Components of a VoIP Network

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© 2006 Cisco Systems, Inc. All rights reserved.

Introducing VoIP

Networks

Benefits of a VoIP Network

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More efficient use of bandwidth and equipment

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Lower transmission costs

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Consolidated network expenses

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Improved employee productivity through features

provided by IP telephony:

IP phones are complete business communication devices. Directory lookups and database applications (XML) Integration of telephony into any

business application Software-based and wireless phones offer mobility.

ƒ

Access to new communications

devices (such as PDAs and

cable set-top boxes)

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© 2006 Cisco Systems, Inc. All rights reserved.

Components of a VoIP Network

© 2006 Cisco Systems, Inc. All rights reserved.

Legacy Analog and VoIP Applications Can

Coexist

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Legacy Analog Interfaces in VoIP Networks

Earth and Magneto Foreign Exchange Office Foreign Exchange Station Analog Interface Type

Trunk, used between switches E&M

Used by the end device side of an FXS–FXO connection FXO

Used by the PSTN or PBX side of an FXS–FXO connection FXS

Description Label

Legacy Analog Interfaces in VoIP Networks

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Digital Interfaces

2048 kbps 64 kbps 1 channel (64 kbps) 30 E1 CCS 2048 kbps 64 kbps 64 kbps 30 E1 CAS 1544 kbps 8 kbps 1 channel (64 kbps) 23 T1 CCS 1544 kbps 8 kbps in-band (robbed-bits in voice channels) 24 (no clean 64 kbps because of

robbed-bit signaling) T1 CAS 192 kbps 48 kbps 1 channel (16 kbps) 2 BRI Total Bandwidth Framing Overhead Signaling

Voice Channels (64 kbps Each) Interface

© 2006 Cisco Systems, Inc. All rights reserved.

Digitizing and

Packetizing Voice

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Basic Voice Encoding: Converting Analog

Signals to Digital Signals

ƒ Step 1: Sample the analog signal.

ƒ Step 2: Quantize sample into a binary expression.

ƒ Step 3: Compress the samples to reduce bandwidth.

Basic Voice Encoding:

Converting Digital Signals to Analog Signals

ƒ Step 1: Decompress the samples.

ƒ Step 2: Decode the samples into voltage amplitudes, rebuilding the PAM signal.

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Determining Sampling Rate with the Nyquist

Theorem

ƒ The sampling rate affects the quality of the digitized signal.

ƒ Applying the Nyquist theorem determines the minimum sampling rate of analog signals.

ƒ Nyquist theorem requires that the sampling rate has to be at least twice the maximum frequency.

© 2006 Cisco Systems, Inc. All rights reserved.

Example: Setting the Correct Voice Sampling

Rate

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Human speech uses 200–9000 Hz.

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Human ear can sense 20–20,000 Hz.

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Traditional telephony systems were designed for

300–3400 Hz.

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Sampling rate for digitizing voice was set to 8000

samples per second, allowing frequencies up to 4000

Hz.

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Quantization

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Quantization is the representation of amplitudes by a

certain value (step).

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A scale with 256 steps is used for quantization.

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Samples are rounded up or down to the closer step.

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Rounding introduces inexactness (quantization noise).

Digital Voice Encoding

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Each sample is encoded using eight bits:

One polarity bit Three segment bits Four step bits

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Required bandwidth for one call is 64 kbps

(8000 samples per second, 8 bits each).

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Circuit-based telephony networks use TDM to combine

multiple 64-kbps channels (DS-0) to a single physical

line.

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Companding

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Companding — compressing and expanding

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There are two methods of companding:

Mu-law, used in Canada, U.S., and Japan A-law, used in other countries

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Both methods use a quasi-logarithmic scale:

Logarithmic segment sizes

Linear step sizes (within a segment)

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Both methods have eight positive and eight negative

segments, with 16 steps per segment.

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An international connection needs to use A-law;

mu-to-A conversion is the responsibility of the mu-law country.

