate telephone: 8,000

samples/sec CD music: 44,100

samples/sec Each sample quantized,

i.e., rounded e.g., 28=256 possible

quantized values Each quantized value

represented by bits 8 bits for 256 values

Example: 8,000 samples/sec, 256 quantized values --> 64,000 bps

Receiver converts it back to analog signal: some quality reduction

Example ratesExample rates CD: 1.411 Mbps MP3: 96, 128, 160

kbps Internet telephony:

5.3 - 13 kbps

Unit - 4

about video compressionabout video compression

Video is sequence of images displayed at constant rate e.g. 24 images/sec

Digital image is array of pixels

Each pixel represented by bits

Redundancy spatial temporal

Examples: MPEG 1 (CD-ROM)

1.5 Mbps MPEG2 (DVD) 3-6

Mbps MPEG4 (often used in

Internet, < 1 Mbps)

Unit - 4

Streaming Multimedia: Streaming Multimedia: UDP UDP oror TCP 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 compensate for network delay 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

Unit - 4

RTSP RTSP

HTTP Does not target multimedia

content No commands for fast

forward, etc.

RTSP: RFC 2326 Client-server application

layer protocol. For user to control display:

rewind, fast forward, pause, resume, repositioning, etc…

What it doesn’t do: does not define how does not define how

audio/video is audio/video is encapsulated for encapsulated for streaming over networkstreaming over network

does not restrict how streamed media is transported; it can be transported over UDP or TCP

does not specify how the does not specify how the media player buffers media player buffers audio/videoaudio/video

Unit - 4

Using a Media Server and RTSPRTSP

The real-time streaming Protocol (RTSPRTSP):

Is control protocol designed to add more functionalities to the streaming process.

Using RTSP we can control the playing of audio/video.

RTSP is a out-of-band control protocol that is similar to the second connection in FTP.

Unit - 4

Using a Media Server and RTSPRTSP

Browser

Client Machine Server Machine

Web Server

1

2

GET: metafile

Response

MediaPlayer

3

MediaServer

4 SETUP

5 RESPONSE

6 PLAY

7 RESPONSE

8 TEARDOWN

9 RESPONSE

Unit - 4

RTSP: out of band controlRTSP: out of band controlFTP uses an “out-of-band”

control channel: A file is transferred over

one TCP connection. Control information

(directory changes, file deletion, file renaming, etc.) is sent over a separate TCP connection.

The “out-of-band” and “in-band” channels use different port numbers.

RTSP messages are also sent out-of-band:

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

The media stream is considered “in-band”.

Unit - 4

RTSP ExampleRTSP Example

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

data connection to streaming server

Unit - 4

Metafile ExampleMetafile 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>

Unit - 4

RTSP OperationRTSP Operation

Unit - 4

RTSP Exchange ExampleRTSP Exchange Example C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Transport: rtp/udp; compression; port=3056; mode=PLAY

S: RTSP/1.0 200 1 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 3 OK

Unit - 4

Real-time interactive Real-time interactive applicationsapplications

PC-2-PC phone instant messaging services

are providing this PC-2-phone

Dialpad Net2phone

videoconference with Webcams

Going to now look at a PC-2-PC Internet phone example in detail

Unit - 4

InteractiveInteractive Multimedia: Internet Multimedia: Internet PhonePhone

Introduce Internet Phone by way of an example

speaker’s audio: alternating talk spurts, silent periods. 64 kbps during talk spurt

pkts generated only during talk spurts 20 msec chunks at 8 Kbytes/sec: 160 bytes data

application-layer header added to each chunk.

Chunk+header encapsulated into UDP segment.

application sends UDP segment into socket every 20 msec during talkspurt.

Unit - 4

Internet Phone: Packet Loss and DelayInternet Phone: Packet Loss and Delay

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.

