Www.ischool.drexel.edu INFO 331 Computer Networking Technology II Chapter 7 Multimedia Networking Dr. Jennifer Booker 1INFO 331 chapter 7

www.ischool.drexel.edu

INFO 331Computer Networking

Technology II Chapter 7

Multimedia Networking

Dr. Jennifer Booker

1INFO 331 chapter 7



• Recent years have seen massive growth in Internet audio and video apps– Streaming video, IP telephony, Internet radio,

teleconferencing, interactive games, distance learning, Netflix, Hulu, etc.

• Older Internet apps (email, WWW, FTP) were very elastic in bandwidth needs, but multimedia is much fussier – delay sensitive– But they’re also more loss-tolerant

2INFO 331 chapter 7



• Multimedia apps are in three categories– Streaming stored audio & video (AV)– Streaming live AV– Conversational voice/VOIP

• This excludes download-stuff-and-play, such as MP3s, iTunes, etc.– Download entire file before play starts – FTP or HTTP are fine for them

3INFO 331 chapter 7


Video properties

• Compression helps speed information transmission rate by reducing its volume

• Long done for static images (JPG, GIF)– Each pixel is 24 bits of color data (RGB),

• A 1024x768 pixel image is 1024x768x24/8 = 2.36 MB

– Compression can reduce image size a factor of 10 without severe quality loss

• Key tradeoff: more compression = more losses

• Huge field (e.g. ISBN 1565921615)4INFO 331 chapter 7


Video properties

• Video needs a high bit rate, typically 100 kbps for minimal video conferencing to 3 Mbps for streaming high def movies– Streaming MP3s or viewing many pictures

might take under 100 MB per hour, but streaming video might take over 1 GB per hr

• Might have several versions of the same content at different quality levels, e.g. from 300 kbps to 3 Mbps

5INFO 331 chapter 7


Video Compression

• Video is a series of images presented at 24 or 30 images per second– Hence the size of the images (X by 0.75*X

pixels), and the color resolution (number of bits per pixel) also affect the raw data volume

– Widescreen image sizes have a 16x9 ratio instead of 4x3 ratio

6INFO 331 chapter 7


Video Compression

• Video compression standards are mostly the MPEG family– MPEG 1 for CD quality video at 1.5 Mbps– MPEG 2 for DVD quality video, 3-6 Mbps– MPEG 4 for object oriented video– H.261 from ITU– Proprietary formats, such as QuickTime

(includes MPEG-4 and H.264), Real networks, etc.

7INFO 331 chapter 7


Audio properties

• Digital audio signals have three major characteristics– The sampling rate (samples per second)– The number of channels sampled (mono,

stereo, etc.)– The number of bits per sample

• These all affect the raw amount of data being sent, before compression is applied

8INFO 331 chapter 7


Audio properties

• Raw audio signals are recorded at some sample rate per second, per channel– 8k, 16k, or 32k samples/sec (Hz) are common

low grade rates– 44.1k (CD quality), 48k, 96k, and even

192kHz sample rates are used for professional audio

• The number of channels used is typically one (mono), two (stereo), or 5 to 7 (surround)

9INFO 331 chapter 7


Audio properties

• Each sample is quantized into some number of bits to describe its relative strength– 8 bits gives 256 values from silent to REALLY LOUD,

which is typical for cheap built-in audio– CD quality uses 16 bits per sample– Pro audio typically uses 24 or 32 bits per sample

• So one minute of absurd quality 7 channel surround would use 192k sample/sec x 7 channels x 32

bits/sample / 8 bit/byte * 60 sec/min = 322.56 MB/min!

10INFO 331 chapter 7


Audio Compression

• This approach for audio is formally called Pulse Code Modulation (PCM)– Mono speech recording at 8kHz sample rate

and 8 bits/sample equals 64 kbps of data– CD quality (44.1 kHz, 16 bit, stereo) = 1.411

Mbps

• Modems can’t handle 64 kbps, and some broadband users can’t consistently get 1.4 Mbps, so compression is needed even for just audio



Audio Compression

• Common audio compression standards include– GSM, G.729, and G.723.3 from ITU– “MPEG 1 layer 3”, a.k.a. MP3, which

compresses to 96, 128, or 160 kbps– AAC (Advanced Audio Coding) is widely used

by Apple



Streaming stored AV

• Here, clients request on-demand compressed AV files that are stored on servers– Content may include lectures, music, TV, etc.– CNN video, YouTube, etc.

