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Benefits of a VoIP Network

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    Benefits of a VoIP Network Benefits of a VoIP Network Document Transcript

    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 1 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 2 Presentation_ID.scr
    • 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 5 3 2 1 4 © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 3 Presentation_ID.scr
    • Digital Interfaces Framing Total Interface Voice Channels (64 kbps Each) Signaling Overhead Bandwidth BRI 2 1 channel (16 kbps) 48 kbps 192 kbps T1 CAS 24 (no clean 64 kbps because of in-band (robbed-bits 8 kbps 1544 kbps robbed-bit signaling) in voice channels) 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. © 2006, Cisco Systems, Inc. All rights reserved. 4 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 5 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 6 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 7 Presentation_ID.scr
    • 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 Modulation Replaces PCM Transmits only the difference between one sample and the next © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 8 Presentation_ID.scr
    • Common Voice Codec Characteristics ITU-T Codec Bit Rate (kbps) Standard G.711 PCM 64 G.726 ADPCM 16, 24, 32 G.728 LDCELP (Low Delay CELP) 16 G.729 CS-ACELP 8 CS-ACELP, but with less G.729A 8 computation © 2006 Cisco Systems, Inc. All rights reserved. Mean Opinion Score © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 9 Presentation_ID.scr
    • A Closer Look at a DSP A DSP is a specialized processor used for telephony applications: DSP Module Voice termination: Works as a compander converting analog voice to digital format and back again Voice Network Module 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 © 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. © 2006, Cisco Systems, Inc. All rights reserved. 10 Presentation_ID.scr
    • Example: DSP Used for Transcoding © 2006 Cisco Systems, Inc. All rights reserved. Encapsulating Voice Packets for Transport © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 11 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 12 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 13 Presentation_ID.scr
    • Protocols Used for VoIP Voice Feature TCP UDP RTP Needs Reliability No Yes No No Reordering Yes Yes No Yes Time- Yes No No Yes stamping Contains As little as Overhead possible unnecessary Low Low information 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. © 2006, Cisco Systems, Inc. All rights reserved. 14 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 15 Presentation_ID.scr
    • cRTP Operation Condition Action The change is The sending side tracks the predicted predictable. change. The predicted change The sending side sends a hash of the is tracked. header. The receiving side The receiving side substitutes the original predicts what the stored header and calculates the constant change is. changed fields. There is an The sending side sends the entire header unexpected change. 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. © 2006, Cisco Systems, Inc. All rights reserved. 16 Presentation_ID.scr
    • 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 – Depends on packetization period (payload size) – Depends on codec bandwidth (bits per sample) IP overhead – Depends on the use of cRTP (including UDP and RTP) Data-link overhead – Depends on protocol (different per link) Tunneling overhead (if – Depends on protocol (IPsec, GRE, or used) MPLS) © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 17 Presentation_ID.scr
    • Bandwidth Implications of Codecs Codec bandwidth is for voice information only. Codec Bandwidth No packetization overhead is included. 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. © 2006, Cisco Systems, Inc. All rights reserved. 18 Presentation_ID.scr
    • VoIP Packet Size and Packet Rate Examples Codec and G.711 G.711 G.729 G.729 Packetization Period 20 ms 30 ms 20 ms 40 ms Codec bandwidth 64 64 8 8 (kbps) Packetization size 160 240 20 40 (bytes) IP overhead 40 40 40 40 (bytes) VoIP packet size 200 280 60 80 (bytes) Packet rate 50 33.33 50 25 (pps) © 2006 Cisco Systems, Inc. All rights reserved. Data-Link Overhead Is Different per Link Data-Link Frame Ethernet Trunk Ethernet MLP Protocol Relay (802.1Q) Overhead 18 6 6 22 [bytes] © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 19 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 20 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 21 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 22 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 23 Presentation_ID.scr
    • VAD Bandwidth-Reduction Examples Data-Link Ethernet Frame Relay Frame Relay MLPP Overhead 18 bytes 6 bytes 6 bytes 6 bytes IP overhead no cRTP cRTP no cRTP cRTP 40 bytes 4 bytes 40 bytes 2 bytes Codec G.711 G.711 G.729 G.729 64 kbps 64 kbps 8 kbps 8 kbps Packetization 20 ms 30 ms 20 ms 40 ms 160 bytes 240 bytes 20 bytes 40 bytes Bandwidth 87.2 kbps 66.67 kbps 26.4 kbps 9.6 kbps without VAD Bandwidth with 56.68 kbps 43.33 kbps 17.16 kbps 6.24 kbps VAD (35% reduction) © 2006 Cisco Systems, Inc. All rights reserved. Introducing QoS © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 24 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 25 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 26 Presentation_ID.scr
