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TOP 5 VOIP CONCEPTS TO KNOW FOR CCNA VOICE
VoIP Basic for CCNA Voice Exam
The demand for IT professionals with a strong network background in Voice and
Telephony technologies is growing.

Brief Info of CCNA Voice Certification
The Cisco Certified Network Associate Voice (CCNA Voice) validates associate-level
knowledge and skills required to administer a voice network. The Cisco CCNA Voice
certification confirms that the required skill set for specialized job roles in voice
technologies such as voice technologies administrator, voice engineer, and voice
manager. It validates skills in VoIP technologies such as IP PBX, IP telephony, handset,
call control, and voicemail solutions.

The CCNA Voice Certification assesses skills and knowledge related to the Cisco
Unified Communications Manager 8.0 (CUCM) 8.0. It is typically employed by large
organizations such as governments, large companies, and colleges. Additionally, the
CCNA Voice Certification assesses skills and knowledge related to the Cisco
CallManager Express (CME) and Cisco Unity Express (CUE) solutions typically used by
medium and small organizations such as companies with less than 2,000 employees,
retail businesses, and small school districts.

The implementation of Voice over IP (VoIP) has become more common as an
organizational alternative to traditional voice services. So here we talk about reviews
the basic concepts of voice operations, and provides a base of knowledge for
deploying and operating these technologies.

The CCNA Voice certification provides the candidate an associate level of knowledge
in several common voice implementations. These implementations can include
devices like IP phones, gateways and gatekeepers, as well as voice concept basics like
call control and QoS. The CCNA Voice certification holder holds such titles as Voice
Administrator, Voice Technician or Telecom Analyst, and has the ability to administer
these types of voice implementations.

Sampling & Quantization (PCM)
Sampling takes a continuous signal and converts it into a discrete signal; quantization
is then used to assign a value representing the sample. The amount of values
available depends on the quantization type used; typically, in voice communications,
8-bit quantization is used. The standard codec that is used on voice networks is
G.711 which samples an 8-bit signal (via quantization) at 8000 times a second.
Signals are digitally represented by 256 values from -127 (11111111) through 127
(01111111) with the first digit representing value polarity. An example of this is
shown in Figure 1.


                                                      http://blog.router-switch.com/
Voice Interfaces (Analog & Digital)
Voice interfaces are what connect the voice device to the outside world. When
implementing VoIP on modern equipment, it is still necessary in most
implementations to utilize some type of voice interface even with the popularity of
VoIP compatible phones. There are two groups of voice interfaces that are used to
connect to other voice equipment, analog and digital.

Analog Voice Interfaces
There are three major types of analog interface, these include:
• Foreign Exchange Office (FXO)
• Foreign Exchange Station (FXS)
• Earth and Magneto (E&M)

FXO and FXS interfaces are used in combination with one another. For example, a
standard phone is a FXO device which connects to a FXS port on a voice device.
Another example of this would be a FXS port on another telephone device which
connects to a FXO port.

The typical duty of an FXO interface is to provide on and off hook indication which is
an indication of whether the analog loop is open or closed.

The typical duty of an FXS interface is to provide battery current, ring voltage and
dial-tone.
E&M is an interface that is traditionally used to connect between voice switch
devices. There are five different types of E&M signaling, with Type 1 being the most
common in the U.S.

Digital Voice Interfaces
                                                     http://blog.router-switch.com/
There are two main voice digital interfaces, Basic Rate Interface (BRI) and Primary
Rate Interface (PRI). A BRI provides up to 2 channels of voice with a smaller signaling
channel. A PRI provides up to 23 channels of voice with an additional 24th channel
which is used for signaling. Both of these interfaces are used more commonly in
voice implementations as a replacement for an E&M analog interface. The use of BRI
and PRI is also common when delivering voice to customer premises.

Call Legs & Dial Peers
A call leg is a commonly used term to reference a logical connection between two
devices. Depending on the number of devices that are used to connect a voice
connection from source to destination, the number of call legs can get quite high.

A dial peer on a Cisco device refers to a physical configuration of a call leg. There are
different dial peer types including a POTS dial peer and a Voice-network dial peer
(commonly VoIP). Figure 2 shows a common example of dial peer implementation.