• Main features of this app type are:– Stored, prerecorded media; pause, ffwd, rew– Streaming, hence can play part of the media while

downloading more of it– Continuous play out – should keep original timing



Conversational voice/VOIP

• This class of apps allows interaction between people at both ends (or many ends) of a connection, such as Internet telephony or voice-over-IP (VOIP)– E.g. Skype – Very delay sensitive

• Transmission delays under 150 ms are good, under 400 ms is okay, and over 400 ms is bad

– Is often loss-tolerant



Streaming live AV

• This is live broadcast of radio or TV over the Internet– Can’t fast forward, since it hasn’t happened yet,

but local storage of what’s been received can allow pausing and rewinding in some cases

– Often accomplished using IP multicasting or IPTV, but more often done via separate unicast streams

– Also has continuous play out, can tolerate some startup delay



Streaming Stored AV

• Streaming stored audio/video has become very popular (YouTube, Netflix, Hulu) because– Disk space is dirt cheap (< $50/TB)– Internet infrastructure is improving, but over 50% of

downstream traffic can be this type– There’s enormous demand to entertain me NOW

• In this multimedia mode, clients request compressed AV files that live on servers– Files are segmented, with special headers used for

encapsulating them



Streaming Stored AV

• Three major types of this traffic– UDP-streaming– HTTP-streaming– Adaptive HTTP-streaming

• All use client buffering to soak up variations in server-client delay



Streaming Stored AV

• UDP streaming sends packets over UDP with a small client-side buffer

– RTP – Real Time Protocol is used to encapsulate the segments

– RTSP – Real Time Streaming Protocol is used for client/server interaction

• Not used much since hard to ensure continuous playout, need for an RTSP server, and firewalls may block UDP



Streaming Stored AV

• HTTP streaming just uses ordinary GET commands to send files over TCP– The HTTP byte-range header helps specify

which parts of the video to get

• Client uses a buffer to prefetch video frames– If buffer gets full, will limit the server send rate

• Most stored video uses this



Streaming Stored AV

• Adaptive HTTP-streaming such as DASH (Dynamic Adaptive Streaming over HTTP) changes the video quality obtained depending on the connection bandwidth• Comcast has 8-10 different quality formats within

MPEG-4• Server has a manifest file with version data• Based on actual download rate pick best version• Likewise can adjust audio resolution



Content Distribution Networks

• Content Distribution Networks (CDNs) are to address the need for lots of people accessing the same stored media often and/or at once– After all, a single server source would have terrible

bandwidth and packet loss issues– A CDN can be private or third party

• Often a content provider (CNN, Google, etc.) will pay a CDN company (Akamai, Limelight) to provide videos to users with as little delay as possible




• The CDN approach is simple – make lots of copies of the media, and put them on lots of servers everywhere

• CDN servers get put throughout the Internet (e.g. 1700 locations for Akamai)

• Often the CDN company will lease data center space to house the servers– Data centers might be located at second or

third tier ISPs




• Google’s CDN has one huge autonomous system with:– 8 mega data centers with ~100,000 servers in

each (6 in USA, 2 in Europe)– 30 ‘bring-home’ clusters of 100-500 servers

worldwide near tier-1 ISPs– Hundreds of ‘enter-deep’ clusters located

within ISPs




• The CDN gets source media (videos) from the customer, and copies them to the servers

• When a user requests content, the nearest (or most available) CDN server delivers it– DNS redirection is used to find the correct CDN

server– This results in URLs with two addresses in them – the

first is the CDN, the second is the file name• http://www.cdn.com/www.cnn.com/zoo/turtle.mpg




• The ‘best’ server for each ISP is determined using the same approaches we saw for BGP routing tables

• CDNs may also be used within a large corporation to stream, for example, training videos locally




• CDN deployment depends on a good cluster selection strategy– Could use geographically closest– Perform periodic real-time measurements of

delay and losses– Use IP anycast, and let BGP decide



Voice-over-IP

• IP provides only “best-effort” service with no guarantees on timing or even delivery

• Packet loss is quite possible with UDP traffic, but TCP may get packets there too late to be useful!



