© 2006 cisco systems, inc. all rights reserved

© 2006 Cisco Systems, Inc. All rights reserved.

Introducing VoIP Networks


Benefits of a VoIP Network More efficient use of bandwidth and equipment Lower transmission costs Consolidated network expenses 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 applicationSoftware-based and wireless phones offer mobility.

Access to new communications devices (such as PDAs and cable set-top boxes)


Components of a VoIP Network


Legacy Analog and VoIP Applications Can Coexist


Legacy Analog Interfaces in VoIP Networks

Analog Interface Type Label Description

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

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

Earth and Magneto E&M Trunk, used between switches


Legacy Analog Interfaces in VoIP Networks

1

1

23

4

5


Digital Interfaces

Interface Voice Channels (64 kbps Each) Signaling Framing Overhead

Total Bandwidth

BRI 2 1 channel (16 kbps) 48 kbps 192 kbps

T1 CAS 24 (no clean 64 kbps because of robbed-bit signaling)

in-band (robbed-bits in voice channels)

8 kbps 1544 kbps

T1 CCS 23 1 channel (64 kbps) 8 kbps 1544 kbps

E1 CAS 30 64 kbps 64 kbps 2048 kbps

E1 CCS 30 1 channel (64 kbps) 64 kbps 2048 kbps


Digitizing and Packetizing Voice


Basic Voice Encoding: Converting Analog Signals to Digital Signals

Step 1: Sample the analog signal.

Step 2: Quantize sample into a binary expression.

Step 3: Compress the samples to reduce bandwidth.


Basic Voice Encoding:Converting Digital Signals to Analog Signals

Step 1: Decompress the samples.

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

Step 3: Reconstruct the analog signal from the PAM signals.


Determining Sampling Rate with the Nyquist Theorem The sampling rate affects the quality of the digitized signal.

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

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


Example: Setting the Correct Voice Sampling Rate Human speech uses 200–9000 Hz.

Human ear can sense 20–20,000 Hz.

Traditional telephony systems were designed for 300–3400 Hz.

Sampling rate for digitizing voice was set to 8000 samples per second, allowing frequencies up to 4000 Hz.


Quantization Quantization is the representation of amplitudes by a

certain value (step).

A scale with 256 steps is used for quantization.

Samples are rounded up or down to the closer step.

Rounding introduces inexactness (quantization noise).


Digital Voice Encoding Each sample is encoded using eight bits:

One polarity bitThree segment bitsFour step bits

Required bandwidth for one call is 64 kbps (8000 samples per second, 8 bits each).

Circuit-based telephony networks use TDM to combine multiple 64-kbps channels (DS-0) to a single physical line.


Companding Companding — compressing and expanding

There are two methods of companding:Mu-law, used in Canada, U.S., and JapanA-law, used in other countries

Both methods use a quasi-logarithmic scale:Logarithmic segment sizesLinear step sizes (within a segment)

Both methods have eight positive and eight negative segments, with 16 steps per segment.

An international connection needs to use A-law; mu-to-A conversion is the responsibility of the mu-law country.


Coding Pulse Code Modulation (PCM)

Digital representation of analog signalSignal is sampled regularly at uniform levelsBasic PCM samples voice 8000 times per secondBasis for the entire telephone system digital hierarchy

Adaptive Differential Pulse Code ModulationReplaces PCMTransmits only the difference between one sample and the next


Common Voice Codec Characteristics

ITU-T Standard Codec Bit Rate (kbps)

G.711 PCM 64

G.726 ADPCM 16, 24, 32

G.728 LDCELP (Low Delay CELP) 16

G.729 CS-ACELP 8

G.729A CS-ACELP, but with less computation 8


Mean Opinion Score


A Closer Look at a DSPA DSP is a specialized processor

used for telephony applications:

Voice termination:Works as a compander converting analog voice to digital format and back againProvides echo cancellation, VAD, CNG, jitter removal, and other benefits

Conferencing: Mixes incoming streams from multiple parties

Transcoding: Translates between voice streams that use different, incompatible codecs

DSP Module

Voice Network Module


DSP Used for Conferencing 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.

