Home | Networking Fundamentals
Google
 
 
  

 

Lesson 1: Networking Basics

Lesson 2: OSI Reference Model

Lesson 3: Introduction to TCP/IP

Lesson 4: LAN Basics

Lesson 5: Understanding Switching

Lesson 6: WAN Basics

Lesson 7: Understanding Routing

Lesson 8: What Is Layer 3 Switching?

Lesson 9: Understanding Virtual LANs

Lesson 10: Understanding Quality of Service

Lesson 11: Security Basics

Lesson 12: Understanding Virtual Private Networks

Lesson 13: Voice Technology Basics

Lesson 14: Network Management Basics

Lesson 15: The Internet

 

 

 

Lesson 13: Voice Technology Basics

Convergence of Voice and Data | Voice Technology Basics | Voice over Data Transports | Applications |

Sample Migration

Voice over Data Networks

Now that you understand how today’s voice networks work, let’s take a look at how real-time voice over a data network works.

Voice over Packet Networks Allow Real-Time Voice on Data Networks

Voice over packet networks provide techniques for sending real-time voice over data networks, including IP, Frame Relay, and Asynchronous Transfer Mode (ATM) networks.


Analog voice is converted into digital voice packets, sent over the data network as data packets, and converted to analog voice on the other end.

Converting from Voice to Data

Analog voice packets are converted to digital data packets with the following steps:

   1. A person speaking into the telephone is an analog voice signal.
   2. Coder-decoder (CODEC) software converts the signal from analog to digital data packets suitable         for transmission over a TCP/IP network.
   3. A digital signal processor (DSP) chip compresses the packets for transmission over the data         network.

The data network can be an IP LAN, or a leased-line, ATM, or Frame Relay network.

Converting from Data Back to Voice

Digital data packets are converted to Analog voice packets with the following steps:

   4. DSP chip uncompresses the packets
   5. CODEC software converts the signal from digital data packets back to analog voice
   6. Recipient listens to the voice on their telephone

The “Enabling” Technologies

What’s made this all possible is that in the last ten years, a lot of things have happened in voice technology:

Access price/performance: Access products and services have increased in price performance.

Processing: Digital signal processors (DSPs) specialize in processing analog wave forms, which voice or video inherently are. Today, DSPs are cheaper and higher powered, enabling more advanced algorithms to compress, synthesize, and process voice and video signals. CPUs within the devices have increased in power as well.

Voice compression: Voice compression is used to save bandwidth. A variety of voice compression schemes provide a variety of levels of bandwidth usage and voice quality. These compression methods often do not interoperate. Modem, fax, and dual tone multifrequency (DTMF) functionality are all impacted by voice-compression methods.

Standards: Advances have been made over the past few years that enable the transmission of voice traffic over traditional public networks, such as Frame Relay (Voice over Frame Relay).
Standards, such as G.729 for voice compression, FRF.11 and FRF.12 for voice over Frame Relay, and the long list of ATM standards enable different types of traffic to come together in a nonproprietary network.
Additionally, the support of Asynchronous Transfer Mode (ATM) for different traffic types, and the ATM Forum’s recent completion of the Voice and Telephony over ATM specification, will speed up the availability of industry-standard solutions for voice over ATM.

Higher-speed infrastructure: In general, the infrastructures to support voice in corporate environments and in the public network environments are much higher-speed now, so they can carry more voice traffic and effectively take on the voice tasks for the corporation.

<<Back [1] [2] [3] [4] [5] [6] [7] [8] [9] [10] [11] [12] [13] [14] [15] [16] [17] Next>>