Internet Phone

• Real-time apps, such as Internet phone and video conferencing, are very sensitive to packet delay, jitter, and packet loss

• See how these issues are handled for Internet phone– Data is generated at 8 kBps– Every 20 ms, a chunk of 160 bytes of data gets a

header attached, and the packet is sent via UDP

• Ideally, these packets all get to the receiver



Internet Phone

• The receiver must decide– When to play back a given packet – What to do if a packet is missing

• Recall packets can be lost if they arrive at a router with a full input or output queue

• From 1-20% packet loss can be tolerated– Forward error correction (FEC) can help make

up for lost packets



Internet Phone

• End-to-end delay can be confusing for the receiver, if it simply takes too long for the data to get there– Recall the under 150 ms, 150-400 ms, and

over 400 ms ranges for good, ok, and bad delays

– Internet phone may discard packets over 400 ms old

• Jitter is typically caused by variation in queuing delays



Internet Phone

• Some jitter problems can be removed by using sequence numbers, time stamps, and playout delays to make sure packets are in order as well as possible

• Playout delay can be fixed or adaptive• For fixed playout delay, anything arriving

after its planned play time is discarded– The amount of delay is typically 150-400 ms,

less under good network conditions



Adaptive playout delay

• Adaptive playout delay is often used because long delay is annoying to the users– Want to make the delay as small as possible

without losing a lot of packets– Delay is reassessed for each talk spurt

(period of transmission)

• Similar to calculation of timeout interval for TCP, use previous history, amended by the most recent data



Recovering from packet loss

• The method for compensating for lost packets is the loss recovery scheme– Here, ‘lost’ means it never got there, or it

got there after its planned play time

• Retransmitting lost packets doesn’t make sense here (why?)

• Instead anticipate loss, using methods like– Forward Error Correction– Interleaving



Forward Error Correction

• Forward Error Correction adds a little redundant data to packets to help allow recreating what missing packets had in them– RAT (Robust Audio Tool) is an open source

Internet phone app that uses FEC and RTP

• There are lots of FEC approaches; we’ll look at two of them




• First is to send a redundant encoded chunk of data after every n normal chunks– The redundant chunk is obtained from

XOR-ing the normal chunks

• If any ONE of the n chunks is missing, it can be reconstructed mathematically– But if two or more packets are lost, tough luck

• This increases transmission rate 100/n%




• The second approach is sneakier– Add a second, lower quality data stream in

parallel with the primary stream (e.g. tack on a 13 kbps stream to a 65 kbps stream)

– When loss occurs, use the low quality stream – This increases playout delay little– Many variations on this approach are

possible, especially to allow for higher loss rates



Interleaving

• Interleaving breaks the stream of data into smaller chunks, and rearranges how the chunks are sent– So if each chunk of data is broken into four pieces,

then the first unit of data sent has chunks number 1, 5, 9, and 13 instead of just 1-4

• The second chunk sent has 2, 6, 10, & 14; the third chunk has 3, 7, 11, & 15, etc.

– That way if a chunk is lost, the loss is distributed across a wider range of time



Interleaving

• This increases latency (have to break up & reassemble chunks)

• Doesn’t increase bandwidth of data – no extra data is being sent

• So how can we fix the data if some is lost?