Unit - 4

constant bit ratetransmission

Cum

ula

tive

data

time

variablenetwork

delay(jitter)

clientreception

constant bit rate playout at client

client playoutdelay

bu

ffere

ddata

Delay JitterDelay Jitter

Consider the end-to-end delays of two consecutive packets: difference can be more or less than 20 msec

Unit - 4

Internet Phone: Fixed Playout DelayInternet Phone: Fixed Playout Delay

Receiver attempts to playout each chunk exactly q msecs after chunk was generated.chunk has time stamp t: play out chunk at

t+q .chunk arrives after t+q: data arrives too

late for playout, data “lost” Tradeoff for q:

large q: less packet losssmall q: better interactive experience

Unit - 4

Fixed Playout DelayFixed Playout Delay

packets

tim e

packetsgenerated

packetsreceived

loss

r

p p '

playout schedulep' - r

playout schedulep - r

• Sender generates packets every 20 msec during talk spurt.• First packet received at time r• First playout schedule: begins at p• Second playout schedule: begins at p’

Unit - 4

Adaptive Playout Delay, IAdaptive Playout Delay, I

packetith receivingafter delay network average of estimated

acketpith for delay network tr

receiverat played is ipacket timethep

receiverby received is ipacket timether

packetith theof timestampt

i

ii

i

i

i

Dynamic estimate of average delay at receiver:

)()1( 1 iiii trudud

where u is a fixed constant (e.g., u = .01).

Goal: minimize playout delay, keeping late loss rate low Approach: adaptive playout delay adjustment:

Estimate network delay, adjust playout delay at beginning of each talk spurt.

Silent periods compressed and elongated. Chunks still played out every 20 msec during talk spurt.

Unit - 4

Adaptive playout delay IIAdaptive playout delay II

Also useful to estimate the average deviation of the delay, vi :

||)1( 1 iiiii dtruvuv

The estimates di and vi are calculated for every received packet, although they are only used at the beginning of a talk spurt.

For first packet in talk spurt, playout time is:

iiii Kvdtp

where K is a positive constant.

Remaining packets in talkspurt are played out periodically

Unit - 4

Adaptive Playout, IIIAdaptive Playout, III

Q: How does receiver determine whether packet is first in a talkspurt?

If no loss, receiver looks at successive timestamps. difference of successive stamps > 20 msec -->talk

spurt begins. With loss possible, receiver must look at both

time stamps and sequence numbers. difference of successive stamps > 20 msec and

sequence numbers without gaps --> talk spurt begins.

Unit - 4

Recovery from packet loss (1)Recovery from packet loss (1)

forward error correction (FEC): simple scheme

for every group of n chunks create a redundant chunk by exclusive OR-ing the n original chunks

send out n+1 chunks, increasing the bandwidth by factor 1/n.

can reconstruct the original n chunks if there is at most one lost chunk from the n+1 chunks

Playout delay needs to be fixed to the time to receive all n+1 packets

Tradeoff: increase n, less

bandwidth waste increase n, longer

playout delay increase n, higher

probability that 2 or more chunks will be lost

Unit - 4

Recovery from packet loss (2)Recovery from packet loss (2)2nd FEC scheme• “piggyback lower quality stream” • send lower resolutionaudio stream as theredundant information• for example, nominal stream PCM at 64 kbpsand redundant streamGSM at 13 kbps.

• Whenever there is non-consecutive loss, thereceiver can conceal the loss. • Can also append (n-1)st and (n-2)nd low-bit ratechunk

Unit - 4

Recovery from packet loss (3)Recovery from packet loss (3)

Interleaving chunks are broken

up into smaller units for example, 4 5 msec units

per chunk Packet contains small units

from different chunks

if packet is lost, still have most of every chunk

has no redundancy overhead

but adds to playout delay

Unit - 4

Summary: Summary: Internet Multimedia: bag of Internet Multimedia: bag of trickstricks

use UDP to avoid TCP congestion control (delays) for time-sensitive traffic

client-side adaptive playout delay: to compensate for delay

server side matches stream bandwidth to available client-to-server path bandwidth chose among pre-encoded stream rates dynamic server encoding rate

error recovery (on top of UDP) FEC, interleaving retransmissions, time permitting conceal errors: repeat nearby data

Unit - 4

Real-Time Protocol (RTP)Real-Time Protocol (RTP)

RTP specifies a packet structure for packets carrying audio and video data

RFC 1889. RTP packet provides

payload type identification

packet sequence numbering

timestamping

RTP runs in the end systems.