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Coding

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Pulse Code Modulation (PCM)

Digital representation of analog signal Signal is sampled regularly at uniform levels Basic PCM samples voice 8000 times per second Basis for the entire telephone system digital hierarchy

ƒ

Adaptive Differential Pulse Code Modulation

Replaces PCM

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Common Voice Codec Characteristics

8 CS-ACELP, but with less

computation G.729A 8 CS-ACELP G.729 16 LDCELP (Low Delay CELP)

G.728 16, 24, 32 ADPCM G.726 64 PCM G.711 Bit Rate (kbps) Codec ITU-T Standard

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A Closer Look at a DSP

A DSP is a specialized processor

used for telephony applications:

ƒ

Voice termination:

Works as a compander converting analog voice to digital format and back again

Provides echo cancellation, VAD, CNG, jitter removal, and other benefits

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Conferencing: Mixes incoming

streams from multiple parties

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Transcoding: Translates between

voice streams that use different,

incompatible codecs

DSP Module

Voice Network Module

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DSP Used for Conferencing

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DSPs can be used in

single- or mixed-mode

conferences:

Mixed mode supports different codecs.

Single mode demands that the same codec to be used by all participants.

ƒ

Mixed mode has fewer

conferences per DSP.

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Example: DSP Used for Transcoding

Encapsulating Voice

Packets for

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Voice Transport in Circuit-Switched Networks

ƒ Analog phones connect to CO switches.

ƒ CO switches convert between analog and digital.

ƒ After call is set up, PSTN provides:

End-to-end dedicated circuit for this call (DS-0)

Synchronous transmission with fixed bandwidth and very low, constant delay

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Voice Transport in VoIP Networks

ƒ Analog phones connect to voice gateways.

ƒ Voice gateways convert between analog and digital.

ƒ After call is set up, IP network provides:

Packet-by-packet delivery through the network Shared bandwidth, higher and variable delays

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Jitter

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Voice packets enter the network at a constant rate.

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Voice packets may arrive at the destination at a

different rate or in the wrong order.

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Jitter occurs when packets arrive at varying rates.

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Since voice is dependent on timing and order, a

process must exist so that delays and queuing issues

can be fixed at the receiving end.

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The receiving router must:

Ensure steady delivery (delay)

Ensure that the packets are in the right order

VoIP Protocol Issues

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IP does not guarantee reliability, flow control, error

detection or error correction.

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IP can use the help of transport layer protocols TCP or

UDP.

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TCP offers reliability, but voice doesn’t need it…do not

retransmit lost voice packets.

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TCP overhead for reliability consumes bandwidth.

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UDP does not offer reliability. But it also doesn’t offer

sequencing…voice packets need to be in the right

order.

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RTP, which is built on UDP, offers all of the functionality

required by voice packets.

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Protocols Used for VoIP

Low

9

Low

9

Contains unnecessary information As little as possible Overhead No Yes

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Yes

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Yes Multiplexing Yes

9

No No Yes Time-stamping Yes

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No Yes

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Yes Reordering No

9

No

9

Yes No Reliability RTP UDP TCP Voice Needs Feature

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Voice Encapsulation

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Digitized voice is encapsulated into RTP, UDP, and IP.

ƒ

By default, 20 ms of voice is packetized into a single IP

packet.

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Voice Encapsulation Overhead

ƒ Voice is sent in small packets at high packet rates.

ƒ IP, UDP, and RTP header overheads are enormous:

For G.729, the headers are twice the size of the payload. For G.711, the headers are one-quarter the size of the payload. ƒ Bandwidth is 24 kbps for G.729 and 80 kbps for G.711, ignoring

Layer 2 overhead.

RTP Header Compression

ƒ Compresses the IP, UDP, and RTP headers

ƒ Is configured on a link-by-link basis

ƒ Reduces the size of the headers substantially (from 40 bytes to 2 or 4 bytes):

4 bytes if the UDP checksum is preserved 2 bytes if the UDP checksum is not sent ƒ Saves a considerable amount of bandwidth

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cRTP Operation

The sending side sends the entire header without compression.

