CH 1 Traditional vs Unified Voice REVIEW Flashcards

1
Q

Analog connections

A
  • Analog phone lines use the properties of electricity to convey changes in voice over ca-
    bling.
  • Signaling includes messages such as dial tone, dialed digits, busy signals.
  • 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).
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2
Q

Loop Start Signaling

A
  • When the phone is lifted off hook, the phone con-
    nects the two wires (tip and ring), causing an electrical signal (48V DC voltage) to flow from the phone
    company central office (CO) into the phone.
  • Home Environments use loop start signaling.
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3
Q

Glare Issue

A
  • 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 be-
    fore the phone has a chance to ring.
  • Not so much a big issue for Home Environments but causes issue in Business Environments.
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4
Q

Ground Start Signaling

A
  • Because of glare issue, Business Enviroments use Ground Start Signaling.
  • This type of signaling in PBX system allows the PBX to separate an answering phone from an incoming phone line, reducing the problem of glare.
  • To receive a dial tone from CO, the PBX must send a ground signal on the wires; this intentionally signals to the CO tht an outgoing call is going to happen.
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5
Q

Two issues with Analog

A
  1. Problem: Analog Signals problem of signal fading over long distances.
  • Solution: Phone Company installed Repeater devices to regenerate the signals as it became weak.
  • Side Effect: Unfortunately, as the analog signal was regenerated, the repeater device was unable to differentiate between the voice traveling over the wire and the line noises.
  1. Problem: With analog connections, the sheer number of wires the phone company had to run to support a large geographical area. With a large number of phones and each phone requiring two wires, the bundles of wire became massive and difficult to maintain.
    * Solution: Digital Signals were developed to send multiple calls over a single wire.
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6
Q

Digital Signals

A
  • Digital signals use numbers to represent levels of voice instead of a combination of electrical signals.
  • Digitizing voice – the process of changing analog voice signals into a series of numbers that you can use to put the voice back together at the other end of the line. (each number sent represents a sound that someone made while speaking into a telephone)
  • Today’s network devices can easily transmit a numeric value any distance a cable can run without any degradation or line noise.
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7
Q

Time-Division Multiplexing

(TDM)

A
  • 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, 4-wire path.
  • The multiple conversations have been digitized (the multiple conversations have been digitized, the numeric values are transmitted in specific time slots that differentiate the separate conversations), the numeric values are transmitted in specific time slots that differentiate the separate conversations.
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8
Q

Issue with Digital Signaling

A
  1. Problem: Phone companies generate 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.
  • Solution: Two Primary Styles of Signaling were created:
    • CAS, and CCS
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9
Q

Channel Associated Signaling (CAS)

A

Signaling information is transmitted using the same bandwidth as the voice.

  • Uses Robbbed-bits -“steals” binary bits that would typically have been used to communicate voice information and use them for signaling.
  • 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), the eighth bit (also called the least significant bit) is stolen for the voice devices to transmit signaling information.
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10
Q

Common Channel Signaling (CCS)

A

Signaling information is transmitted using a separate, dedicated signaling channel.

  • Dedicates 1 of 24 the DS0 channels from a T1 or E1 link for signaling info. This is 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.
  • T1 lines, the 24th time slot (channel) is always the signaling channel.
  • E1 lines, the 17th time slot (channel) is always the signaling channel.
  • The most popular protocol used is Q.931, which is the signaling protocol used for ISDN circuits.
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11
Q

Q.931

A
  • The most popular signaling protocol used for ISDN circuits.
  • A signaling protocol used by ISDN CCS implementation - call setup, tear down in CCS.
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12
Q

Public Switched Telephone Network (PSTN)

A
  • Its primary purpose is to establish worldwide pathways to allow people to easily connect, converse, and disconnect.
  • The modern PSTN is now a worldwide network (much like the Internet).
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13
Q

Pieces that make up the PSTN

A
  • 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.
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14
Q

Why use PBX or Key system in business environments

A

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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15
Q

Key System

A
  • Geared around Small Business Environments (less than 50 users)
  • Typically support fewere features and have a “shared line” - meaning if user1 were to use line 1 of the 4 phone lines, line 1 of 4 would appear busy for all users in that office.
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16
Q

Private Branch Exchange (PBX)

A

Looks like a large box full of cards. Each card:

  • 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.