Error concealment

• Voice is relatively easy to fix if a little data is missing– Loss rates under 15% for packets 4-40 ms long

• One technique is simply repetition – copy the previous packet and play it again– Easy to do and usually works ok

• Or can try to interpolate between before and after packets– More tricky computationally, but sounds better



Real-time Interactive Protocols

• Demand for real time interactive applications is huge, so there has been a lot of work on protocols to help make such apps easier to develop

• Here we’ll look at two common ones– RTP, the Real-time Transport Protocol– SIP



RTP

• RTP defines a standard AV chunk-of-data structure for data, sequence numbers, timestamps, etc. – The actual data format can be a proprietary

format, or other AV formats such as PCM, GSM, MP3, MPEG, H.263, etc.

• RTP is defined by RFC 3550, and usually runs over UDP (no fixed port number)

• Often used for Internet telephony apps



Application

AV dataFormat can be PCM, GSM, MP3, MPEG, H.263, proprietary AV formats, etc.

Header infoTimestamp, sequence number

Pass to transport layer (UDP)

Pass to network layer (IP)

RTP



RTP

• An RTP packet consists of the RTP header plus the data– The RTP header is at least 12 B (3 rows x 4B ea.)

• Apps that both use RTP have a better chance of cooperating (e.g. if two users have different apps at each end of a phone call)

• RTP doesn’t guarantee data delivery, or sequence– Routers can’t tell RTP data from anything else



RTP

• Every data source (video camera, microphone) can have an RTP data stream– Video conference could have four RTP streams to

send video each direction, and audio each way– Some compression formats (MPEG-1 and -2)

combine audio and video into one stream

• RTP can be used with unicast or multicast– A group of streams from the same origin form an

RTP session



RTP Header Fields

• The RTP header is very simple– First line has:

• Nine bits of misc. identifiers (version, how long the CSRC list will be, if custom extensions are used, etc.)

• Payload type (7 b)• Sequence number (16 b)

– Timestamp (32 b)– SSRC (32 b)– CSRC list (0 to 15 lines of 32 b each)



RTP Header

• The Payload Type is a numeric code that identifies what format the data has – PCM, GSM, MPEG, etc.

• The Sequence Number starts at a random value, and “increments by one for each RTP data packet sent” [RFC 3550]

• The Timestamp is when the first data was sampled; change from one to the next is inversely proportional to the sampling rate



RTP Header

• The SSRC is the synchronization source identifier – just a random number to identify the source– Only goal is that no two SSRCs are the same

in a session

• The CSRC list is the contributing sources for audio data – Is often used for mixing, or to identify specific

data sources (Fred vs. Wilma)47INFO 331 chapter 7


RTP App Development

• RTP can be implemented two ways– Use RFC 3550 and manually implement RTP

headers within your application– Use existing API libraries (e.g. in C or Java) to

implement RTP for you, and have your app call them

– Hmm, I wonder which is easier?

• RTP can be coded as an app-layer protocol, or used in conjunction with UDP via the API



SIP

• The Session Initiation Protocol (SIP) is defined by RFC 5411– SIP can be over UDP or TCP, and uses port

5060

• It is designed to handle Internet telephone across LANs (not local phone exchanges)– It allows calls to be placed over IP– Allows caller to determine IP of callee (varies)– Can add media streams during call



SIP Session Example

• Caller sends an INVITE message– Tells caller and callee IP addresses, media

format and encoding, type of encapsulation (RTP) and receiving port number

• Callee sends response msg– Confirms IP address, encoding,

encapsulation, and port number

• Caller sends ACK to callee, and the call can take place



SIP Session Example

time time

Bob 'ste rm inal rings

A lice

167.180.112.24

Bob

193.64 .210.89

port 5060

port 38060

Law aud io

G SMport 48753

INVITE [email protected]=IN IP4 167.180.112.24m=audio 38060 RTP/AVP 0port 5060

200 OKc=IN IP4 193.64.210.89

m=audio 48753 RTP/AVP 3

ACKport 5060



SIP

• So what?– The two parties can be using completely

different encoding and encapsulation methods, yet carry on the call

– Also note that the control messages stay over port 5060, yet the media are over two other, negotiated ports – so SIP uses out-of-band control messages