Mixed mode has fewer conferences per DSP.


Example: DSP Used for Transcoding


Encapsulating Voice Packets for Transport


Voice Transport in Circuit-Switched Networks

Analog phones connect to CO switches. CO switches convert between analog and digital. 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


Voice Transport in VoIP Networks

Analog phones connect to voice gateways. Voice gateways convert between analog and digital. After call is set up, IP network provides:

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


Jitter Voice packets enter the network at a constant rate.

Voice packets may arrive at the destination at a different rate or in the wrong order.

Jitter occurs when packets arrive at varying rates.

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.

The receiving router must:Ensure steady delivery (delay)Ensure that the packets are in the right order


VoIP Protocol Issues IP does not guarantee reliability, flow control, error

detection or error correction.

IP can use the help of transport layer protocols TCP or UDP.

TCP offers reliability, but voice doesn’t need it…do not retransmit lost voice packets.

TCP overhead for reliability consumes bandwidth.

UDP does not offer reliability. But it also doesn’t offer sequencing…voice packets need to be in the right order.

RTP, which is built on UDP, offers all of the functionality required by voice packets.


Protocols Used for VoIP

Feature Voice Needs TCP UDP RTP

Reliability No Yes No No

Reordering Yes Yes No YesTime-

stamping Yes No No Yes

Overhead As little as possible

Contains unnecessary information Low Low

Multiplexing Yes Yes Yes No


Voice Encapsulation

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

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


Voice Encapsulation Overhead

Voice is sent in small packets at high packet rates.

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.

Bandwidth is 24 kbps for G.729 and 80 kbps for G.711, ignoring Layer 2 overhead.


RTP Header Compression

Compresses the IP, UDP, and RTP headers

Is configured on a link-by-link basis

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

4 bytes if the UDP checksum is preserved2 bytes if the UDP checksum is not sent

Saves a considerable amount of bandwidth


cRTP Operation

Condition Action

The change is predictable.

The sending side tracks the predicted change.

The predicted change is tracked.

The sending side sends a hash of the header.

The receiving side predicts what the constant change is.

The receiving side substitutes the original stored header and calculates the changed fields.

There is an unexpected change.

The sending side sends the entire header without compression.


When to Use RTP Header Compression Use cRTP:

Only on slow links (less than 2 Mbps)If bandwidth needs to be conserved

Consider the disadvantages of cRTP:Adds to processing overheadIntroduces additional delays

Tune cRTP—set the number of sessions to be compressed (default is 16).


Calculating Bandwidth Requirements for VoIP


Factors Influencing Encapsulation Overhead and Bandwidth

Factor Description

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

Packetization size (payload size)

– Depends on packetization period– Depends on codec bandwidth

(bits per sample)

IP overhead (including UDP and RTP)

– Depends on the use of cRTP

Data-link overhead – Depends on protocol (different per link)

Tunneling overhead (if used)

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


Bandwidth Implications of Codecs Codec bandwidth is for voice

information only.

No packetization overhead is included.

Codec Bandwidth

G.711 64 kbps

G.726 r32 32 kbps

G.726 r24 24 kbps

G.726 r16 16 kbps

G.728 16 kbps

G.729 8 kbps


How the Packetization Period Impacts VoIP Packet Size and Rate High packetization period results in:

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


VoIP Packet Size and Packet Rate Examples

Codec andPacketization Period

G.711 20 ms

G.711 30 ms

G.729 20 ms

G.729 40 ms

Codec bandwidth (kbps) 64 64 8 8

Packetization size (bytes) 160 240 20 40

IP overhead(bytes) 40 40 40 40

VoIP packet size (bytes) 200 280 60 80

Packet rate(pps) 50 33.33 50 25


Data-Link Overhead Is Different per Link

Data-Link Protocol Ethernet Frame

Relay MLP Ethernet Trunk (802.1Q)

Overhead [bytes] 18 6 6 22


Security and Tunneling Overhead IP packets can be secured by IPsec.

