Chapter 1 – CCNA VOICE

The Cisco Certified Network Associate (CCNA) Voice certification validates associate-level knowledge and skills required to administer a voice network. The CCNA Voice certification confirms that an individual has the required skill set for specialized job roles in voice technologies including Voice Administrator, Voice Engineer, and Voice Manager. It validates skills in VoIP technologies such as IP PBX, IP telephony, handset, call control, and voicemail solutions.

Prerequisite: CCENT, CCNA Routing and Switching, or CCIE/CCDE Certification. Effective October 1, 2013, you need only one more exam to achieve CCNA concentration if you meet these prerequisites: 640-554 IINS for Security or 640-722 IUWNE for Wireless or 640-461 ICOMM for Voice.

Required Exam: ICOMM 640-461.

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This chapter walks you through the foundations of the public switched telephone network (PSTN), private branch exchange (PBX) systems, and analog and digital circuitry.
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Each analog circuit is composed of a pair of wires. One wire is the ground, or positive side of the connection (often called the tip). The other wire is the battery, or negative side of the connection (often called the ring). You’ll commonly hear phone technicians talk about these wires as the “tip and ring.” These two wires are what power the analog phone and allow it to function, just like the wires that connect your car battery to the car
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The jagged line over the wires in the analog phone in Figure 1-3 represents a broken circuit. Anytime the phone is on hook, the phone separates the two wires, preventing electric signal from flowing through the phone. When the phone is lifted off hook, the phone connects the two wires, causing an electrical signal (48V DC voltage) to flow from the phone company central office (CO) into the phone. This is known as (((loop start signaling))).
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Loop start signaling is the typical signaling type used in home environments. Loop start signaling is susceptible to a problem known as glare
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Glare occurs when you pick up the phone to make an outgoing call at the same time as a call comes in on the phone line before the phone has a chance to ring.
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If a call comes in for x5002 at the same time as x5000 picks up the phone, the key system will connect the two signals, causing x5000 to receive the call for x5002. This happens because the loop start signal from x5000 seizes the outgoing PSTN line at the same time as the key system receives the incoming call on the same PSTN line. This is an instance of glare.
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Because of glare, most modern PBX systems designed for larger, corporate environments use ground start signaling. Ground start signaling originated from its implementation in pay phone systems. Many years ago, when a person lifted the handset of a pay phone, he did not receive a dial tone until he dropped in a coin.
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Many other types of signaling exist in the analog world. These include supervisory signaling (on hook, off hook, ringing), informational signaling (dial tone, busy, ringback, and so on), and address signaling (dual-tone multifrequency (DTMF) and Pulse).
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Analog signaling was a massive improvement over tin cans and string, but still posed plenty of problems of their own. First, an analog electrical signal experiences degradation (signal fading) over long distances. To increase the distance the analog signal could travel, the phone company had to install repeaters (shown in Figure 1-5) to regenerate the signal as it became weak
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Simply put, digital signals use numbers to represent levels of voice instead of a combination of electrical signals. When someone talks about “digitizing voice,” they are speaking of the process of changing analog voice signals into a series of numbers
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Digital voice uses a technology known as time-division multiplexing (TDM). TDM allows voice networks to carry multiple conversations at the same time over a single, four-wire path. Because the multiple conversations have been digitized, the numeric values are transmitted in specific time slots (thus, the “time division”) that differentiate the separate conversations.
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Corporations use digital voice connections to the PSTN as T1 circuits in the United States, Canada, and Japan. A T1 circuit is built from 24 separate 64-kbps channels known as a digital signal 0 (DS0). Each one of these channels is able to support a single voice call. Corporations in areas outside the United States, Canada, and Japan use E1 circuits, which allow you to use up to 30 DS0s for voice calls.
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With analog circuits, supervisory signals were passed by connecting the tip and ring wires together. The phone company generated informational and address signals through specific frequencies of electricity. By solving the problems associated with analog signaling, digital signaling also removed the typical signaling capabilities. To solve this, two primary styles of signaling were created for digital circuits:
• Channel associated signaling (CAS): Signaling information is transmitted using the same bandwidth as the voice.
• Common channel signaling (CCS): Signaling information is transmitted using a separate, dedicated signaling channel.
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Channel Associated Signaling
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T1 digital connections that use CAS actually “steal” binary bits that would typically have been used to communicate voice information and use them for signaling. Initially, this may seem crazy; if you take the binary bits that are used to resynthesize the voice, won’t the voice quality drop significantly? Although the voice quality does drop some, the number of binary bits stolen for signaling information is small enough that the change in voice quality is not noticeable. robbed bit signaling (RBS).
The voice device running the T1 line uses the eighth bit on every sixth sample in each T1 channel (DS0).