– SIP requires ACK of all messages, so either UDP or TCP can be used



SIP Addressing

• SIP addresses can look like email addresses, or cite IP addresses, or be a phone number, or even a full personal name– [email protected]– [email protected]– [email protected]



SIP Messages

• The SIP protocol is huge – 269 pages – so only a brief overview is in order

• The INVITE message typically is addressed to an email-like address, e.g. [email protected]

• Each device the message passes through adds a Via: header with that device’s IP

• A SIP proxy finds the IP of the device the callee is currently using



SIP Messages

• Every SIP user is associated with a SIP registrar– When a SIP device is launched, it registers

with the registrar to give its current IP address

• So the SIP proxy asks the callee’s registrar for their IP address– The proxy might be redirected to another

registrar if the callee is not nearby



SIP applications

• SIP has been described for voice over IP, but can also be used for any media – video, even text messaging

• SIP software is widely available for many common functions, on various platforms (PC/Mac/Linux)



Network support for multimedia

• Three main approaches for handling AV– Make do with Internet’s best effort status quo– Provide differentiated service through packet

marking, policing, and scheduling– Provide pre-connection guarantees of quality

of service (QoS) through differentiated service plus call admission and signaling



Fix the Internet!

• Others insist the Internet doesn’t need massive changes– Let ISPs upgrade bandwidth as customers

demand it– Increase ISP caching for common requested

stored AV– Add content distribution networks (CDNs) for

paid stored media, conveniently near network edges



Fix the Internet!

• Second approach is to add pricing at the network & transport layers, to pay for better service (Diffserv approach)– We’ve seen net neutrality battles over this

option



Fix the Internet!

• Some argue the Internet should allow for end-to-end bandwidth guarantees, like virtual circuit networks can provide– Would require massive changes to routers to

establish fixed paths for some service types– Hard guarantees mean you’re certain to

receive the QoS you paid for– Soft guarantees mean you’re likely to receive

the QoS you paid for



Dimensioning best-effort networks

• Many techniques we’ve seen can be used to improve the quality of service of multimedia apps– We want low RTT, little jitter, few packets lost

• Another solution is to throw enough money at the network to avoid all these problems– Given enough resources, none of these

problems occur!



Dimensioning best-effort networks

• To do so requires adequate bandwidth provisioning, and at a larger scale, good network dimensioning; issues include – Model traffic demand between network

end points– Clear performance requirements– Model performance for a given workload – Model to predict minimum bandwidth to

meet all requirements



Beyond Best Effort

• So lots of techniques have been used to get the most out of the Internet’s ‘best effort’ approach– The current quality of service (QoS) has no

guarantees– How can we improve on that?

• Change the architecture of the Internet!• Look at a sample network to examine the

problems



A Sample Network



A Sample Network

• In this example, the local networks are assumed much faster than the 1.5 Mbps connection between them– Two apps are competing for that 1.5 Mbps of

bandwidth, e.g. leaving the first router (R1)

• Scenario 1: If hosts H1 is sending FTP to H3, and then H2 tries to send audio traffic to H4, the router R1 will already be full, making for lost audio packets



A Sample Network

• But IF we could mark audio packets versus FTP packets, we could give priority to audio packets, since they are delay sensitive– Principle 1: packet marking lets a router

distinguish between different classes of traffic

• Scenario 2 – what if the FTP traffic was on a high priority (paid) connection, but the audio traffic was normal free Internet– Principle 1a: packet classification lets a router

distinguish between different classes of traffic



A Sample Network

• The distinction is important– Marking packets for type of payload is one

way to classify them, but hardly the only one– The packets could be prioritized by the

originating IP address (which could imply whether premium service was used)

• Scenario 3: Bad application– What if the audio app should get priority, but

it abuses that and hogs the whole link?