Additionally, IP packets or data-link frames can be tunneled over a variety of protocols.

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.


Extra Headers in Security and Tunneling Protocols

Protocol Header Size (bytes)

IPsec transport mode 30–53

IPsec tunnel mode 50–73

L2TP/GRE 24

MPLS 4

PPPoE 8


Example: VoIP over IPsec VPN G.729 codec (8 kbps)

20-ms packetization period

No cRTP

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


Total Bandwidth Required for a VoIP Call

Total bandwidth of a VoIP call, as seen on the link, is important for:Designing the capacity of the physical linkDeploying Call Admission Control (CAC)Deploying QoS


Total Bandwidth Calculation Procedure Gather required packetization information:

Packetization period (default is 20 ms) or sizeCodec bandwidth

Gather required information about the link:cRTP enabledType of data-link protocolIPsec or any tunneling protocols used

Calculate the packetization size or period.

Sum up packetization size and all headers and trailers.

Calculate the packet rate.

Calculate the total bandwidth.


Bandwidth Calculation Example


Quick Bandwidth Calculation Total packet size Total bandwidth requirement

————————— = ————————————————

Payload size Nominal bandwidth requirement

Total packet size = All headers + payload

Parameter Value

Layer 2 header 6 to 18 bytes

IP + UDP + RTP headers 40 bytes

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

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

Example: G.729 with Frame Relay:

Total bandwidth requirement = (6 + 40 + 20 bytes) * 8 kbps ————————————— = 26.4 kbps 20 bytes


VAD Characteristics Detects silence (speech pauses)

Suppresses transmission of “silence patterns”

Depends on multiple factors:Type of audio (for example, speech or MoH)Level of background noiseOther factors (for example, language, character of speaker, or type of call)

Can save up to 35 percent of bandwidth


VAD Bandwidth-Reduction ExamplesData-Link Overhead

Ethernet

18 bytes

Frame Relay

6 bytes

Frame Relay

6 bytes

MLPP

6 bytes

IP overhead no cRTP

40 bytes

cRTP

4 bytes

no cRTP

40 bytes

cRTP

2 bytes

Codec G.711

64 kbps

G.711

64 kbps

G.729

8 kbps

G.729

8 kbps

Packetization 20 ms

160 bytes

30 ms

240 bytes

20 ms

20 bytes

40 ms

40 bytes

Bandwidth without VAD

87.2 kbps 66.67 kbps 26.4 kbps 9.6 kbps

Bandwidth with VAD (35% reduction)

56.68 kbps 43.33 kbps 17.16 kbps 6.24 kbps


Introducing QoS


Traditional Nonconverged Network

Traditional data traffic characteristics:Bursty data flowFIFO access Not overly time-sensitive; delays OK Brief outages are survivable

http://www.cisco.com/#www.cisco.com


Converged Network Realities

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.


Converged Network Quality Issues Lack of bandwidth: Multiple flows compete for a limited

amount of bandwidth.

End-to-end delay (fixed and variable): Packets have to traverse many network devices and links; this travel adds up to the overall delay.

Variation of delay (jitter): Sometimes there is a lot of other traffic, which results in varied and increased delay.

Packet loss: Packets may have to be dropped when a link is congested.


Measuring Available Bandwidth

The maximum available bandwidth is the bandwidth of the slowest link. Multiple flows are competing for the same bandwidth, resulting in much less

bandwidth being available to one single application. A lack in bandwidth can have performance impacts on network applications.


Increasing Available Bandwidth

Upgrade the link (the best but also the most expensive solution). Improve QoS with advanced queuing mechanisms to forward the important packets first. Compress the payload of Layer 2 frames (takes time). Compress IP packet headers.


Using Available Bandwidth Efficiently

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

Voice: LLQ and RTP header compressionInteractive traffic: CBWFQ and TCP header compression

Voice(Highest)

Data(High)

Data(Medium)

Data(Low)

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


Types of Delay

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.

Queuing delay: The time a packet resides in the output queue of a router. Serialization delay: The time it takes to place the “bits on the wire.” Propagation delay: The time it takes for the packet to cross the link from one end to the other.