There is an

unexpected change.

The receiving side substitutes the original stored header and calculates the

changed fields. The receiving side

predicts what the constant change is.

The sending side sends a hash of the header.

The predicted change is tracked.

The sending side tracks the predicted change.

The change is predictable.

Action Condition

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When to Use RTP Header Compression

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Use cRTP:

Only on slow links (less than 2 Mbps)

If bandwidth needs to be conserved

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Consider the disadvantages of cRTP:

Adds to processing overhead

Introduces additional delays

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Tune cRTP—set the number of sessions to be

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Calculating

Bandwidth

Requirements for

VoIP

Factors Influencing Encapsulation Overhead

and Bandwidth

– Depends on protocol (IPsec, GRE, or MPLS)

Tunneling overhead(if used)

– Depends on protocol (different per link)

Data-link overhead

– Depends on the use of cRTP

IP overhead

(including UDP and RTP)

– Depends on packetization period

– Depends on codec bandwidth

(bits per sample) Packetization size

(payload size)

– Derived from packetization period(the period over which encoded voice bits are collected for encapsulation)

Packet rate

Description Factor

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Bandwidth Implications of Codecs

ƒ Codec bandwidth is for voice information only. ƒ No packetization overhead is included. 8 kbps G.729 16 kbps G.728 16 kbps G.726 r16 24 kbps G.726 r24 32 kbps G.726 r32 64 kbps G.711 Bandwidth Codec

© 2006 Cisco Systems, Inc. All rights reserved.

How the Packetization Period Impacts VoIP

Packet Size and Rate

ƒ High packetization period results in:

Larger IP packet size (adding to the payload) Lower packet rate (reducing the IP overhead)

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VoIP Packet Size and Packet Rate Examples

25 50 33.33 50 Packet rate (pps) 80 60 280 200

VoIP packet size (bytes) 40 40 40 40 IP overhead (bytes) 40 20 240 160 Packetization size (bytes) 8 8 64 64 Codec bandwidth (kbps) G.729 40 ms G.729 20 ms G.711 30 ms G.711 20 ms Codec and Packetization Period

Data-Link Overhead Is Different per Link

6 Frame Relay 6 MLP 22 18 Overhead [bytes] Ethernet Trunk (802.1Q) Ethernet Data-Link Protocol

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Security and Tunneling Overhead

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IP packets can be secured by IPsec.

ƒ

Additionally, IP packets or data-link frames can be

tunneled over a variety of protocols.

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Characteristics of IPsec and tunneling protocols are:

The original frame or packet is encapsulated into another protocol.

The added headers result in larger packets and higher bandwidth requirements.

The extra bandwidth can be extremely critical for voice packets because of the transmission of small packets at a

high rate.

© 2006 Cisco Systems, Inc. All rights reserved.

Extra Headers in Security and Tunneling

Protocols

8

PPPoE

4

MPLS

24

L2TP/GRE

50–73

IPsec tunnel mode

30–53

IPsec transport mode

Header Size (bytes)

Protocol

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Example: VoIP over IPsec VPN

ƒ G.729 codec (8 kbps)

ƒ 20-ms packetization period

ƒ No cRTP

ƒ IPsec ESP with 3DES and SHA-1, tunnel mode

Total Bandwidth Required for a VoIP Call

ƒ Total bandwidth of a VoIP call, as seen on the link, is important for:

Designing the capacity of the physical link Deploying Call Admission Control (CAC) Deploying QoS

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Total Bandwidth Calculation Procedure

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Gather required packetization information:

Packetization period (default is 20 ms) or size Codec bandwidth

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Gather required information about the link:

cRTP enabled

Type of data-link protocol

IPsec or any tunneling protocols used

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Calculate the packetization size or period.

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Sum up packetization size and all headers and trailers.

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Calculate the packet rate.

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Calculate the total bandwidth.

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© 2006 Cisco Systems, Inc. All rights reserved.