A Sample Network

– Instead of allowing 1.0 Mbps for audio, the app tries to take the whole 1.5 Mbps, thereby shutting off the FTP service

– Just like circuit switching, we want to isolate each traffic flow, so a misbehaving app doesn’t ruin the link for everyone

– Principle 2: Some degree of isolation is desirable to protect traffic flows from each other

– This might imply the need for traffic cops, to ensure app compliance



A Sample Network

• So two approaches could address these concerns and improve the QoS– Mark packets, and police app behavior to

share bandwidth fairly– Or, logically isolate traffic flows, and

designate some specific bandwidth for each

• But if the flows are isolated, need to make sure full bandwidth is used if a traffic flow isn’t active



A Sample Network

– Principle 3: When traffic flows are isolated, need to use resources (bandwidth, router buffers) as efficiently as possible

• Scenario 4: Two 1.0 Mbps audio apps over a 1.5 Mbps link– Both audio apps can’t get the full bandwidth needed –

each would lose 25% of their packets– To maintain any decent QoS level, the network should

allow or block the flow• Telephones have done this for years!



A Sample Network

– So to guarantee a minimum level of QoS we need a call admission process

– Principle 4: a call admission process is needed to admit or block calls from the network, based on the call’s QoS requirements

• Now that we have the principles of guaranteeing QoS, look at how these are implemented in the Internet



Scheduling and Policing

• Recall at the link level, packets from various flows are multiplexed and queued on the output buffers to go onto a link

• The link-scheduling discipline is the process for queuing packets, which is vital for making QoS guarantees

• Look at FIFO, priority, round robin, and WFQ approaches for link-scheduling



FIFO Scheduling

• FIFO (first-in first-out), also called first-come first-served (FCFS) is the McDonald’s approach to scheduling– Packets are queued in the order in which they

arrived at the device– If the output buffer gets full, the packet-

discarding policy is used to decide which packets are dropped



Priority Queuing

• Here, packets are grouped by their priority – a separate queue for each level of priority– How they are grouped could be by type of service,

source IP, or other methods

• The highest priority queue is emptied before the next priority queue is transmitted– If the high priority queue is empty, a lower priority

packet will still be sent immediately

• Within each queue, FIFO applies



Round Robin Queuing

• In round robin, packets are sorted into classes which have no relative priority– One packet from class 1 is sent, then one

from class 2, etc.– Then go back to class 1 and keep repeating

the cycle

• A work-conserving round robin tries to keep the link busy, so will check the next class if one is empty



Weighted Fair Queuing (WFQ)

• WFQ is a variation on round robin queuing– Here each class is assigned a ‘weight,’ which

determines how much transmission time they get

– In short, some classes are more equal than others

– The weights for all classes total 100%– Incoming packets are classified into the

correct class, which has its own queue

• WFQ is very widely used76INFO 331 chapter 7


Policing

• We may need to control the rate data are allowed to enter the network – a traffic cop

• Could control three aspects– Long term average data rate– Peak or max rate– Burst size – max number of packets in a short

time interval



The Leaky Bucket Analogy

• Bucket holds ‘b’ tokens (packets)

• New tokens are created at rate ‘r’ tokens per sec

• If bucket is full, new tokens are discarded

WFQ

token rate, r

bucket size, b

per-flowrate, R

D = b/ Rmax

arrivingtraffi c

WFQ

token rate, r

bucket size, b

per-flowrate, R

D = b/ Rmax

WFQ

token rate, r

bucket size, b

per-flowrate, R

D = b/ RmaxD = b/ Rmax

arrivingtraffi c



The Leaky Bucket Analogy

• The token generation rate, r, limits the long term average rate

• In any time ‘t’, the max number of tokens which could be added to the network is r*t + b, so that limits the peak rate

• The size of the bucket limits the max burst size – No more than b tokens can be removed

in a short time



WFQ + Leaky buckets

• If we combine the WFQ scheduling with the leaky bucket policing options, what happens to the delay for any class of packets?– If the class’ bucket is full, and ‘b1’ packets

arrive, the rate the bucket is emptied is at least R*wi/sum(w) where R is the link rate, wi is the weight of this class, and sum(w) is the sum of all active classes’ weights



WFQ + Leaky buckets

• So the time for the bi packets to be handled is time = quantity/rate – di = bi/[R*wi/sum(w)] – Hence there is a provable maximum delay

• As long as the rate this class can handle packets is less than [R*wi/sum(w)], the above di equation is the max delay in a WFQ queue



Integrated vs Differentiated Services

• Now we’ve covered the types of mechanisms needed to provide QoS guarantees in the Internet– Cool. So, how is this being implemented?

