Andy’s Cisco Exam Notes

Conquering Cisco, one exam at a time!

List the four key quality issues with converged networks?

Posted by engineeringtop on January 23, 2008

  • available bandwidth
  • end to end delay
  • delay variation
  • packet loss

Posted in IP Quality of Service | Leave a Comment »

With QoS feature in place, there can be up to 10 concurrent VoIP calls over a company WAN link. Is there need for CAC? With no CAC, what will happen when there are more than 10 concurrent calls?

Posted by engineeringtop on January 23, 2008

Yes, CAC compliments QoS configurations so that WAN bandwidth does not get oversubscribed. With no CAC over 10 concurrent calls, packets will be dropped indiscriminately and experience poor voice quality on all calls not just on those after 10.

Posted in Cisco VoIP Implementations | Tagged: , | Leave a Comment »

What is CAC?

Posted by engineeringtop on January 23, 2008

Call Admission Control is a feature that is configured to limit the number of concurrent calls. CAC is configured so that WAN bandwidth does not get oversubscribed by VOIP calls.  CAC compliments QoS configurations.

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List at least 3 voice gateway functions on a Cisco router

Posted by engineeringtop on January 23, 2008

On a Cisco router, voice gateway functions include the following:

  • Connect traditional telephony devices
  • Convert analog signals to digital and vice versa
  • Encapsulate  digital voice into IP packets
  • Perform voice compression
  • Provide DSP resources for conferencing and transcoding
  • Provide Cisco Survivable Remote Site (SRST)
  • Act as the call agent (Cisco Unified CallManager Express)

Posted in Cisco VoIP Implementations | Tagged: , , , | Leave a Comment »

List at least three important components of enterprise voice implementations.

Posted by engineeringtop on January 22, 2008

  • gateway
  • gatekeeper
  • Call Manager
  • Cisco IP Phones

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What is VAD?

Posted by engineeringtop on January 22, 2008

VAD stands for Voice Activity Detection and is a feature available in voice-enabled networks. It suppresses the transmission of silence; therefore, it might result in up to 35 percent bandwidth savings. The success of VAD depends on the types of audio, the level of background noise, and other factors.

Posted in Cisco VoIP Implementations | Tagged: , | Leave a Comment »

Briefly list the steps necessary to compute the total bandwidth for a VOIP call.

Posted by engineeringtop on January 22, 2008

  1. Determine the codec and the packetization period
  2. Determine link-specific information (cRTP, Layer 2 protocol, security protocol and tunneling?)
  3. Calculate packetization size (size of voice payload)
  4. Calculate the total frame size
  5. Calculate the packet rate (1/packetization period)
  6. Calculate the total bandwidth

Equation:

Total packet size    =    Total bandwidth

payload size                   nominal bandwidth

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Name at least three tunneling and security protocols and their associated overhead?

Posted by engineeringtop on January 22, 2008

  • IPsec transport mode  30 to 53 bytes
  • IPsec Tunnel mode      50 to 73 bytes
  • L2TP                              24 bytes
  • GRE                                24 bytes
  • MPLS                              4 bytes
  • PPPoE                             8 bytes

Posted in Cisco VoIP Implementations | Tagged: , , , , , , | Leave a Comment »

What are the sizes of Ethernet, 802.1Q, Frame-Relay, and Multilink PPP (MLP) overhead?

Posted by engineeringtop on January 21, 2008

  • Ethernet 18 bytes
  • 802.1Q  22 bytes
  • Frame Relay 6 bytes
  • Multilink PPP (MLP) 6 bytes

Posted in Cisco VoIP Implementations | Tagged: , , , , | Leave a Comment »

What is the relationship between packet rate and packetization period?

Posted by engineeringtop on January 21, 2008

The Packet rate and Packetization period are reciprocal.

For example, if the packetization period is 20ms (.020 seconds) then the packet rate is equal to 1 over 0.020 (1/0.20) or 50 pps (packets per second).

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List at least three factors that influence bandwidth requirements of VOIP?

Posted by engineeringtop on January 21, 2008

  • packet rate and packetization size
  • IP overhead
  • Data link overhead
  • Tunneling overhead

*Packetization size and packet rate are inversely related*

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What is cRTP?

Posted by engineeringtop on January 21, 2008

Compressed RTP (cRTP)  or RTP Header Compression. cRTP reduces the IP, UDP, and RTP headers from 40 to 2 bytes (without a checksum) and to 4 bytes (with a checksum).  cRTP provides a significant bandwidth savings but is is only recommended for use on slow inks (less than 2 Mbps).

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What feature does RTP provide to complement UDP?

Posted by engineeringtop on January 21, 2008

RTP provides Sequence Numbering (Reordering) and Time Stamping.