Quick Bandwidth Calculation

Total packet size Total bandwidth requirement ————————— = ————————————————

Payload size Nominal bandwidth requirement

Total packet size = All headers + payload

8 kbps for G.729, 64 kbps for G.711 Nominal bandwidth

20 bytes for G.729, 160 bytes for G.711 Payload size (20-ms sample interval)

40 bytes IP + UDP + RTP headers 6 to 18 bytes Layer 2 header Value Parameter

Example: G.729 with Frame Relay:

Total bandwidth requirement = (6 + 40 + 20 bytes) * 8 kbps

————————————— = 26.4 kbps 20 bytes

VAD Characteristics

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Detects silence (speech pauses)

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Suppresses transmission of “silence patterns”

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Depends on multiple factors:

Type of audio (for example, speech or MoH) Level of background noise

Other factors (for example, language, character of speaker, or type of call)

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VAD Bandwidth-Reduction Examples

6.24 kbps 17.16 kbps 43.33 kbps 56.68 kbps Bandwidth with VAD (35% reduction) 9.6 kbps 26.4 kbps 66.67 kbps 87.2 kbps Bandwidth without VAD 40 ms 40 bytes 20 ms 20 bytes 30 ms 240 bytes 20 ms 160 bytes Packetization G.729 8 kbps G.729 8 kbps G.711 64 kbps G.711 64 kbps Codec cRTP 2 bytes no cRTP 40 bytes cRTP 4 bytes no cRTP 40 bytes IP overhead MLPP 6 bytes Frame Relay 6 bytes Frame Relay 6 bytes Ethernet 18 bytes Data-Link Overhead

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Traditional Nonconverged Network

ƒ Traditional data traffic characteristics:

Bursty data flow FIFO access

Not overly time-sensitive; delays OK Brief outages are survivable

Converged Network Realities

ƒ Converged network realities:

Constant small-packet voice flow competes with bursty data flow.

Critical traffic must have priority. Voice and video are time-sensitive. Brief outages are not acceptable.

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Converged Network Quality Issues

ƒ

Lack of bandwidth:

Multiple flows compete for a limited

amount of bandwidth.

ƒ

End-to-end delay (fixed and variable):

Packets have to

traverse many network devices and links; this travel

adds up to the overall delay.

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Variation of delay (jitter):

Sometimes there is a lot of

other traffic, which results in varied and increased

delay.

ƒ

Packet loss:

Packets may have to be dropped when a

link is congested.

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Measuring Available Bandwidth

ƒ The maximum available bandwidth is the bandwidth of the slowest link.

ƒ Multiple flows are competing for the same bandwidth, resulting in much less bandwidth being available to one single application.

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Increasing Available Bandwidth

ƒ Upgrade the link (the best but also the most expensive solution).

ƒ Improve QoS with advanced queuing mechanisms to forward the important packets first.

ƒ Compress the payload of Layer 2 frames (takes time).

ƒ Compress IP packet headers.

Using Available Bandwidth Efficiently

ƒ Using advanced queuing and header compression mechanisms, the available bandwidth can be used more efficiently:

Voice:LLQ and RTP header compression

Interactive traffic:CBWFQ and TCP header compression

Data (Low) Data (Medium) Data (High) Voice (Highest) 1 1 2 2 3 3 3 4 4 4 4 4 3 2 1 1 Voice • LLQ • RTP header compression Data • CBWFQ • TCP header compression

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Types of Delay

ƒ Processing delay:The time it takes for a router to take the packet from an input interface, examine the packet, and put the packet into the output queue of the output interface.

ƒ Queuing delay:The time a packet resides in the output queue of a router.

ƒ Serialization delay:The time it takes to place the “bits on the wire.”

ƒ Propagation delay:The time it takes for the packet to cross the link from one end to the other.

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The Impact of Delay and Jitter on Quality

ƒ End-to-end delay:The sum of all propagation, processing, serialization, and queuing delays in the path

ƒ Jitter:The variation in the delay.