RTP packets are encapsulated in UDP segments

Interoperability: If two Internet phone applications run RTP, then they may be able to work together

Unit - 4

RTP runs on top of UDPRTP runs on top of UDP

RTP libraries provide a transport-layer interface that extend UDP:

• port numbers, IP addresses• payload type identification• packet sequence numbering• time-stamping

Unit - 4

RTP ExampleRTP Example Consider sending 64

kbps PCM-encoded voice over RTP.

Application collects the encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk.

The audio chunk along with the RTP header form the RTP packet, which is encapsulated into a UDP segment.

RTP header indicates type of audio encoding in each packet sender can change

encoding during a conference.

RTP header also contains sequence numbers and timestamps.

Unit - 4

RTP and QoSRTP and QoS

RTP does not provide any mechanism to ensure timely delivery of data or provide other quality of service guarantees.

RTP encapsulation is only seen at the end systems: it is not seen by intermediate routers. Routers providing best-effort service do not

make any special effort to ensure that RTP packets arrive at the destination in a timely matter.

Unit - 4

RTP HeaderRTP Header

Payload Type (7 bits): Indicates type of encoding currently being used. If sender changes encoding in middle of conference, sender informs the receiver through this payload type field.

•Payload type 0: PCM mu-law, 64 kbps•Payload type 3, GSM, 13 kbps•Payload type 7, LPC, 2.4 kbps•Payload type 26, Motion JPEG•Payload type 31. H.261•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 and to restore packet sequence.

Unit - 4

RTP Header (2)RTP Header (2)

Timestamp field (32 bytes long). Reflects the sampling instant of the first byte in the RTP data packet. For audio, timestamp clock typically increments by

one for each sampling period (for example, each 125 usecs for a 8 KHz sampling clock)

if application generates chunks of 160 encoded samples, then timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive.

SSRC field (32 bits long). Identifies the source of the RTP stream. Each stream in a RTP session should have a distinct SSRC.

Unit - 4

RTSP/RTP Programming RTSP/RTP Programming AssignmentAssignment

Build a server that encapsulates stored video frames into RTP packets grab video frame, add RTP headers, create

UDP segments, send segments to UDP socket

include seq numbers and time stamps client RTP provided for you

Also write the client side of RTSP issue play and pause commands server RTSP provided for you

Unit - 4

Real-Time Control Protocol (RTCP)Real-Time Control Protocol (RTCP)

Works in conjunction with RTP.

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

Statistics include number of packets sent, number of packets lost, interarrival jitter, etc.

Feedback can be used to control performance Sender may modify

its transmissions based on feedback

Unit - 4

RTCP - ContinuedRTCP - Continued

- For an RTP session there is typically a single multicast address; all RTP and RTCP packets belonging to the session use the multicast address.

- RTP and RTCP packets are distinguished from each other through the use of distinct port numbers.

- To limit traffic, each participant reduces his RTCP traffic as the number of conference participants increases.

Unit - 4

RTCP PacketsRTCP Packets

Receiver report packets: fraction of packets

lost, last sequence number, average interarrival jitter.

Sender report packets: SSRC of the RTP

stream, the current time, the number of packets sent, and the 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.

Unit - 4

Synchronization of StreamsSynchronization of Streams

RTCP can synchronize different media streams within a RTP session.

Consider videoconferencing app for which each sender generates one RTP stream for video and one for audio.

Timestamps in RTP packets tied to the video and audio sampling clocks not tied to the wall-

clock time

Each RTCP sender-report packet contains (for the most recently generated packet in the associated RTP stream): timestamp of the RTP

packet wall-clock time for when

packet was created. Receivers can use this

association to synchronize the playout of audio and video.

Unit - 4

RTCP Bandwidth ScalingRTCP Bandwidth Scaling

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

Example Suppose one sender,

sending video at a rate of 2 Mbps. Then RTCP attempts to limit its traffic to 100 Kbps.