Note: PBX consideration: From a network equipment perspective, it is a “single point of failure”; although true, most PBX systems offer 99.99% uptime with lifespan of 7 to 10 years.

17
Q

Signaling System 7 (SS7)

A

A common signaling protocol must be used, simialrt to the way TCP/IP operates in the data realm, the voice signaling protocol used around the world is SS7.

  • SS7 is an out-of-band (CCS-style) signaling method used to communicate, call setup, routing, billing, and info messages between 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 info signaling (such as ring back) to the calling device.

Note: SS7 is primarily a telephony service provider technology, you do not typically directly interface with SS7.

18
Q

PSTN numbering scheme

A

Just as data networks use IP addressing to organize and locate resources, voice networks use a numbering plan to organize and locate telephones all around the world.

When connecting to the PSTN, you must use a valid, E.164 standard address for your telephone system.

19
Q

E.164

A

E.164 is an international numbering plan created by the International Telecommunication Union (ITU).

20
Q

International Telecommunication Union (ITU)

A
  • Country code
  • National destination code
  • Subscriber number

Note: E.164 numbers are limited to a maximum length of 15 digits.

1(country code) - 514-555 (National Destination code) - 1234 (Subscriber number)

21
Q

North American Numbering Plan (NANP)

A
  • Country code
  • Area code
  • Central office or exchange code
  • Station code
22
Q

Quantization

A
  • Dr. Nyquist found he could accurately reconstruct audio streams by taking samples (a single byte) that numbered twice the highest audio frequency used.
  • This process of converting the analog wave into digital, numeric values is known as quantization
23
Q

Nyquist Theorem

A

Describes the method of converting analog audio signals into digital format by sampling at twice the highest frequency of the audio.

  • Nyquist believed that you can accurately reproduce an audio signal by sampling at twice the highest frequency.
  • A sample is a numeric value that consumes a single byte of information.
24
Q

G.711 codec

A
  • Uncompressed audio codec consuming 64 kbps of bandwidth
  • the “common ground” between all VoIP devices.
  • MOS = 4.1
  • NON-Cisco
25
Q

There are two forms of the G.711 codec

A

μ-law (used primary in the United States and Japan)

a-law (used everywhere else).

26
Q

Compressed codec

A

The process of 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, it is able to reduce bandwidth down via kbps for each call; a fairly massive reduction in bandwidth.

27
Q

G.729

A
  • Compressed audio codec consuming 8 kbps of bandwidth.
  • MOS = 3.9
  • Cisco
28
Q

G.728

A
  • Compressed audio codec consuming 16 kbps of bandwidth.
  • MOS =3.61
29
Q

G.726

A
  • Compressed audio codec consuming 32 kbps of bandwidth.
  • MOS = 3.85
30
Q

Mean Opinion Score (MOS)

A

In voice digitization years, to rate the quality of the various voice codecs, the Mean Opinion Score was created.

A listener listens to a called say a phrase, and rates the clarity of the phrase on a scale of 1-5.

31
Q

Audio Codec Bandwidth, MOS values chart

A
  • G.711 - Bandwidth 64 kbps, MOS 4.1
  • G.726 - Bandwidth 32kbps, MOS 3.85
  • G.728 - Bandwidth 16kbps, MOS 3.61
  • G.729 - Bandwidth 8 kbps, MOS 3.92
32
Q

Digital Signal Processors (DSP)

A
  • Responsible for converting voice to digitized packets for transmission.
  • Takes load off processor, similar to video cards.
  • Looks like SIMMS
33
Q

Real-time Transport Protocol (RTP)

A
  • RTP operates at the transport layer of the OSI model on top of UDP.
  • Assigns an EVEN port numbers from 16384 – 327676.
  • RTP adds time stamps and sequence numbers to the header information.

o Sequence numbers – This allows the remote device to put the packets back in order when it receives them at the remote end
o Time stamp – Uses a buffer to remove jitter (slight delays) between the packets to give a smooth audio playout.

34
Q

Real-time Transport Control Protocol (RTCP)

A
  • Uses ODD port number from 16384 – 327676.
  • Sends packet every 5 seconds.
  • Once call is established, it delivers stats:

o Packet count
o Packet delay
o Packet loss
o Jitter (delay)