Codec
A codec is a process of coding and decoding a signal. Typically, a voice codec is used
to compress the voice signal into a specific bit-rate in order to be transferred
between devices. The most commonly used codec is G.711 which is associated with
uncompressed (mostly) voice. Table 1 shows a list of the most commonly used
modern codecs.
Codec        Acronym      Name                               Bit Rate
G.711        PCM          Pulse Code Modulation              64-kbps
G.722        SB-ADPCM Sub-Band ADPCM                         48, 56, 64-kbps
G.722.1      MLT          Modulated Lapped Transform         24 and 32-kbps
G.722.2      ACELP        Algebraic Code Excited Linear 6.60,          8.85,    12.65,
                          Prediction Coder                   14.25, 15.85, 18.25,
                                                             19.85,       23.05     or
                                                             23.85-kbps
G.723.1      ACELP        Algebraic Code Excited Linear 5.3-kbps
(5.3-kbps)                Prediction Coder
G.723.1      MP-MLQ       Multi Pulse-Maximum Likelihood 6.3-kbps
(6.3-kbps)                Quantization
G.726        ADPCM        Adaptive Differential Pulse Code 16, 24, 32 and 40-kbps
                          Modulation
G.728        LDCELP       Low Delay Code Exited Linear 16-kbps
                          Prediction

                                                       http://blog.router-switch.com/
G.729       CS-ACELP     Conjugate Structure Algebraic CELP 8-kbps
                         (High Complexity)
G.729A      CS-ACELP     Conjugate Structure Algebraic CELP 8-kbps
            Annex A      Annex A (Lower Complexity then
                         G.729) (Medium Complexity)
G.729B      CS-ACELP     Conjugate Structure Algebraic CELP 8-kbps
                         (G.729 with silence compression
                         support) (High Complexity)
G.729AB     CS-ACELP     Conjugate Structure Algebraic CELP 8-kbps
                         (G.729 A with silence compression
                         support) (Medium Complexity)

Numbering Plan
The numbering plan is simply a plan that is followed in the allocation of phone
numbers. Most people are familiar with this assignment of various area codes and
prefixes throughout the country. In North America, this plan is called the North
American Numbering Plan (NANP). This plan defines the use of a 10 digit phone
number which includes a three digit Numbering Plan Area (NPA) that is used to
identify the area (also called the area code), the NXX that is used to identify the
central office and a four digit station code which identifies the specific end device.

The overview of voice operations covered in this article should help in understanding
the fundamental concepts that are part of a voice certification.
---Original reading at Petri IT Knowledgebase

More Related CCNA Voice Certification Tips:
10 Things You Should Know About the Cisco CCNA Voice Certification
The Benefits of Cisco VoIP




                                                     http://blog.router-switch.com/

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Top 5 voip concepts to know for ccna voice