RTCP gives 75% of this rate to the receivers; remaining 25% to the sender

The 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 the entire session) and dividing by allocated rate.

Unit - 4

SIPSIP

Session Initiation Protocol Comes from IETFSIP long-term vision All telephone calls and video conference

calls take place over the Internet People are identified by names or e-mail

addresses, rather than by phone numbers.

You can reach the callee, no matter where the callee roams, no matter what IP device the callee is currently using.

Unit - 4

SIP ServicesSIP Services

Setting up a call Provides mechanisms

for caller to let callee know she wants to establish a call

Provides mechanisms so that caller and callee can agree on media type and encoding.

Provides mechanisms 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 and hold

calls

Unit - 4

Setting up a call to a known IP Setting up a call to a known IP addressaddress

• Alice’s SIP invite message indicates her port number & IP address. Indicates encoding that Alice 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

Unit - 4

Setting up a call (more)Setting up a call (more) Codec negotiation:

Suppose Bob doesn’t have PCM ulaw encoder.

Bob will instead reply with 606 Not Acceptable Reply and list encoders he can use.

Alice can then send a new INVITE message, advertising an appropriate encoder.

Rejecting the call Bob can reject

with replies “busy,” “gone,” “payment required,” “forbidden”.

Media can be sent over RTP or some other protocol.

Unit - 4

Example of SIP messageExample of SIP message

INVITE sip:bob@domain.com SIP/2.0Via: SIP/2.0/UDP 167.180.112.24From: sip:alice@hereway.comTo: sip:bob@domain.com Call-ID: a2e3a@pigeon.hereway.comContent-Type: 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 will be necessary.

• Alice sends and receives SIP messages using the SIP default port number 506.

• Alice specifies in Via:header that SIP client sends and receives SIP messages over UDP

Unit - 4

Name translation and user locataionName 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

Unit - 4

SIP RegistrarSIP Registrar

REGISTER sip:domain.com SIP/2.0Via: SIP/2.0/UDP 193.64.210.89 From: sip:bob@domain.comTo: sip:bob@domain.comExpires: 3600

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

(similar function needed by Instant Messaging)

Register Message:

Unit - 4

SIP ProxySIP Proxy

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

Note: proxy is analogous to local DNS server

Unit - 4

ExampleExampleCaller jim@umass.edu with places a call to keith@upenn.edu

(1) Jim sends INVITEmessage to umass SIPproxy. (2) Proxy forwardsrequest to upenn registrar server. (3) upenn server returnsredirect response,indicating that it should try keith@eurecom.fr

(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

Unit - 4

Comparison with H.323Comparison 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 and codecs.

SIP is a single component. Works with RTP, but does not mandate it. Can be combined with other protocols and services.

H.323 comes from the ITU (telephony).

SIP comes from IETF: Borrows much of its concepts from HTTP. SIP has a Web flavor, whereas H.323 has a telephony flavor.

SIP uses the KISS principle: Keep it simple stupid.

Unit - 4

Content distribution networks (CDNs)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

origin server in North America

CDN distribution node

CDN serverin S. America CDN server

in Europe

CDN serverin Asia

Unit - 4

Content distribution networks (CDNs)Content distribution networks (CDNs)

Content replication CDN (e.g., Akamai)

customer is the content provider (e.g., CNN)

CDN replicates customers’ content in CDN servers. When provider updates content, CDN updates servers

origin server in North America

CDN distribution node

CDN serverin S. America CDN server

in Europe

CDN serverin Asia

Unit - 4

CDN exampleCDN example

origin server (www.foo.com) distributes HTML replaces: http://www.foo.com/sports.ruth.gif