ƒ In best-effort networks, propagation and serialization delays are fixed, while processing and queuing delays are unpredictable.

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Ways to Reduce Delay

ƒ Upgrade the link (the best solution but also the most expensive).

ƒ Forward the important packets first.

ƒ Enable reprioritization of important packets.

ƒ Compress the payload of Layer 2 frames (takes time).

ƒ Compress IP packet headers.

Reducing Delay in a Network

ƒ Customer routers perform:

TCP/RTP header compression LLQ

Prioritization ƒ ISP routers perform:

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The Impacts of Packet Loss

ƒ Telephone call:“I cannot understand you. Your voice is breaking up.”

ƒ Teleconferencing:“The picture is very jerky. Voice is not synchronized.”

ƒ Publishing company:“This file is corrupted.”

ƒ Call center:“Please hold while my screen refreshes.”

© 2006 Cisco Systems, Inc. All rights reserved.

Types of Packet Drops

ƒ Tail drops occur when the output queue is full. Tail drops are common and happen when a link is congested.

ƒ Other types of drops, usually resulting from router congestion, include input drop, ignore, overrun, and frame errors. These errors can often be solved with hardware upgrades.

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Ways to Prevent Packet Loss

ƒ Upgrade the link (the best solution but also the most expensive). ƒ Guarantee enough bandwidth for sensitive packets.

ƒ Prevent congestion by randomly dropping less important packets before congestion occurs.

Traffic Policing and Traffic Shaping

Time Traf fic Traffic Rate Time Traf

fic Traffic Rate Time Traf fic Traffic Rate Time Traf fic Traffic Rate Policing Shaping

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Reducing Packet Loss in a Network

ƒ Problem:Interface congestion causes TCP and voice packet drops, resulting in slowing FTP traffic and jerky speech quality.

ƒ Conclusion:Congestion avoidance and queuing can help.

ƒ Solution:Use WRED and LLQ.

© 2006 Cisco Systems, Inc. All rights reserved.

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© 2006 Cisco Systems, Inc. All rights reserved.

What Is Quality of Service?

Two Perspectives

ƒ

The user perspective

Users perceive that their applications are performing properly

Voice, video, and data

ƒ

The network manager perspective

Need to manage bandwidth allocations to deliver the desired application performance

Control delay, jitter, and packet loss

Different Types of Traffic Have Different

Needs

Sensitivity to QoS Metrics N N N Bulk Data Email File Transfer N N Y Transactional/ Interactive Y Y N Streaming Video Y Y Y Interactive Voice and Video Packet Loss Jitter Delay Application Examples Need to manage bandwidth allocations ƒ Real-time applications especially

sensitive to QoS

Interactive voice Videoconferencing

ƒ Causes of degraded performance

Congestion losses Variable queuing delays

ƒ The QoS challenge

Manage bandwidth allocations to deliver the desired application performance

Control delay, jitter, and packet loss

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Implementing QoS

Step 1:Identify types of traffic and their requirements.

Step 2:Divide traffic into classes.

Step 3:Define QoS policies for each class.

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Step 2: Define Traffic Classes

Scavenger Class

Less than Best Effort

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Step 3: Define QoS Policy

ƒ

A QoS policy is a

network-wide definition of

the specific levels of QoS

that are assigned to

different classes of

network traffic.

Quality of Service Operations

How Do QoS Tools Work?

Classification and Marking Queuing and (Selective) Dropping Post-Queuing Operations

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Selecting an

Appropriate QoS

Policy Model

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Three QoS Models

The network recognizes classes that require QoS.

Differentiated Services (DiffServ)

Applications signal to the network that the applications require certain QoS parameters. Integrated

Services (IntServ)

No QoS is applied to packets. If it is not

important when or how packets arrive, the best-effort model is appropriate.

Best effort

Characteristics

Model

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Best-Effort Model

ƒ

Internet was initially based on a best-effort packet

delivery service.

ƒ

Best-effort is the default mode for all traffic.

ƒ

There is no differentiation among types of traffic.

