Voice Packetization, Codecs and Voice Quality

Voice Packetization, Codecs and Voice Quality
The "nuts and bolts" of Voice over IP: 
 • How voice is digitized and coded, 
 • Time stamps applied with the RTP protocol, and
 • How the result is carried in UDP, IP packets and MAC frames.

You'll learn about codecs and compression, and understand factors like delay, jitter and packet loss, what causes them, and how they affect sound quality. 
 
VoIP Module 4
Detailed Outline

4. Voice Packetization, Codecs and Voice Quality
4.1 Voice Packetization
...... 4.1.1 Microphone Produces a Voltage Analog
...... 4.1.2 Digitization
...... 4.1.3 Coding
...... 4.1.4 Segmentation and Packetization
4.2 Measuring Voice Quality
...... 4.2.1 Mean Opinion Score
...... 4.2.2 Toll-Quality Speech
4.3 Factors Affecting Voice Quality
...... 4.3.1 Codec, Delay, Jitter and Packet Loss
...... 4.3.2 Network Load
4.4 Codecs and Compression
...... 4.4.1 Coder / Decoder
...... 4.4.2 Tradeoffs
...... 4.4.3 CBR vs. VBR
...... 4.4.4 Standards
4.5 Delay
...... 4.5.1 Coding Delay
...... 4.5.2 Network Delay
...... 4.5.3 Design Goal
4.6 Jitter
...... 4.6.1 IP: No Guarantees
...... 4.6.2 Variable Delivery Time
...... 4.6.3 Variable Network Loading
...... 4.6.4 Jitter Buffer
4.7 RTP
...... 4.7.1 Real-Time Transport Protocol
...... 4.7.2 Segmentation
...... 4.7.3 Time Stamp
...... 4.7.4 Error Recovery
4.8 VoIP Protocol Stack: RTP, UDP, IP, MAC
...... 4.8.1 Moving the RTP PDU
...... 4.8.2 UDP Error Detection and Recovery
...... 4.8.3 Port Number
...... 4.8.4 RTP/UDP/IP/Ethernet Protocol Stack
...... 4.8.5 Header Compression
4.9 Packet Loss
...... 4.9.1 Packet Loss Mechanism
...... 4.9.2 Recovery
4.10 Tips for Maximizing Voice Quality
...... 4.10.1 Network Level Testing
...... 4.10.2 VoIP Application Level Testing
...... 4.10.3 Strategies