with http://www.cdn.com/www.foo.com/sports/ruth.gif

HTTP request for

www.foo.com/sports/sports.html

DNS query for www.cdn.com

HTTP request for

www.cdn.com/www.foo.com/sports/ruth.gif

1

2

3

Origin server

CDNs authoritative DNS server

NearbyCDN server

CDN company (cdn.com) distributes gif files uses its authoritative

DNS server to route redirect requests

Unit - 4

More about CDNsMore about CDNs

routing requests CDN creates a “map”, indicating

distances from leaf ISPs and CDN nodes when query arrives at authoritative DNS

server: server determines ISP from which query

originates uses “map” to determine best CDN server

CDN nodes create application-layer overlay network

Unit - 4

Chapter 7 outlineChapter 7 outline 7.1 Multimedia

Networking Applications 7.2 Streaming stored

audio and video 7.3 Real-time

Multimedia: Internet Phone study

7.4 Protocols for Real-Time Interactive Applications RTP,RTCP,SIP

7.5 Distributing Multimedia: content distribution networks

7.6 Beyond Best Effort

7.7 Scheduling and Policing Mechanisms

7.8 Integrated Services and Differentiated Services

7.9 RSVP

Unit - 4

Improving QOS in IP NetworksImproving QOS in IP NetworksThus far: “making the best of best effort”Future: next generation Internet with QoS guarantees

RSVP: signaling for resource reservations Differentiated Services: differential guarantees Integrated Services: firm guarantees

simple model for sharing and congestion studies:

Unit - 4

Principles for QOS GuaranteesPrinciples for QOS Guarantees

Example: 1MbpsI P 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

Unit - 4

Principles for QOS Guarantees Principles for QOS Guarantees (more)(more)

what if applications misbehave (audio sends higher than declared rate) policing: force source adherence to bandwidth allocations

marking and policing at network edge: similar to ATM UNI (User Network Interface)

provide protection (isolation) for one class from othersPrinciple 2

Unit - 4

Principles for QOS Guarantees Principles for QOS Guarantees (more)(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

Unit - 4

Principles for QOS Guarantees Principles for QOS Guarantees (more)(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

Unit - 4

Summary of QoS Principles Summary of QoS Principles

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

Unit - 4

Scheduling And Policing Scheduling And Policing MechanismsMechanisms

scheduling: choose next packet to send on link FIFO (first in first out) scheduling: send in order of arrival to queue

real-world example? 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

Unit - 4

Scheduling Policies: moreScheduling 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..

Real world example?

Unit - 4

Scheduling Policies: still moreScheduling Policies: still moreround robin scheduling: multiple classes cyclically scan class queues, serving one from each class (if available) real world example?

Unit - 4

Scheduling Policies: still moreScheduling Policies: still more

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

service in each cycle real-world example?

Unit - 4

Policing MechanismsPolicing MechanismsGoal: limit traffic to not exceed declared parametersThree 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 or 6000

packets per min have same average! Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 1500 ppm peak

rate (Max.) Burst Size: max. number of pkts sent consecutively (with

no intervening idle)

Unit - 4

Policing MechanismsPolicing Mechanisms

Token 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).

Unit - 4

Policing Mechanisms (more)Policing Mechanisms (more)

token bucket, WFQ combine to provide guaranteed upper bound on delay, i.e., QoS guarantee!

WFQ

token rate, r

bucket size, b

per-flowrate, R

D = b/Rmax

arrivingtraffic

Unit - 4

Chapter 7 outlineChapter 7 outline 7.1 Multimedia

Networking Applications 7.2 Streaming stored

audio and video 7.3 Real-time

Multimedia: Internet Phone study

7.4 Protocols for Real-Time Interactive Applications RTP,RTCP,SIP

7.5 Distributing Multimedia: content distribution networks

7.6 Beyond Best Effort

7.7 Scheduling and Policing Mechanisms

7.8 Integrated Services and Differentiated Services

7.9 RSVP

Unit - 4

IETF Integrated ServicesIETF 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?

Unit - 4

Intserv: QoS guarantee scenarioIntserv: QoS guarantee scenario Resource reservation

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

QoS-sensitive scheduling (e.g.,

WFQ)

request/reply

Unit - 4

Call AdmissionCall 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

Unit - 4

Intserv QoS: Service models Intserv QoS: Service models [rfc2211, rfc [rfc2211, rfc 2212]2212]

Guaranteed service: worst case traffic arrival:

leaky-bucket-policed source simple (mathematically

provable) bound on delay [Parekh 1992, Cruz 1988]

Controlled load service: "a quality of service

closely approximating the QoS that same flow would receive from an unloaded network element."