ƒ

Best-effort model is similar to using standard mail—

“The mail will arrive when the mail arrives.”

ƒ

Benefits:

Highly scalable

No special mechanisms required

ƒ

Drawbacks:

No service guarantees No service differentiation

Integrated Services (IntServ) Model Operation

ƒ Ensures guaranteed delivery and

predictable behavior of the network for applications.

ƒ Provides multiple service levels.

ƒ RSVP is a signaling protocol to reserve resources for specified QoS parameters.

ƒ The requested QoS parameters are then linked to a packet stream.

ƒ Streams are not established if the required QoS parameters cannot be met.

ƒ Intelligent queuing mechanisms needed to provide resource reservation in terms of:

Guaranteed rate

Controlled load (low delay, high throughput)

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Benefits and Drawbacks of the IntServ Model

ƒ

Benefits:

Explicit resource admission control (end to end)

Per-request policy admission control (authorization object, policy object)

Signaling of dynamic port numbers (for example, H.323)

ƒ

Drawbacks:

Continuous signaling because of stateful architecture

Flow-based approach not scalable to large implementations, such as the public Internet

© 2006 Cisco Systems, Inc. All rights reserved.

The Differentiated Services Model

ƒ Overcomes many of the limitations best-effort and IntServ models

ƒ Uses the soft QoS provisioned-QoS model rather than the hard QoS signaled-QoS model

ƒ Classifies flows into aggregates (classes) and provides appropriate QoS for the classes

ƒ Minimizes signaling and state maintenance requirements on each network node

ƒ Manages QoS characteristics on the basis of per-hop behavior (PHB)

ƒ You choose the level of service for each traffic class

Edge Edge Interior Edge DiffServ Domain End Station End Station

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Implement the

DiffServ QoS Model

Lesson 4.1: Introducing Classification and Marking

Classification

ƒ

Classification is the process of identifying and

categorizing traffic into classes, typically based upon:

Incoming interface IP precedence DSCP

Source or destination address Application

ƒ

Without classification, all packets are treated the same.

ƒ

Classification should take place as close to the source

as possible.

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Marking

ƒ

Marking is the QoS feature component that “colors” a

packet (frame) so it can be identified and distinguished

from other packets (frames) in QoS treatment.

ƒ

Commonly used markers:

Link layer: CoS (ISL, 802.1p) MPLS EXP bits Frame Relay Network layer: DSCP IP precedence

© 2006 Cisco Systems, Inc. All rights reserved.

Classification and Marking in the LAN with

IEEE 802.1Q

ƒ IEEE 802.1p user priority field is also called CoS.

ƒ IEEE 802.1p supports up to eight CoSs. ƒ IEEE 802.1p focuses on support for

QoS over LANs and 802.1Q ports. ƒ IEEE 802.1p is preserved through the

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Classification and Marking in the Enterprise

DiffServ Model

ƒ

Describes services associated with traffic classes,

rather than traffic flows.

ƒ

Complex traffic classification and conditioning is

performed at the network edge.

ƒ

No per-flow state in the core.

ƒ

The goal of the DiffServ model is scalability.

ƒ

Interoperability with non-DiffServ-compliant nodes.

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Classification Tools

IP Precedence and DiffServ Code Points

ƒ

IPv4

: three most significant bits of ToS byte are called

IP Precedence (IPP)—other bits unused

ƒ

DiffServ

: six most significant bits of ToS byte are called

DiffServ Code Point (DSCP)—remaining two bits used

for flow control

ƒ

DSCP is backward-compatible with IP precedence

7 6 5 4 3 2 1 0

ID Offset TTL Proto FCS IP SA IP DA Data Len

Version Length

ToS Byte

DiffServ Code Point (DSCP) IP ECN

IPv4 Packet

IP Precedence Unused Standard IPv4

DiffServ Extensions

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IP Precedence and DSCP Compatibility

ƒ Compatibility with current IP precedence usage (RFC 1812)

ƒ Differentiates probability of timely forwarding:

(xyz000) >= (abc000) if xyz > abc

ƒ That is, if a packet has DSCP value of 011000, it has a greater probability of timely forwarding than a packet with DSCP value of 001000.