The Impact of Delay and Jitter on Quality

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

Jitter: The variation in the delay.

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


Ways to Reduce Delay

Upgrade the link (the best solution but also the most expensive). Forward the important packets first. Enable reprioritization of important packets. Compress the payload of Layer 2 frames (takes time). Compress IP packet headers.


Reducing Delay in a Network

Customer routers perform:TCP/RTP header compressionLLQPrioritization

ISP routers perform:Reprioritization according to the QoS policy


The Impacts of Packet Loss

Telephone call: “I cannot understand you. Your voice is breaking up.” Teleconferencing: “The picture is very jerky. Voice is not synchronized.” Publishing company: “This file is corrupted.” Call center: “Please hold while my screen refreshes.”


Types of Packet Drops

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

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.


Ways to Prevent Packet Loss

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

Guarantee enough bandwidth for sensitive packets.

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


Reducing Packet Loss in a Network

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

Conclusion: Congestion avoidance and queuing can help.

Solution: Use WRED and LLQ.


Implementing QoS


What Is Quality of Service? Two Perspectives

The user perspectiveUsers perceive that their applications are performing properly

Voice, video, and data

The network manager perspectiveNeed to manage bandwidth allocations to deliver the desired application performance

Control delay, jitter, andpacket loss


Different Types of Traffic Have Different Needs

Application Examples

Sensitivity to QoS Metrics

Delay Jitter Packet Loss

Interactive Voice and Video Y Y Y

Streaming Video N Y Y

Transactional/ Interactive Y N N

Bulk DataEmail

File TransferN N N

Need to managebandwidth allocations

Real-time applications especially sensitive to QoS

Interactive voiceVideoconferencing

Causes of degraded performanceCongestion lossesVariable queuing delays

The QoS challengeManage bandwidth allocations to deliver the desired application performanceControl delay, jitter, and packet loss


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.


Step 2: Define Traffic Classes

Scavenger Class

Less than Best Effort


Step 3: Define QoS Policy 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 OperationsHow Do QoS Tools Work?

Classification and Marking

Queuing and (Selective) Dropping

Post-Queuing Operations


Selecting an Appropriate QoS Policy Model


Three QoS Models

Model CharacteristicsBest effort No QoS is applied to packets. If it is not

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

Integrated Services

(IntServ)

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

Differentiated Services

(DiffServ)

The network recognizes classes that require QoS.


Best-Effort Model Internet was initially based on a best-effort packet

delivery service.

Best-effort is the default mode for all traffic.

There is no differentiation among types of traffic.

Best-effort model is similar to using standard mail—“The mail will arrive when the mail arrives.”

Benefits:Highly scalableNo special mechanisms required

Drawbacks:No service guaranteesNo service differentiation


Integrated Services (IntServ) Model Operation Ensures guaranteed delivery and

predictable behavior of the network for applications.

Provides multiple service levels.

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

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

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

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

Guaranteed rateControlled load (low delay, high throughput)


Benefits and Drawbacks of the IntServ Model 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)

Drawbacks:Continuous signaling because of stateful architectureFlow-based approach not scalable to large implementations, such as the public Internet


The Differentiated Services Model

Overcomes many of the limitations best-effort and IntServ models Uses the soft QoS provisioned-QoS model rather than the hard QoS

signaled-QoS model Classifies flows into aggregates (classes) and provides appropriate QoS for

the classes Minimizes signaling and state maintenance requirements on each network

node Manages QoS characteristics on the basis of per-hop behavior (PHB) You choose the level of service for each traffic class

Edge

Edge

Interior

Edge

DiffServ Domain

End Station

End Station


Implement the DiffServ QoS Model

Lesson 4.1: Introducing Classification and Marking


Classification Classification is the process of identifying and

categorizing traffic into classes, typically based upon:Incoming interfaceIP precedenceDSCPSource or destination addressApplication

Without classification, all packets are treated the same.

Classification should take place as close to the source as possible.


Marking 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.