• Two major architectures are trying to provide an answer – Intserv and Diffserv– Integrated Services = Intserv– Differentiated Services = Diffserv



Intserv

• Intserv provides a way for QoS to be specified for a given session– Kind of like setting up a virtual circuit

• Intserv depends on two key features– Must know what resources are already

occupied by routers along a desired path– Must do call setup to prepare for a session



Intserv

• Call setup includes– A session must define the QoS needs

• An Rspec defines the type of QoS service desired• A Tspec defines the traffic the session will be sending

– See RFC 2210 and 2215 for Rspec and Tspec– Then a signaling protocol, RSVP, is used to set up

along the path’s routers– Each router (element) decides whether to allow the

session



Intserv

• Intserv provides two different types of QoS– Guaranteed QoS – the bounds on queuing

delays for each router are assured• See RFC 2212 for gory details how this is done

– Controlled-load Network Service – the session will receive a “close approximation” to the QoS an unloaded element would provide

• Oddly, this QoS is not quantified• RFC 2211, since you were wondering



Diffserv

• Diffserv (RFC 2475) was in response to problems with Intserv:– Scalability problems with per-flow (session)

reservations • Gets very time consuming for large networks

– Flexibility in service classes – Intserv only provides specific service classes

• Want ability to grade or rank service classes (low/med/high, silver/gold/platinum, etc.)



Diffserv

• Diffserv assumes the whole network will be aware of Diffserv protocols, and that most traffic flows are under Diffserv– You will be assimilated!

• Two main functions under Diffserv– At the edges of the network, hosts or first hop

routers mark the DS field in the packet header– Within the core of the network, traffic is

handled based on its class’ per-hop behavior87INFO 331 chapter 7


Diffserv

• Analogy– The marking process is like getting your hand

stamped at a night club, or having a VIP or backstage pass, or getting comped for spending too much money at a casino

• Packets are classified based on source or destination IP or port, or (transport) protocol– There could be different marking algorithms

for various classes of flows



Diffserv

• Classes of flows could follow a traffic profile– May limit their peak transmission rate, or burstiness– A metering function could be used to control flow rate

into the network, or shape flow rates

• The differentiated services (DS) field (RFC 3260) replaces the type of service and traffic class fields from IPv4 and IPv6– The packets are marked using the DS field



Diffserv

• Within the network, each class’ per-hop behavior (PHB) is critical to controlling its handling– PHB influences the forwarding behavior

(performance) at a Diffserv node– No method for implementing behavior is given– Only key is that they are externally measurable

• Two PHBs have been defined – expedited forwarding (EF), and assured forwarding (AF)



Diffserv

• Expedited forwarding (EF, RFC 3246) requires the departure rate of a class must equal or exceed a certain rate– Implies isolation from other traffic classes, since the

absolute rate is guaranteed

• Assured forwarding (AF, RFC 2597) divides traffic into four classes– Each is guaranteed bandwidth and buffer space– Per class there are three levels of drop preference



Intserv, Diffserv, and Reality

• So while techniques exist to provide QoS guarantees, they aren’t widely used

• Why? The Internet is highly distributed, and not everyone will agree 1) to implement QoS methods, 2) on the type of QoS needed, and 3) on how to pay for it

• And if traffic levels are low, ‘best effort’ will provide the same service as Intserv/Diffserv!



Summary

• Multimedia is huge in the Internet, and getting bigger!– May replace circuit switched telephones

• Here we reviewed the types of multimedia apps, looked at methods for best-effort networking, added methods for QoS guarantees, and outlined architectures for providing QoS


Documents

Www.ischool.drexel.edu INFO 331 Computer Networking Technology II Chapter 7 Multimedia Networking Dr. Jennifer Booker 1INFO 331 chapter 7