Per-Hop Behaviors

ƒ

DSCP selects PHB throughout the network:

DefaultPHB (FIFO, tail drop)

Class-selectorPHB (IP precedence)

EFPHB

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Standard PHB Groups

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Expedited Forwarding (EF) PHB

ƒ EF PHB:

Ensures a minimum departure rate

Guarantees bandwidth—class guaranteed an amount of bandwidth with prioritized forwarding

Polices bandwidth—class not allowed to exceed the guaranteed amount (excess traffic is dropped)

ƒ DSCP value of 101110:Looks like IP precedence 5 to non-DiffServ-compliant devices:

Bits 5 to 7: 101 = 5 (same 3 bits are used for IP precedence) Bits 3 and 4: 11 = No drop probability

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Assured Forwarding (AF) PHB

ƒ

AF PHB:

Guarantees bandwidth

Allows access to extra bandwidth, if available

ƒ

Four standard classes: AF1, AF2, AF3, and AF4

ƒ

DSCP value range of aaadd0:

aaais a binary value of the class

ddis drop probability

AF PHB Values

ƒ Each AF class uses three DSCP values.

ƒ Each AF class is independently forwarded with its guaranteed bandwidth.

ƒ Congestion avoidance is used within each class to prevent congestion within the class.

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Mapping CoS to Network Layer QoS

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QoS Service Class

ƒ

A QoS service class is a logical grouping of packets

that are to receive a similar level of applied quality.

ƒ

A QoS service class can be:

A single user (such as MAC address or IP address) A department, customer (such as subnet or interface) An application (such as port numbers or URL)

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Implementing QoS Policy Using a QoS Service

Class

QoS Service Class Guidelines

ƒ Profile applications to their basic network requirements.

ƒ Do not over engineer provisioning; use no more than four to five traffic classes for data traffic:

Voice applications: VoIP

Mission-critical applications: Oracle, SAP, SNA Interactive applications: Telnet, TN3270 Bulk applications: FTP, TFTP

Best-effort applications: E-mail, web

Scavenger applications: Nonorganizational streaming and video applications (Kazaa, Yahoo)

ƒ Do not assign more than three applications to mission-critical or transactional classes.

ƒ Use proactive policies before reactive (policing) policies.

ƒ Seek executive endorsement of relative ranking of application priority prior to rolling out QoS policies for data.

(48)

© 2006 Cisco Systems, Inc. All rights reserved.

Classification and Marking Design

QoS Baseline Marking Recommendations

Application L3 Classification

DSCP PHB

IPP CoS

Transactional Data 2 AF21 18 2

Call Signaling 3 CS3* 24 3

Streaming Video 4 CS4 32 4

Video Conferencing 4 AF41 34 4

Voice 5 EF 46 5

Network Management 2 CS2 16 2

L2

Bulk Data 1 AF11 10 1

Scavenger 1 CS1 8 1

Routing 6 CS6 48 6

Mission-Critical Data 3 AF31* 26 3

Best Effort 0 0 0 0

© 2006 Cisco Systems, Inc. All rights reserved.

How Many Classes of Service Do I Need?

4/5 Class Model Scavenger Critical Data Call Signaling Realtime 8 Class Model Critical Data Video Call Signaling Best Effort Voice Bulk Data Network Control Scavenger 11 Class Model Network Management Call Signaling Streaming Video Transactional Data Interactive-Video Voice Best Effort IP Routing Mission-Critical Data Scavenger Bulk Data Time Best Effort

(49)

© 2006 Cisco Systems, Inc. All rights reserved.

Trust Boundaries: Classify Where?

ƒ For scalability, classification should be enabled as close to the edge as possible, depending on the capabilities of the device at:

Endpoint or end system Access layer

Distribution layer

Trust Boundaries: Mark Where?

Figure

Updating...

References