    • 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 Voice 1 1 Voice (Highest) • LLQ • RTP header Data compression 2 2 (High) 4 3 2 1 1 Data 3 3 3 Data (Medium) • CBWFQ • TCP header compression Data 4 4 4 4 (Low) 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 © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 27 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 28 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 29 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 30 Presentation_ID.scr
    • 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 Traffic Traffic Traffic Rate Traffic Rate Policing Time Time Traffic Traffic Traffic Rate Traffic Rate Shaping Time Time © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 31 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 32 Presentation_ID.scr
    • What Is Quality of Service? Two Perspectives The user perspective Users perceive that their applications are performing properly Voice, video, and data The network manager perspective Need to manage bandwidth allocations to deliver the desired application performance Control delay, jitter, and packet loss © 2006 Cisco Systems, Inc. All rights reserved. Different Types of Traffic Have Different Needs Sensitivity to Real-time applications especially QoS Metrics Application sensitive to QoS Examples Packet Interactive voice Delay Jitter Loss Videoconferencing Interactive Voice and Video Y Y Y Causes of degraded performance Congestion losses Streaming Video N Y Y Variable queuing delays Transactional/ The QoS challenge Interactive Y N N Manage bandwidth allocations to Bulk Data deliver the desired application Email N N N performance File Transfer Control delay, jitter, and packet loss Need to manage bandwidth allocations © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 33 Presentation_ID.scr
    • 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 Less than Best Effort Class © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 34 Presentation_ID.scr
    • 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 Operations How Do QoS Tools Work? Classification Queuing and Post-Queuing and Marking (Selective) Dropping Operations © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 35 Presentation_ID.scr
    • 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 Applications signal to the network that the Services applications require certain QoS parameters. (IntServ) Differentiated The network recognizes classes that require Services QoS. (DiffServ) © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 36 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 37 Presentation_ID.scr
    • 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 End Station Edge Interior Edge DiffServ Domain End Station © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 38 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 39 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 40 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 41 Presentation_ID.scr
    • Classification Tools IP Precedence and DiffServ Code Points Version ToS Len ID Offset TTL Proto FCS IP SA IP DA Data Length Byte IPv4 Packet 7 6 5 4 3 2 1 0 Standard IPv4 IP Precedence Unused DiffServ Code Point (DSCP) IP ECN DiffServ Extensions 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 © 2006 Cisco Systems, Inc. All rights reserved. IP ToS Byte and DS Field Inside the IP Header © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 42 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 43 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 44 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 45 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 46 Presentation_ID.scr
    • 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. © 2006, Cisco Systems, Inc. All rights reserved. 47 Presentation_ID.scr
    • Classification and Marking Design QoS Baseline Marking Recommendations L3 Classification L2 Application IPP PHB DSCP CoS Routing 6 CS6 48 6 Voice 5 EF 46 5 Video Conferencing 4 AF41 34 4 Streaming Video 4 CS4 32 4 Mission-Critical Data 3 AF31* 26 3 Call Signaling 3 CS3* 24 3 Transactional Data 2 AF21 18 2 Network Management 2 CS2 16 2 Bulk Data 1 AF11 10 1 Best Effort 0 0 0 0 Scavenger 1 CS1 8 1 © 2006 Cisco Systems, Inc. All rights reserved. How Many Classes of Service Do I Need? 4/5 Class Model 8 Class Model 11 Class Model Voice Voice Realtime Interactive-Video Video Streaming Video Call Signaling Call Signaling Call Signaling IP Routing Network Control Network Management Critical Data Mission-Critical Data Critical Data Transactional Data Bulk Data Bulk Data Best Effort Best Effort Best Effort Scavenger Scavenger Scavenger Time © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 48 Presentation_ID.scr
    • 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 © 2006 Cisco Systems, Inc. All rights reserved. Trust Boundaries: Mark Where? For scalability, marking should be done as close to the source as possible. © 2006 Cisco Systems, Inc. All rights reserved. © 2006, Cisco Systems, Inc. All rights reserved. 49 Presentation_ID.scr