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As you can see from Figure 1-8, the 24 channels of the digital T1 circuit carry only voice data for the first five frames that they send. On the sixth frame (marked with an S in Figure 1-8), the eighth bit (also called the least significant bit) is stolen for the voice devices to transmit signaling information. This process occurs for every sixth frame after this (12th, 18th, 24th, and so on). This stolen bit relays the signaling information for each respective DS0 channel. For example, the bits stolen from the third DS0 channel relay the signaling information only for that channel.
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Common Channel Signaling
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CCS dedicates one of the DS0 channels from a T1 or E1 link for signaling information. This is often called out-of-band signaling because the signaling traffic is sent completely separate from the voice traffic. As a result, a T1 connection using CCS has only 23 usable DS0s for voice. Because CCS dedicates a full channel of the circuit for signaling, the “stolen bit” method of signaling using ABCD bits is no longer necessary. Rather, a full signaling protocol sends the necessary information for all voice channels. The most popular signaling protocol used is Q.931, which is the signaling protocol used for ISDN circuits.
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CCS is the most popular connection used between voice systems worldwide because it offers more flexibility with signaling messages, more bandwidth for the voice bearer channels, and higher security (because the signaling is not embedded in the voice channel). CCS also allows PBX vendors to communicate proprietary messages (and features) between their PBX systems using ISDN signaling, whereas CAS does not offer any of these capabilities.
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When using CCS configurations with T1 lines, the 24th time slot is always the signaling channel. When using CCS configurations with E1 lines, the 17th time slot is always the signaling channel.
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—–Pieces of PSTN
• Analog telephone: Able to connect directly to the PSTN and is the most common device on the PSTN. Converts audio into electrical signals.
• Local loop: The link between the customer premises (such as a home or business) and the telecommunications service provider.
• CO switch: Provides services to the devices on the local loop. These services include signaling, digit collection, call routing, setup, and teardown.
• Trunk: Provides a connection between switches. These switches could be CO or private.
• Private switch: Allows a business to operate a “miniature PSTN” inside its company. This provides efficiency and cost savings because each phone in the company does not require a direct connection to the CO switch.
• Digital telephone: Typically connects to a PBX system. Converts audio into binary 1s and 0s, which allows more efficient communication than analog.
—— PSTN Components.png
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Understanding PBX and Key Systems
Many businesses have hundreds or even thousands of phones they support in the organization. If the company purchases a direct PSTN connection for each one of these phones, the cost would be astronomical. Instead, most organizations choose to use a PBX or key system internally to manage in-house phones. These systems allow internal users to make phone calls inside the office without using any PSTN resources. Calls to the PSTN forward out the company’s PSTN trunk link.
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• Line cards: Provide the connection between telephone handsets and the PBX system.
• Trunk cards: Provide connections from the PBX system to the PSTN or other PBX systems.
• Control complex: Provides the intelligence behind the PBX system; all call setup, routing, and management functions are contained in the control complex.
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Although key systems often have a shared-line feature set, many key systems have numerous features that allow them to operate just like a PBX system, but with fewer ports
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Connections to and Between the PSTN
When you want to connect to the PSTN, you have a variety of options. Home users and small offices can connect using analog ports. Each two-wire analog connection has the capability to support a single call. For home users, a single, analog connection to the PSTN may be sufficient. For small offices, the number of incoming analog connections directly relates to the office size and average call volume. As businesses grow, you can consolidate the multiple analog connections into one or more digital T1 or E1 connections [conections to the PSTN.PNG]
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For all the telephony providers of the world to communicate together, a common signaling protocol must be used, similar to the way TCP/IP operates in the data realm. The voice signaling protocol used around the world is SS7.
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SS7 is an out-of-band (CCS-style) signaling method used to communicate call setup, routing, billing, and informational messages between telephone company COs around the world. When a user makes a call, the first CO to receive the call performs an SS7 lookup to locate the number. Once the destination is found, SS7 is responsible for routing the call through the voice network to the destination and providing all informational signaling (such as ring back) to the calling device.
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Organizations managing their own internal telephony systems can develop any internal number scheme that best fits the company needs (similar to private IP addressing). However, when connecting to the PSTN, you must use a valid, E.164 standard address for your telephone system. E.164 is an international numbering plan created by the International Telecommunication Union (ITU). Each number in the E.164 numbering plan contains the following components:
• Country code
• National destination code
• Subscriber number
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Note
E.164 numbers are limited to a maximum length of 15 digits.
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Audio frequencies vary based on the volume, pitch, and so on that comprise the sound. Here are a few key facts:
• The average human ear is able to hear frequencies from 20–20,000 Hz.
• Human speech uses frequencies from 200–9,000 Hz.
• Telephone channels typically transmit frequencies from 300–3,400 Hz.
• The Nyquist theorem is able to reproduce frequencies from 300–4,000 Hz.