Commonly used markers:Link layer:

CoS (ISL, 802.1p)MPLS EXP bitsFrame Relay

Network layer:DSCPIP precedence


Classification and Marking in the LAN with IEEE 802.1Q

IEEE 802.1p user priority field is also called CoS.

IEEE 802.1p supports up to eight CoSs.

IEEE 802.1p focuses on support for QoS over LANs and 802.1Q ports.

IEEE 802.1p is preserved through the LAN, not end to end.


Classification and Marking in the Enterprise


DiffServ Model Describes services associated with traffic classes,

rather than traffic flows.

Complex traffic classification and conditioning is performed at the network edge.

No per-flow state in the core.

The goal of the DiffServ model is scalability.

Interoperability with non-DiffServ-compliant nodes.

Incremental deployment.


Classification ToolsIP Precedence and DiffServ Code Points

IPv4: three most significant bits of ToS byte are called IP Precedence (IPP)—other bits unused

DiffServ: six most significant bits of ToS byte are called DiffServ Code Point (DSCP)—remaining two bits used for flow control

DSCP is backward-compatible with IP precedence

7 6 5 4 3 2 1 0

ID Offset TTL Proto FCS IP SA IP DA DataLenVersion Length

ToSByte

DiffServ Code Point (DSCP) IP ECN

IPv4 Packet

IP Precedence UnusedStandard IPv4

DiffServ Extensions


IP ToS Byte and DS Field Inside the IP Header


IP Precedence and DSCP Compatibility

Compatibility with current IP precedence usage (RFC 1812)

Differentiates probability of timely forwarding:(xyz000) >= (abc000) if xyz > abc

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

DSCP selects PHB throughout the network:Default PHB (FIFO, tail drop)Class-selector PHB (IP precedence) EF PHBAF PHB


Standard PHB Groups


Expedited Forwarding (EF) PHB

EF PHB:Ensures a minimum departure rateGuarantees bandwidth—class guaranteed an amount of bandwidth with prioritized forwardingPolices bandwidth—class not allowed to exceed the guaranteed amount (excess traffic is dropped)

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 probabilityBit 2: Just 0


Assured Forwarding (AF) PHB

AF PHB:Guarantees bandwidthAllows access to extra bandwidth, if available

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

DSCP value range of aaadd0:aaa is a binary value of the classdd is drop probability


AF PHB Values

Each AF class uses three DSCP values.

Each AF class is independently forwarded with its guaranteed bandwidth.

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


Mapping CoS to Network Layer QoS


QoS Service Class A QoS service class is a logical grouping of packets

that are to receive a similar level of applied quality.

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)A network destination (such as tunnel interface or VPN)


Implementing QoS Policy Using a QoS Service Class


QoS Service Class Guidelines Profile applications to their basic network requirements.

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

Voice applications: VoIPMission-critical applications: Oracle, SAP, SNAInteractive applications: Telnet, TN3270Bulk applications: FTP, TFTPBest-effort applications: E-mail, webScavenger applications: Nonorganizational streaming and video applications (Kazaa, Yahoo)

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

Use proactive policies before reactive (policing) policies.

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


Classification and Marking DesignQoS Baseline Marking Recommendations

ApplicationL3 Classification

DSCPPHBIPP CoS

Transactional Data 18AF212 2

Call Signaling 24CS3*3 3

Streaming Video 32CS44 4

Video Conferencing 34AF414 4

Voice 46EF5 5

Network Management 16CS22 2

L2

Bulk Data 10AF111 1

Scavenger 8CS11 1

Routing 48CS66 6

Mission-Critical Data 26AF31*3 3

Best Effort 000 0


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 SignalingStreaming Video

Transactional Data

Interactive-VideoVoice

Best Effort

IP Routing

Mission-Critical Data

Scavenger

Bulk Data

Time

Best Effort


Trust Boundaries: Classify Where?

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

Endpoint or end systemAccess layerDistribution layer


Trust Boundaries: Mark Where?

For scalability, marking should be done as close to the source as possible.

© 2006 cisco systems, inc. all rights reserved

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