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 application Software-based and wireless phones offer mobility. Access to new communications devices (such as PDAs and cable set-top boxes) 1
Components of a VoIP Network Legacy Analog and VoIP Applications Can Coexist 2
Legacy Analog Interfaces in VoIP Networks Analog Interface Type Foreign Exchange Station Foreign Exchange Office Earth and Magneto Label FXS FXO E&M Description Used by the PSTN or PBX side of an FXS FXO connection Used by the end device side of an FXS FXO connection Trunk, used between switches Legacy Analog Interfaces in VoIP Networks 1 3 1 4 5 2 3
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 4
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. 5
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. 6
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 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. 7
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. 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 8
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 9
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 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. 10
Example: DSP Used for Transcoding Encapsulating Voice Packets for Transport 11
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 network Shared bandwidth, higher and variable delays 12
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. 13
Protocols Used for VoIP Feature Voice Needs TCP UDP RTP Reliability No Yes No No Reordering Yes Yes No Yes Timestamping 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. 14
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 preserved 2 bytes if the UDP checksum is not sent Saves a considerable amount of bandwidth 15
crtp Operation Condition Action The change is predictable. The predicted change is tracked. The receiving side predicts what the constant change is. There is an unexpected change. The sending side tracks the predicted change. The sending side sends a hash of the header. The receiving side substitutes the original stored header and calculates the changed fields. 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 overhead Introduces additional delays Tune crtp set the number of sessions to be compressed (default is 16). 16
Calculating Bandwidth Requirements for VoIP Factors Influencing Encapsulation Overhead and Bandwidth Factor Packet rate Packetization size (payload size) IP overhead (including UDP and RTP) Data-link overhead Tunneling overhead (if used) Description Derived from packetization period (the period over which encoded voice bits are collected for encapsulation) Depends on packetization period Depends on codec bandwidth (bits per sample) Depends on the use of crtp Depends on protocol (different per link) Depends on protocol (IPsec, GRE, or MPLS) 17
Bandwidth Implications of Codecs Codec bandwidth is for voice information only. No packetization overhead is included. Codec G.711 Bandwidth 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) 18
VoIP Packet Size and Packet Rate Examples Codec and Packetization 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 19
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 IPsec transport mode IPsec tunnel mode L2TP/GRE MPLS PPPoE Header Size (bytes) 30 53 50 73 24 4 8 20
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 link Deploying Call Admission Control (CAC) Deploying QoS 21
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. Bandwidth Calculation Example 22
Quick Bandwidth Calculation Total packet size Total bandwidth requirement = Payload size Nominal bandwidth requirement Total packet size = All headers + payload Parameter Layer 2 header IP + UDP + RTP headers Payload size (20-ms sample interval) Nominal bandwidth Value 6 to 18 bytes 40 bytes 20 bytes for G.729, 160 bytes for G.711 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 noise Other factors (for example, language, character of speaker, or type of call) Can save up to 35 percent of bandwidth 23
VAD Bandwidth-Reduction Examples Data-Link Overhead Ethernet 18 bytes Frame Relay 6 bytes Frame Relay 6 bytes MLPP 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 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 24
Traditional Nonconverged Network Traditional data traffic characteristics: Bursty data flow FIFO access Not overly time-sensitive; delays OK Brief outages are survivable 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. 25
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. 26
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 Voice (Highest) Data (High) 1 1 2 2 Voice LLQ RTP header compression 4 3 2 1 1 Data (Medium) Data (Low) 3 3 3 4 4 4 4 Data CBWFQ TCP header compression 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 27
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. 28
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 compression LLQ Prioritization ISP routers perform: Reprioritization according to the QoS policy 29
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. 30
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 Traffic Traffic Rate Policing Traffic Traffic Rate Time Time Traffic Traffic Rate Shaping Traffic Traffic Rate Time Time 31
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 32
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 Different Types of Traffic Have Different Needs 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 Application Examples Interactive Voice and Video Streaming Video Transactional/ Interactive Bulk Data Email File Transfer Delay Y N Y N Sensitivity to QoS Metrics Jitter Y Y N N Packet Loss Need to manage bandwidth allocations Y Y N N 33
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 34
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 Operations How Do QoS Tools Work? Classification and Marking Queuing and (Selective) Dropping Post-Queuing Operations 35
Selecting an Appropriate QoS Policy Model Three QoS Models Model Best effort Integrated Services (IntServ) Differentiated Services (DiffServ) Characteristics No QoS is applied to packets. If it is not important when or how packets arrive, the besteffort model is appropriate. Applications signal to the network that the applications require certain QoS parameters. The network recognizes classes that require QoS. 36
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 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) 37
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 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 Edge Interior DiffServ Domain End Station 38
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 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. 39
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 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. 40
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. 41
Classification Tools IP Precedence and DiffServ Code Points Version Length ToS Byte Len ID Offset TTL Proto FCS IP SA IP DA Data IPv4 Packet 7 6 5 4 3 2 1 0 IP Precedence Unused DiffServ Code Point (DSCP) IP ECN Standard IPv4 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 IP ToS Byte and DS Field Inside the IP Header 42
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 PHB AF PHB 43
Standard PHB Groups 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-diffservcompliant 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 44
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 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. 45
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) 46
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: 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. 47
Classification and Marking Design QoS Baseline Marking Recommendations Application IPP L3 Classification PHB DSCP L2 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 How Many Classes of Service Do I Need? 4/5 Class Model Realtime Call Signaling Critical Data Best Effort Scavenger Time 8 Class Model Voice Video Call Signaling Network Control Critical Data Bulk Data Best Effort Scavenger 11 Class Model Voice Interactive-Video Streaming Video Call Signaling IP Routing Network Management Mission-Critical Data Transactional Data Bulk Data Best Effort Scavenger 48
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 Trust Boundaries: Mark Where? For scalability, marking should be done as close to the source as possible. 49