WFQ

token rate, r

bucket size, b

per-flowrate, R

D = b/Rmax

arrivingtraffic

Unit - 4

IETF Differentiated ServicesIETF Differentiated ServicesConcerns with Intserv: Scalability: signaling, maintaining per-flow router state difficult with

large number of flows Flexible Service Models: Intserv has only two classes. Also want

“qualitative” service classes “behaves like a wire” relative service distinction: Platinum, Gold, Silver

Diffserv approach: simple functions in network core, relatively complex functions at edge

routers (or hosts) Do’t define define service classes, provide functional components to

build service classes

Unit - 4

Edge router: per-flow traffic

management

marks packets as 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 Assured Forwarding

Diffserv ArchitectureDiffserv Architecture

scheduling

...

r

b

marking

Unit - 4

Edge-router Packet MarkingEdge-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

Unit - 4

Classification and ConditioningClassification 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 PHB that the packet will receive

2 bits are currently unused

Unit - 4

Classification and ConditioningClassification 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

Unit - 4

Forwarding (PHB)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

Unit - 4

Forwarding (PHB)Forwarding (PHB)

PHBs being developed: Expedited Forwarding: pkt departure

rate of a class equals or exceeds specified rate logical link with a minimum guaranteed rate

Assured Forwarding: 4 classes of traffic each guaranteed minimum amount of

bandwidth each with three drop preference partitions

Unit - 4

Chapter 7 outlineChapter 7 outline 7.1 Multimedia

Networking Applications 7.2 Streaming stored

audio and video 7.3 Real-time

Multimedia: Internet Phone study

7.4 Protocols for Real-Time Interactive Applications RTP,RTCP,SIP

7.5 Distributing Multimedia: content distribution networks

7.6 Beyond Best Effort

7.7 Scheduling and Policing Mechanisms

7.8 Integrated Services and Differentiated Services

7.9 RSVP

Unit - 4

Signaling in the InternetSignaling in the Internet

connectionless (stateless)

forwarding by IP routers

best effort service

no network signaling protocols

in initial IP design

+ =

New requirement: reserve resources along end-to-end path (end system, routers) for QoS for multimedia applications

RSVP: Resource Reservation Protocol [RFC 2205] “ … allow users to communicate requirements to

network in robust and efficient way.” i.e., signaling !

earlier Internet Signaling protocol: ST-II [RFC 1819]

Unit - 4

RSVP Design GoalsRSVP Design Goals

1. accommodate heterogeneous receivers (different bandwidth along paths)

2. accommodate different applications with different resource requirements

3. make multicast a first class service, with adaptation to multicast group membership

4. leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes

5. control protocol overhead to grow (at worst) linear in # receivers

6. modular design for heterogeneous underlying technologies

Unit - 4

RSVP: does not…RSVP: does not… specify how resources are to be reserved

rather: a mechanism for communicating needs determine routes packets will take

that’s the job of routing protocolssignaling decoupled from routing

interact with forwarding of packetsseparation of control (signaling) and data

(forwarding) planes

Unit - 4

RSVP: overview of operationRSVP: overview of operation senders, receiver join a multicast group

done outside of RSVP senders need not join group

sender-to-network signaling path message: make sender presence known to routers path teardown: delete sender’s path state from routers

receiver-to-network signaling reservation message: reserve resources from sender(s)

to receiver reservation teardown: remove receiver reservations

network-to-end-system signaling path error reservation error

Unit - 4

Path msgs: RSVP Path msgs: RSVP sender-to-networksender-to-network signalingsignaling

path message contents: address: unicast destination, or multicast group flowspec: bandwidth requirements spec. filter flag: if yes, record identities of upstream

senders (to allow packets filtering by source) previous hop: upstream router/host ID refresh time: time until this info times out

path message: communicates sender info, and reverse-path-to-sender routing info later upstream forwarding of receiver

reservations

Unit - 4

RSVP: simple audio conferenceRSVP: simple audio conference

H1, H2, H3, H4, H5 both senders and receivers

multicast group m1 no filtering: packets from any sender

forwarded audio rate: b only one multicast routing tree possible

H2

H5

H3

H4H1

R1 R2 R3

Unit - 4

inout

inout

inout

RSVP: building up path stateRSVP: building up path state

H1, …, H5 all send path messages on m1:

