• Chapter 1: Cisco VOIP Implementations
• Chapter 2: IP Quality of Service
• Chapter 3: Classification, Marking, and NBAR
• Chapter 4: Congestion Management and Queuing
• Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficieny Mechanisms
• Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS
• Chapter 7: Implementing AutoQoS
• Chapter 8: Wireless LAN QoS Implementation
• Chapter 9: 802.1x and Configuring Encryption and Authentication
• Chapter 10: WLAN Management

Chapter 1: Cisco VOIP Implementations

Benefits of packet telephony:

• More efficient use of bandwidth
• Consolidated network expenses (converged infrastructure)
• Improved employee productivity
• Access to a variety of communication devices (soft phones, PDAs, etc.)

Packet telephony components:

• Phones
• Gateways - Interconnect packet- and circuit-switched voice networks
• Multipoint Control Units (MCU) - Conference hardware; comprised of a multipoint controller and optional multipoint processor
• Application/database servers - TFTP, XML services, etc.
• Gatekeepers - Provide call routing (name-address resolution) and Call Admission Control (CAC, permission granting for call setup)
• Call agents - Responsible for call routing, address translation, call setup, etc. in a centralized call control model
• Video end points
• Digital Signal Processor (DSP) - Implementation of voice and/or video codec(s)

Analog interfaces:

• Foreign Exchange Office (FXO) - Faces upstream PSTN; acts like an analog phone
• Foreign Exchange Station - Faces analog phones; acts like a CO switch
• E&M - Used to connect gateways, PBX switches, or CO switches

Phone call stages:

1. Call setup - Call routing, CAC, parameter negotiation (IP addresses, UDP ports, codec)
2. Call maintenance - Statistics and error collection
3. Call tear-down - Notification of call end, frees resources on control devices

Call control:

• Distributed - H.323 and Session Initiation Protocol (SIP); all functionality is performed by the end nodes
• Centralized - Media Gateway Control Protocol (MGCP); end points rely on centralized call agent(s) for call routing, CAC, etc.

Analog to Digital Conversion

1. Sampling - Capturing voice as a Pulse Amplitude Modulation (PAM) stream
2. Quantization - Assigning numeric value to each sample in a PAM stream
3. Encoding - Representation of the quantized values in binary format
4. Compression (optional)

The Nyquist theorem states that an analog signal must be sampled at at least twice its highest frequency to be accurately reconstructed by the receiving end; a 4KHz voice signal is sampled at 8KHz.

Comparing codec quality:

• Mean Opinion Score (MOS) - humans judge quality relative to an in-person conversation on a scale of 1 to 5.
• Perceptual Speech Quality Measurement (PSQM) - Automated; 0 = best, 6.5 = worst
• Perceptual Analysis Measurement System (PAMS) - Predictive
• Perceptual Evaluation of Speech Quality (PESQ) - Predictive

Codecs:

• G.711 - Normal PCM; 64Kbps
• G.726 - Adaptive Differential PCM (ADPCM); three possible implementations (r32, r24, r16) use 32Kbps, 24Kbps, and 16Kbps respectively by sending only 4, 3, or 2 bits per sample
• G.722 - Wideband speech encoding; input signal is split into two sub-bands, each encoded with a modified version of ADPCM; 64Kbps, 56Kbps, or 48Kbps
• G.728 - Low Delay Code Exited Linear Prediction (LDCELP); expresses wave shapes of five samples with 10-bit values; 16Kbps
• G.729 - Conjugative Structure Algebraic Code Exited Linear Prediction (CS-ACELP); like G.728 but with ten samples; 8Kbps

Digital Signal Processors (DSPs) are processors dedicated to processing voice, and are found in pluggable Packet Voice DSP Modules (PVDMs).

DSP services:

• Voice termination
• Transcoding (between two different codecs)
• Conferencing

Bandwidth Utilization

Overhead: IP (20 bytes) + UDP (8 bytes) + RTP (12 bytes) = 40 bytes

Overhead can be greatly reduced by using Compressed RTP (cRTP), which requires only 2 bytes (4 bytes with checksum).

Because of the processor overhead involved, cRTP should only be used on slow links.

VOIP bandwidth calculation:

1. Determine the codec and packetization period (samples per packet)
2. Determine protocol overhead (cRTP, tunneling, etc)
3. Calculate the packetization size (amount of voice data per packet)
4. Add the lower layer protocol headers to calculate the total frame size (RTP/UDP/IP or cRTP + IPsec, etc)
5. Calculate the packet rate (inverse of packetization period) in packets per second
6. Calculate total bandwidth (#4 multiplied by #5)

Voice Activity Detection (VAD) detects silence on the line and momentarily stops generating data to conserve bandwidth.

Cisco Unified CallManager Functions

• Call processing
• Dial plan administration
• Signaling and device control
• Phone feature administration
• Directory and XML services
• Provides a programming interface to external applications

Survivable Remote Site Telephony (SRST) provides bare VOIP services to branch phones should the connection to a central CallManager be lost