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Because the first bit in Figure 1-13 is a 1, you read the number as positive. The remaining seven bits represent the number 52. This is the digital value used for one voice sample. Now, remember, the Nyquist theorem dictates that you need to take 8,000 of those samples every single second. Doing the math, figure 8,000 samples a second times the 8 bits in each sample, and you get 64,000 bits per second. It’s no coincidence that uncompressed audio (including the G.711 audio codec) consumes 64 kbps. Once the sampling device assigns numeric values to all these analog signals, a router can place them into a packet and send them across a network.
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The last and optional step in the digitization process is to apply compression measures. Advanced codecs, such as G.729, allow you to compress the number of samples sent and thus use less bandwidth. This is possible because sampling human voice 8,000 times a second produces many samples that are similar or identical. For example, say the word “cow” out loud to yourself (provided you are in a relatively private area). That takes about a second to say, right? If not, say it slower until it does. Now, listen to the sounds you are making. There’s the distinguished “k” sound that starts the word, then you have the “ahhhhhh” sound in the middle, followed by the “wa” sound at the end. If you were to break that into 8,000 individual samples, chances are most of them would sound the same.
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The process G.729 (and most other compressed codecs) uses to compress this audio is to send a sound sample once and simply tell the remote device to continue playing that sound for a certain time interval. This is often described as “building a codebook” of the human voice traveling between the two endpoints. Using this process, G.729 is able to reduce bandwidth down to 8 kbps for each call; a fairly massive reduction in bandwidth.
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If you stay in the Cisco realm for long, you will hear two codecs continually repeated: G.711 and G.729. This is because Cisco designed all its IP phones with the ability to code in either of these two formats. G.711 is the “common ground” between all VoIP devices. For example, if a Cisco IP phone is attempting to communicate with an Avaya IP phone, they may support different compressed codecs, but can at least agree on G.711 when communicating.
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G.729 comes in two different variants: G.729a (annex A) and G.729b (annex B). G.729a sacrifices some audio quality to achieve a much more processor-efficient coding process. G.729b introduces support for Voice Activity Detection (VAD), which makes voice transmissions more efficient.
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Moving into the realm of VoIP, the network now requires the router to convert loads of voice into digitized, packetized transmissions. This task would easily overwhelm the resources you have on the router. This is where DSPs come into play. DSPs offload the processing responsibility for voice-related tasks from the processor of the router. This is similar to the idea of purchasing an expensive video card for a PC to offload the video processing responsibility from the PC’s processor.
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DSPs typically come in chips to install in your Cisco router that look like old memory SIMMs,
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Cisco provides a DSP calculator that provides the number of DSP chips you need to purchase based on the voice network you are supporting. This tool can be found at http://www.cisco.com/web/applicat/dsprecal/index.html (Cisco.com login required). Keep in mind that a growing network will always require more DSP resources. It is usually best to pack the router full with as many DSP resources as you can fit in it; you’re going to need them!
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Understanding RTP and RTCP
When you walk into the VoIP world, you encounter a whole new host of protocol standards. Think of the Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP) as the protocols of voice. RTP operates at the transport layer of the OSI model on top of UDP. Having two transport layer protocols is odd, but that’s exactly what is happening here. UDP provides the services it always does: port numbers (that is, session multiplexing) and header checksums (which ensure that the header information does not become corrupted). RTP adds time stamps and sequence numbers to the header information. This allows the remote device to put the packets back in order when it receives them at the remote end (function of the sequence number) and use a buffer to remove jitter (slight delays) between the packets to give a smooth audio playout (function of the time stamp). Figure 1-15 represents the RTP header information contained in a packet.
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The Payload Type field in the RTP header is used to designate what type of RTP is in use. You can use RTP for audio or video purposes.
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Once two devices attempt to establish an audio session, RTP engages and chooses a random, even UDP port number from 16,384 to 32,767 for each RTP stream. Keep in mind that RTP streams are one way. If you are having a two-way conversation, the devices establish dual RTP streams, one in each direction. The audio stream stays on the initially chosen port for the duration of the audio session. (The devices do not dynamically change ports during a phone call.)
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At the time the devices establish the call, RTCP also engages. Although this protocol sounds important, its primary job is statistics reporting. It delivers statistics between the two devices participating in the call, which include:
Packet count
• Packet delay
• Packet loss
• Jitter (delay variations)
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Although this information is useful, it is not nearly as critical as the actual RTP audio streams. Keep this in mind when you configure QoS settings.
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RTCP creates a separate session over UDP between the two devices by using an odd-numbered port from the same range. Throughout the call duration, the devices send RTCP packets at least once every 5 seconds. The Cisco Unified Communication Manager Express (CME) router can log and report this information, which allows you to determine the issues that are causing call problems (such as poor audio, call disconnects, and so on) on the network.
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RTCP uses the odd-numbered port following the RTP port. For example, if the RTP audio uses port 17,654, the RTCP port for the session will be 17,655.
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