(address=m1, Tspec=b, filter-spec=no-filter,refresh=100)

Suppose H1 sends first path message

H2

H5

H3

H4H1

R1 R2 R3L1

L2 L3

L4L5

L6 L7

L5 L7L6

L1L2 L6 L3

L7L4m1:

m1:

m1:

Unit - 4

inout

inout

inout

RSVP: building up path stateRSVP: building up path state

next, H5 sends path message, creating more state in routers

H2

H5

H3

H4H1

R1 R2 R3L1

L2 L3

L4L5

L6 L7

L5 L7L6

L1L2 L6 L3

L7L4

L5

L6L1

L6

m1:

m1:

m1:

Unit - 4

inout

inout

inout

RSVP: building up path stateRSVP: building up path state

H2, H3, H5 send path msgs, completing path state tables

H2

H5

H3

H4H1

R1 R2 R3L1

L2 L3

L4L5

L6 L7

L5 L7L6

L1L2 L6 L3

L7L4

L5

L6L1

L6L7

L4L3L7

L2m1:

m1:

m1:

Unit - 4

reservation msgs: reservation msgs: receiver-to-network receiver-to-network signalingsignaling

reservation message contents: desired bandwidth: filter type:

• no filter: any packets address to multicast group can use reservation

• fixed filter: only packets from specific set of senders can use reservation

• dynamic filter: senders who’s p[ackets can be forwarded across link will change (by receiver choce) over time.

filter spec reservations flow upstream from receiver-to-

senders, reserving resources, creating additional, receiver-related state at routers

Unit - 4

RSVP: RSVP: receiverreceiver reservation example reservation example 11

H1 wants to receive audio from all other senders

H1 reservation msg flows uptree to sources

H1 only reserves enough bandwidth for 1 audio stream

reservation is of type “no filter” – any sender can use reserved bandwidth

H2

H5

H3

H4H1

R1 R2 R3L1

L2 L3

L4L5

L6 L7

Unit - 4

inout

RSVP: RSVP: receiverreceiver reservation example reservation example 11

H1 reservation msgs flows uptree to sources

routers, hosts reserve bandwidth b needed on downstream links towards H1

H2

H5

H3

H4H1

R1 R2 R3L1

L2 L3

L4L5

L6 L7

L1L2 L6

L6L1(b)

inout

L5L6 L7

L7L5 (b)

L6

inout

L3L4 L7

L7L3 (b)

L4L2

b

bb

b

bb

b

m1:

m1:

m1:

Unit - 4

inout

RSVP: RSVP: receiverreceiver reservation example 1 reservation example 1 (more)(more)

next, H2 makes no-filter reservation for bandwidth b

H2 forwards to R1, R1 forwards to H1 and R2 (?)

R2 takes no action, since b already reserved on L6

H2

H5

H3

H4H1

R1 R2 R3L1

L2 L3

L4L5

L6 L7

L1L2 L6

L6L1(b)

inout

L5L6 L7

L7L5 (b)

L6

inout

L3L4 L7

L7L3 (b)

L4L2

b

bb

b

bb

b

b

b

(b)m1:

m1:

m1:

Unit - 4

inout

RSVP: RSVP: receiverreceiver reservation: issues reservation: issues

What if multiple senders (e.g., H3, H4, H5) over link (e.g., L6)?

arbitrary interleaving of packets L6 flow policed by leaky bucket: if H3+H4+H5

sending rate exceeds b, packet loss will occur

H2

H5

H3

H4H1

R1 R2 R3L1

L2 L3

L4L5

L6 L7

L1L2 L6

L6L1(b)

inout

L5L6 L7

L7L5 (b)

L6

inout

L3L4 L7

L7L3 (b)

L4L2

b

bb

b

bb

b

b

b

(b)m1:

m1:

m1:

