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© 2006 Cisco Systems, Inc. All rights reserved.
Introducing VoIP Networks
© 2006 Cisco Systems, Inc. All rights reserved.
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 application
Software-based and wireless phones offer mobility.
Access to new communications devices (such as PDAs and cable set-top boxes)
© 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
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
Legacy Analog Interfaces in VoIP Networks
1
1
23
4
5
© 2006 Cisco Systems, Inc. All rights reserved.
Digital Interfaces
Interface Voice Channels (64 kbps Each) SignalingFraming 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
© 2006 Cisco Systems, Inc. All rights reserved.
Digitizing and Packetizing Voice
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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).
© 2006 Cisco Systems, Inc. All rights reserved.
Digital Voice Encoding Each sample is encoded using eight bits:
One polarity bit
Three segment bits
Four 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.
© 2006 Cisco Systems, Inc. All rights reserved.
Companding Companding — compressing and expanding
There are two methods of companding:Mu-law, used in Canada, U.S., and Japan
A-law, used in other countries
Both methods use a quasi-logarithmic scale:Logarithmic segment sizes
Linear 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.
© 2006 Cisco Systems, Inc. All rights reserved.
Coding 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
Adaptive Differential Pulse Code ModulationReplaces PCM
Transmits only the difference between one sample and the next
© 2006 Cisco Systems, Inc. All rights reserved.
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.729ACS-ACELP, but with less computation
8
© 2006 Cisco Systems, Inc. All rights reserved.
Mean Opinion Score
© 2006 Cisco Systems, Inc. All rights reserved.
A Closer Look at a DSP
A DSP is a specialized processor used for telephony applications:
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
Conferencing: Mixes incoming streams from multiple parties
Transcoding: Translates between voice streams that use different, incompatible codecs
DSP Module
Voice Network Module
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Example: DSP Used for Transcoding
© 2006 Cisco Systems, Inc. All rights reserved.
Encapsulating Voice Packets for Transport
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
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 network
Shared bandwidth, higher and variable delays
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Protocols Used for VoIP
FeatureVoice Needs
TCP UDP RTP
Reliability No Yes No No
Reordering Yes Yes No YesTime-
stampingYes No No Yes
Overhead As little as possible
Contains unnecessary information
Low Low
Multiplexing Yes Yes Yes No
© 2006 Cisco Systems, Inc. All rights reserved.
Voice Encapsulation
Digitized voice is encapsulated into RTP, UDP, and IP.
By default, 20 ms of voice is packetized into a single IP packet.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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 preserved
2 bytes if the UDP checksum is not sent
Saves a considerable amount of bandwidth
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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 overhead
Introduces additional delays
Tune cRTP—set the number of sessions to be compressed (default is 16).
© 2006 Cisco Systems, Inc. All rights reserved.
Calculating Bandwidth Requirements for VoIP
© 2006 Cisco Systems, Inc. All rights reserved.
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)
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
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)
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
Data-Link Overhead Is Different per Link
Data-Link Protocol
EthernetFrame Relay
MLPEthernet Trunk
(802.1Q)
Overhead [bytes]
18 6 6 22
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
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 link
Deploying Call Admission Control (CAC)
Deploying QoS
© 2006 Cisco Systems, Inc. All rights reserved.
Total Bandwidth Calculation Procedure Gather required packetization information:
Packetization period (default is 20 ms) or size
Codec bandwidth
Gather required information about the link:cRTP enabled
Type of data-link protocol
IPsec 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.
© 2006 Cisco Systems, Inc. All rights reserved.
Bandwidth Calculation Example
© 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
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
© 2006 Cisco Systems, Inc. All rights reserved.
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 noise
Other factors (for example, language, character of speaker, or type of call)
Can save up to 35 percent of bandwidth
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
Introducing QoS
© 2006 Cisco Systems, Inc. All rights reserved.
Traditional Nonconverged Network
Traditional data traffic characteristics:Bursty data flow
FIFO access
Not overly time-sensitive; delays OK
Brief outages are survivable
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Using Available Bandwidth Efficiently
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
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
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Reducing Delay in a Network
Customer routers perform:TCP/RTP header compression
LLQ
Prioritization
ISP routers perform:Reprioritization according to the QoS policy
© 2006 Cisco Systems, Inc. All rights reserved.
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.”
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Traffic Policing and Traffic Shaping
Time
Tra
ffic
Traffic Rate
Time
Tra
ffic Traffic Rate
Time
Tra
ffic
Traffic Rate
Time
Tra
ffic
Traffic Rate
Policing
Shaping
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Implementing QoS
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
Different Types of Traffic Have Different Needs
Application Examples
Sensitivity to QoS Metrics
Delay JitterPacket 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 voice
Videoconferencing
Causes of degraded performance
Congestion losses
Variable queuing delays
The QoS challenge
Manage bandwidth allocations to deliver the desired application performance
Control delay, jitter, and packet loss
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Step 2: Define Traffic Classes
Scavenger Class
Less than Best Effort
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Quality of Service OperationsHow Do QoS Tools Work?
Classification and Marking
Queuing and (Selective) Dropping
Post-Queuing Operations
© 2006 Cisco Systems, Inc. All rights reserved.
Selecting an Appropriate QoS Policy Model
© 2006 Cisco Systems, Inc. All rights reserved.
Three QoS Models
Model Characteristics
Best 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.
© 2006 Cisco Systems, Inc. All rights reserved.
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 scalable
No special mechanisms required
Drawbacks:No service guarantees
No service differentiation
© 2006 Cisco Systems, Inc. All rights reserved.
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 rate
Controlled load (low delay, high throughput)
© 2006 Cisco Systems, Inc. All rights reserved.
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 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
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
© 2006 Cisco Systems, Inc. All rights reserved.
Implement the DiffServ QoS Model
Lesson 4.1: Introducing Classification and Marking
© 2006 Cisco Systems, Inc. All rights reserved.
Classification Classification is the process of identifying and
categorizing traffic into classes, typically based upon:Incoming interface
IP precedence
DSCP
Source or destination address
Application
Without classification, all packets are treated the same.
Classification should take place as close to the source as possible.
© 2006 Cisco Systems, Inc. All rights reserved.
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 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
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Classification and Marking in the Enterprise
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 2006 Cisco Systems, Inc. All rights reserved.
IP ToS Byte and DS Field Inside the IP Header
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Per-Hop Behaviors
DSCP selects PHB throughout the network:Default PHB (FIFO, tail drop)
Class-selector PHB (IP precedence)
EF PHB
AF PHB
© 2006 Cisco Systems, Inc. All rights reserved.
Standard PHB Groups
© 2006 Cisco Systems, Inc. All rights reserved.
Expedited Forwarding (EF) PHB
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)
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
Bit 2: Just 0
© 2006 Cisco Systems, Inc. All rights reserved.
Assured Forwarding (AF) PHB
AF PHB:Guarantees bandwidth
Allows 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 class
dd is drop probability
© 2006 Cisco Systems, Inc. All rights reserved.
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.
© 2006 Cisco Systems, Inc. All rights reserved.
Mapping CoS to Network Layer QoS
© 2006 Cisco Systems, Inc. All rights reserved.
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)
© 2006 Cisco Systems, Inc. All rights reserved.
Implementing QoS Policy Using a QoS Service Class
© 2006 Cisco Systems, Inc. All rights reserved.
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: 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)
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.
© 2006 Cisco Systems, Inc. All rights reserved.
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
© 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
© 2006 Cisco Systems, Inc. All rights reserved.
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 system
Access layer
Distribution layer
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Trust Boundaries: Mark Where?
For scalability, marking should be done as close to the source as possible.