  • 1. TOP 5 VOIP CONCEPTS TO KNOW FOR CCNA VOICE VoIP Basic for CCNA Voice Exam The demand for IT professionals with a strong network background in Voice and Telephony technologies is growing. Brief Info of CCNA Voice Certification The Cisco Certified Network Associate Voice (CCNA Voice) validates associate-level knowledge and skills required to administer a voice network. The Cisco CCNA Voice certification confirms that the required skill set for specialized job roles in voice technologies such as voice technologies administrator, voice engineer, and voice manager. It validates skills in VoIP technologies such as IP PBX, IP telephony, handset, call control, and voicemail solutions. The CCNA Voice Certification assesses skills and knowledge related to the Cisco Unified Communications Manager 8.0 (CUCM) 8.0. It is typically employed by large organizations such as governments, large companies, and colleges. Additionally, the CCNA Voice Certification assesses skills and knowledge related to the Cisco CallManager Express (CME) and Cisco Unity Express (CUE) solutions typically used by medium and small organizations such as companies with less than 2,000 employees, retail businesses, and small school districts. The implementation of Voice over IP (VoIP) has become more common as an organizational alternative to traditional voice services. So here we talk about reviews the basic concepts of voice operations, and provides a base of knowledge for deploying and operating these technologies. The CCNA Voice certification provides the candidate an associate level of knowledge in several common voice implementations. These implementations can include devices like IP phones, gateways and gatekeepers, as well as voice concept basics like call control and QoS. The CCNA Voice certification holder holds such titles as Voice Administrator, Voice Technician or Telecom Analyst, and has the ability to administer these types of voice implementations. Sampling & Quantization (PCM) Sampling takes a continuous signal and converts it into a discrete signal; quantization is then used to assign a value representing the sample. The amount of values available depends on the quantization type used; typically, in voice communications, 8-bit quantization is used. The standard codec that is used on voice networks is G.711 which samples an 8-bit signal (via quantization) at 8000 times a second. Signals are digitally represented by 256 values from -127 (11111111) through 127 (01111111) with the first digit representing value polarity. An example of this is shown in Figure 1. http://blog.router-switch.com/
  • 2. Voice Interfaces (Analog & Digital) Voice interfaces are what connect the voice device to the outside world. When implementing VoIP on modern equipment, it is still necessary in most implementations to utilize some type of voice interface even with the popularity of VoIP compatible phones. There are two groups of voice interfaces that are used to connect to other voice equipment, analog and digital. Analog Voice Interfaces There are three major types of analog interface, these include: • Foreign Exchange Office (FXO) • Foreign Exchange Station (FXS) • Earth and Magneto (E&M) FXO and FXS interfaces are used in combination with one another. For example, a standard phone is a FXO device which connects to a FXS port on a voice device. Another example of this would be a FXS port on another telephone device which connects to a FXO port. The typical duty of an FXO interface is to provide on and off hook indication which is an indication of whether the analog loop is open or closed. The typical duty of an FXS interface is to provide battery current, ring voltage and dial-tone. E&M is an interface that is traditionally used to connect between voice switch devices. There are five different types of E&M signaling, with Type 1 being the most common in the U.S. Digital Voice Interfaces http://blog.router-switch.com/
  • 3. There are two main voice digital interfaces, Basic Rate Interface (BRI) and Primary Rate Interface (PRI). A BRI provides up to 2 channels of voice with a smaller signaling channel. A PRI provides up to 23 channels of voice with an additional 24th channel which is used for signaling. Both of these interfaces are used more commonly in voice implementations as a replacement for an E&M analog interface. The use of BRI and PRI is also common when delivering voice to customer premises. Call Legs & Dial Peers A call leg is a commonly used term to reference a logical connection between two devices. Depending on the number of devices that are used to connect a voice connection from source to destination, the number of call legs can get quite high. A dial peer on a Cisco device refers to a physical configuration of a call leg. There are different dial peer types including a POTS dial peer and a Voice-network dial peer (commonly VoIP). Figure 2 shows a common example of dial peer implementation. Codec A codec is a process of coding and decoding a signal. Typically, a voice codec is used to compress the voice signal into a specific bit-rate in order to be transferred between devices. The most commonly used codec is G.711 which is associated with uncompressed (mostly) voice. Table 1 shows a list of the most commonly used modern codecs. Codec Acronym Name Bit Rate G.711 PCM Pulse Code Modulation 64-kbps G.722 SB-ADPCM Sub-Band ADPCM 48, 56, 64-kbps G.722.1 MLT Modulated Lapped Transform 24 and 32-kbps G.722.2 ACELP Algebraic Code Excited Linear 6.60, 8.85, 12.65, Prediction Coder 14.25, 15.85, 18.25, 19.85, 23.05 or 23.85-kbps G.723.1 ACELP Algebraic Code Excited Linear 5.3-kbps (5.3-kbps) Prediction Coder G.723.1 MP-MLQ Multi Pulse-Maximum Likelihood 6.3-kbps (6.3-kbps) Quantization G.726 ADPCM Adaptive Differential Pulse Code 16, 24, 32 and 40-kbps Modulation G.728 LDCELP Low Delay Code Exited Linear 16-kbps Prediction http://blog.router-switch.com/
  • 4. G.729 CS-ACELP Conjugate Structure Algebraic CELP 8-kbps (High Complexity) G.729A CS-ACELP Conjugate Structure Algebraic CELP 8-kbps Annex A Annex A (Lower Complexity then G.729) (Medium Complexity) G.729B CS-ACELP Conjugate Structure Algebraic CELP 8-kbps (G.729 with silence compression support) (High Complexity) G.729AB CS-ACELP Conjugate Structure Algebraic CELP 8-kbps (G.729 A with silence compression support) (Medium Complexity) Numbering Plan The numbering plan is simply a plan that is followed in the allocation of phone numbers. Most people are familiar with this assignment of various area codes and prefixes throughout the country. In North America, this plan is called the North American Numbering Plan (NANP). This plan defines the use of a 10 digit phone number which includes a three digit Numbering Plan Area (NPA) that is used to identify the area (also called the area code), the NXX that is used to identify the central office and a four digit station code which identifies the specific end device. The overview of voice operations covered in this article should help in understanding the fundamental concepts that are part of a voice certification. ---Original reading at Petri IT Knowledgebase More Related CCNA Voice Certification Tips: 10 Things You Should Know About the Cisco CCNA Voice Certification The Benefits of Cisco VoIP http://blog.router-switch.com/