Unit - 4

RSVP: example 2RSVP: example 2

H1, H4 are only senders send path messages as before, indicating

filtered reservation Routers store upstream senders for each

upstream link H2 will want to receive from H4 (only)

H2 H3

H4H1

R1 R2 R3L1

L2 L3

L4L6 L7

H2 H3

L2 L3

Unit - 4

RSVP: example 2RSVP: example 2 H1, H4 are only senders

send path messages as before, indicating filtered reservation

H2 H3

H4H1

R1 R3L1

L2 L3

L4L6 L7

H2 H3

L2 L3

L2(H1-via-H1 ; H4-via-R2 )L6(H1-via-H1 )L1(H4-via-R2 )

in

out

L6(H4-via-R3 )L7(H1-via-R1 )

in

out

L1, L6

L6, L7

L3(H4-via-H4 ; H1-via-R3 )L4(H1-via-R2 )L7(H4-via-H4 )

in

out

L4, L7

R2

Unit - 4

RSVP: example 2RSVP: example 2 receiver H2 sends reservation message for

source H4 at bandwidth b propagated upstream towards H4, reserving b

H2 H3

H4H1

R1 R3L1

L2 L3

L4L6 L7

H2 H3

L2 L3

L2(H1-via-H1 ;H4-via-R2 )L6(H1-via-H1 )L1(H4-via-R2 )

in

out

L6(H4-via-R3 )L7(H1-via-R1 )

in

out

L1, L6

L6, L7

L3(H4-via-H4 ; H1-via-R2 )L4(H1-via-62 )L7(H4-via-H4 )

in

out

L4, L7

R2

(b)

(b)

(b)

L1

bb b

b

Unit - 4

RSVP: RSVP: soft-statesoft-state senders periodically resend path msgs to refresh

(maintain) state receivers periodically resend resv msgs to refresh

(maintain) state path and resv msgs have TTL field, specifying

refresh interval

H2 H3

H4H1

R1 R3L1

L2 L3

L4L6 L7

H2 H3

L2 L3

L2(H1-via-H1 ;H4-via-R2 )L6(H1-via-H1 )L1(H4-via-R2 )

in

out

L6(H4-via-R3 )L7(H1-via-R1 )

in

out

L1, L6

L6, L7

L3(H4-via-H4 ; H1-via-R3 )L4(H1-via-62 )L7(H4-via-H4 )

in

out

L4, L7

R2

(b)

(b)

(b)

L1

bb b

b

Unit - 4

RSVP: RSVP: soft-statesoft-state suppose H4 (sender) leaves without performing teardown

H2 H3

H4H1

R1 R3L1

L2 L3

L4L6 L7

H2 H3

L2 L3

L2(H1-via-H1 ;H4-via-R2 )L6(H1-via-H1 )L1(H4-via-R2 )

in

out

L6(H4-via-R3 )L7(H1-via-R1 )

in

out

L1, L6

L6, L7

L3(H4-via-H4 ; H1-via-R3 )L4(H1-via-62 )L7(H4-via-H4 )

in

out

L4, L7

R2

(b)

(b)

(b)

L1

bb b

b

eventually state in routers will timeout and disappear!

gonefishing!

Unit - 4

The many uses of reservation/path The many uses of reservation/path refreshrefresh

recover from an earlier lost refresh message expected time until refresh received must be longer

than timeout interval! (short timer interval desired) Handle receiver/sender that goes away without

teardown Sender/receiver state will timeout and disappear

Reservation refreshes will cause new reservations to be made to a receiver from a sender who has joined since receivers last reservation refresh E.g., in previous example, H1 is only receiver, H3 only

sender. Path/reservation messages complete, data flows H4 joins as sender, nothing happens until H3 refreshes

reservation, causing R3 to forward reservation to H4, which allocates bandwidth

Unit - 4

RSVP: reflectionsRSVP: reflections

multicast as a “first class” service receiver-oriented reservations use of soft-state

Unit - 4

Multimedia Networking: SummaryMultimedia Networking: Summarymultimedia applications and

requirementsmaking the best of today’s best

effort servicescheduling and policing

mechanismsnext generation Internet: Intserv,

RSVP, Diffserv