Posted in Cisco VoIP Implementations | Tagged: , | Leave a Comment »

Which TCP/IP protocols are responsible for transporting voice? What are the sizes of these protocol headers?

Posted by engineeringtop on January 21, 2008

  • IP 20 bytes
  • UDP 8 bytes
  • RTP 12 bytes

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What is a DSP?

Posted by engineeringtop on January 21, 2008

Digital Signal Processor – processors on voice gateways. They provide voice termination, transcoding, and conferencing.

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What is MOS?

Posted by engineeringtop on January 21, 2008

Mean Opinion Score is an ITU Standard technique used for measuring of voice codecs.  MOS values which are subjective and expessed by humans, range from 1 (worst) to 5 (perfect or equivalent to direct conversation.

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Name at least 3 main codec/compression standards and specify their bandwidth requirements

Posted by engineeringtop on January 21, 2008

  • G.711 (PCM) 64Kbps
  • G.726 (ADPCM) 16, 24, 32 Kbps
  • G.728 (LDCELP) 16Kbps
  • G.729 (CS-ACELP) 8Kbps
  • G.729-A 8Kbps

Posted in Cisco VoIP Implementations | Tagged: , , , , | Leave a Comment »

Name and explain the quantization methods used in North America and other countries

Posted by engineeringtop on January 21, 2008

Two variations of logarithmic quantization exist:  μ-law and A-Law.  Bell developed μ-law and it is the method that is most used in North American and Japan.  The ITU modified μ-law and introduced A-Law, which is common in countries outside of North America, except Japan.  When signals have to be exchanged between a μ-law country and an A-Law country in the PSTN, the μ-law country must change its signaling to accommodate the A-Law Country.

Posted in Cisco VoIP Implementations | Tagged: , , | Leave a Comment »

What are the two main quantization techniques?

Posted by engineeringtop on January 21, 2008

Linear quantization and logarithmic quantization

Posted in Cisco VoIP Implementations | Leave a Comment »

List the steps for converting digital signals to analog signals

Posted by engineeringtop on January 21, 2008

  1. Decompression (optional)
  2. Decoding and FILTERING
  3. Reconstructing the analog signal

Posted in Cisco VoIP Implementations | Leave a Comment »

Based on the Nyquist theorem, what should be the minimum sampling rate of analog signal?

Posted by engineeringtop on January 21, 2008

The sampling rate must be at least twice the maximum signal frequency. Because the maximum voice frequency over a telephone channel was considered 4000 Hz, based on the Nyquist theorem, a rate of 8000 samples per second is required.

Posted in Cisco VoIP Implementations | Leave a Comment »

List the steps for converting analog signals to digital signals.

Posted by engineeringtop on January 17, 2008

  1. Sampling
  2. Quantization
  3. Encoding
  4. Compression (optional)

Posted in Cisco VoIP Implementations | Tagged: , , , , , , | Leave a Comment »

What are the two main models of call control?

Posted by engineeringtop on January 17, 2008

The two main models of call control are distributed call control and centralized call control. Exampls of distrbuted call control included H.323 and SIP. An example of centralized call control is MGCP.

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List threes stages of a phone call.

Posted by engineeringtop on January 17, 2008

  1. Call setup
  2. Call maintenance
  3. Callterdown

Posted in Cisco VoIP Implementations | Leave a Comment »

List at least two digital interface options th connect VOIP equipment to PBXs or the PSTN

Posted by engineeringtop on January 17, 2008

  • BRI
  • T1 CAS
  • T1  CCS
  • E1 CAS
  • E1 CCS

Posted in Cisco VoIP Implementations | Tagged: , , , , | Leave a Comment »

List at least 3 types of analog interfaces through which legacy analog devices can connect to a VoIP network.

Posted by engineeringtop on January 16, 2008

  • FXS (foreign exchange station)
  • FXO (foreign exchange office)
  • E&M (Earth and Magneto or Ear and Mouth)

Posted in Cisco VoIP Implementations | Tagged: , , , , , | Leave a Comment »

List at least three important components of a packet telephony network

Posted by engineeringtop on January 16, 2008

  • phones
  • gateways
  • Multipoint Control Units (MCU)
  • Application and Database Servers
  • gatekeepers
  • call agents
  • video end points
  • DSP

Posted in Cisco VoIP Implementations | Tagged: , , , , , | Leave a Comment »

List at Least 3 benefits of packet telephony networks.

Posted by engineeringtop on January 16, 2008

  • More efficient use of bandwidth and equipment, and lower transmission costs
  • Consolidated network expenses
  • Improved employee productivity
  • Access to new communication devices

Posted in Cisco VoIP Implementations | Tagged: , | Leave a Comment »