A systematic approach to understanding the basics of voice over IP
# Understand the basics of enterprise and public telephony networking, IP networking, and how voice is transported over IP networks
# Learn the various caveats of converging voice and data networks
# Examine the basic VoIP signaling protocols (H.323, MGCP/H.248, SIP) and primary legacy voice signaling protocols (ISDN, C7/SS7)
# Explore how VoIP can run the same applications as the existing telephony system but in a more cost-efficient and scalable manner
# Delve into such VoIP topics as jitter, latency, packet loss, codecs, QoS tools, and security
Voice over IP (VoIP) has become an important factor in network communications, promising lower operational costs, greater flexibility, and a variety of enhanced applications. To help you understand VoIP networks, Voice over IP Fundamentals provides a thorough introduction to the basics of VoIP.
Voice over IP Fundamentals explains how a basic IP telephony infrastructure is built and works today, major concepts concerning voice and data networking, and transmission of voice over data networks. You’ll learn how voice is signaled through legacy telephone networks, how IP signaling protocols are used to interoperate with current telephony systems, and how to ensure good voice quality using quality of service (QoS).
Even though Voice over IP Fundamentals is written for anyone seeking to understand how to use IP to transport voice, its target audience comprises both voice and data networking professionals. In the past, professionals working in voice and data networking did not have to understand each other’s roles. However, in this world of time-division multiplexing (TDM) and IP convergence, it is important to understand how these technologies work together. Voice over IP Fundamentals explains all the details so that voice experts can understand data networking and data experts can understand voice networking.
The second edition of this best-selling book includes new chapters on the importance of billing and mediation in a VoIP network, security, and the common types of threats inherent when packet voice environments, public switched telephone networks (PSTN), and VoIP interoperate. It also explains enterprise and service-provider applications and services.
Article Information
Contents
1. Delay/Latency
2. Jitter
3. Pulse Code Modulation
4. Voice Compression
5. Echo
6. Packet Loss
7. Voice Activity Detection
8. Digital-to-Analog Conversion
9. Tandem Encoding
10. Transport Protocols
11. Dial-Plan Design
12. End Office Switch Call-Flow Versus IP Phone Call
13. Summary
14. References
Article Description
This chapter explains many of the issues facing Voice over IP (VoIP) and ways in which Cisco addresses these issues.
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Voice over IP Fundamentals, 2nd Edition
Voice over IP Fundamentals, 2nd Edition
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To create a proper network design, it is important to know all the caveats and inner workings of networking technology. This chapter explains many of the issues facing Voice over IP (VoIP) and ways in which Cisco addresses these issues.
Communications via the Public Switched Telephone Network (PSTN) has its own set of problems, which are covered in Chapter 1, “Overview of the PSTN and Comparisons to Voice over IP,” and Chapter 2, “Enterprise Telephony Today.” VoIP technology has many similar issues and a whole batch of additional ones. This chapter details these various issues and explains how they can affect packet networks.
The following issues are covered in this chapter:
* Delay/latency
* Jitter
* Pulse Code Modulation (PCM)
* Voice compression
* Echo
* Packet loss
* Voice activity detection
* Digital-to-analog conversion
* Tandem encoding
* Transport protocols
* Dial-plan design
Delay/Latency
VoIP delay or latency is characterized as the amount of time it takes for speech to exit the speaker’s mouth and reach the listener’s ear.
Three types of delay are inherent in today’s telephony networks: propagation delay, serialization delay, and handling delay. Propagation delay is caused by the length a signal must travel via light in fiber or electrical impulse in copper-based networks. Handling delay—also called processing delay—defines many different causes of delay (actual packetization, compression, and packet switching) and is caused by devices that forward the frame through the network.
Serialization delay is the amount of time it takes to actually place a bit or byte onto an interface. Serialization delay is not covered in depth in this book because its influence on delay is relatively minimal.
Propagation Delay
Light travels through a vacuum at a speed of 186, 000 miles per second, and electrons travel through copper or fiber at approximately 125, 000 miles per second. A fiber network stretching halfway around the world (13, 000 miles) induces a one-way delay of about 70 milliseconds (70 ms). Although this delay is almost imperceptible to the human ear, propagation delays in conjunction with handling delays can cause noticeable speech degradation.
Handling Delay
As mentioned previously, devices that forward the frame through the network cause handling delay. Handling delays can impact traditional phone networks, but these delays are a larger issue in packetized environments. The following paragraphs discuss the different handling delays and how they affect voice quality.
In the Cisco IOS VoIP product, the Digital Signal Processor (DSP) generates a speech sample every 10 ms when using G.729. Two of these speech samples (both with 10 ms of delay) are then placed within one packet. The packet delay is, therefore, 20 ms. An initial look-ahead of 5 ms occurs when using G.729, giving an initial delay of 25 ms for the first speech frame.
Vendors can decide how many speech samples they want to send in one packet. Because G.729 uses 10 ms speech samples, each increase in samples per frame raises the delay by 10 ms. In fact, Cisco IOS enables users to choose how many samples to put into each frame.
Cisco gave DSP much of the responsibility for framing and forming packets to keep router/ gateway overhead low. The Real-Time Transport Protocol (RTP) header, for example, is placed on the frame in the DSP instead of giving the router that task.
Queuing Delay
A packet-based network experiences delay for other reasons. Two of these are the time necessary to move the actual packet to the output queue (packet switching) and queuing delay.
When packets are held in a queue because of congestion on an outbound interface, the result is queuing delay. Queuing delay occurs when more packets are sent out than the interface can handle at a given interval.
The actual queuing delay of the output queue is another cause of delay. You should keep this factor to less than 10 ms whenever you can by using whatever queuing methods are optimal for your network. This subject is covered in greater detail in Chapter 8, “Quality of Service.”
The International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.114 recommendation specifies that for good voice quality, no more than 150 ms of one-way, end-to-end delay should occur, as shown in Figure 7-1. With the Cisco VoIP implementation, two routers with minimal network delay (back to back) use only about 60 ms of end-to-end delay. This leaves up to 90 ms of network delay to move the IP packet from source to destination.
vf330701.gif
Figure 7-1 End-to-End Delay
As shown in Figure 7-1, some forms of delay are longer, although accepted, because no other alternatives exist. In satellite transmission, for example, it takes approximately 250 ms for a transmission to reach the satellite, and another 250 ms for it to come back down to Earth. This results in a total delay of 500 ms. Although the ITU-T recommendation notes that this is outside the acceptable range of voice quality, many conversations occur every day over satellite links. As such, voice quality is often defined as what users will accept and use.
In an unmanaged, congested network, queuing delay can add up to two seconds of delay (or result in the packet being dropped). This lengthy period of delay is unacceptable in almost any voice network. Queuing delay is only one component of end-to-end delay. Another way end-to-end delay is affected is through jitter.
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