Foundation Summary

The "Foundation Summary" section of each chapter lists the most important facts from the chapter. Although this section does not list every fact from the chapter that will be on the CCDA exam, a well-prepared CCDA candidate should at a minimum know all the details in each "Foundation Summary" before taking the exam.

This chapter covered the following topics that you need to master for the CCDA exam:

■ Traditional voice architectures—The architecture of TDM voice networks. You must understand PSTN technologies and limitations.

■ Integrated multiservice networks—IP telephony architectures and components.

■ IPT design—Design issues, QoS mechanisms, and IPT best practices. Table 15-8 summarizes technologies and concepts used in voice network design.

Table 15-8 Voice Technologies

Technology

Description

BHT

Busy-hour traffic. Expressed in Erlangs.

CCS

Centum Call Second. One call on a channel for 100 seconds.

CDR

Call Detail Record.

FXS

Foreign Exchange Station.

FXO

Foreign Exchange Office.

E&M

Ear and mouth—analog trunk.

Erlang

Measure of total voice traffic volume in one hour. 1 Erlang = 360 CCS.

VAD

Voice Activity Detection.

RTP

Real-time Transport Protocol. Carries coded voice. Runs over UDP.

RTCP

RTP Control Protocol.

Codec

Coder-decoder. Transforms analog signals into digital bit streams.

H.323

ITU framework for multimedia protocols. Used to control Cisco IOS gateways.

MGCP

Media Gateway Control Protocol. Used to control IOS gateways.

continues continues

Table 15-8 Voice Technologies (Continued)

Technology

Description

SIP

Session Initiation Protocol. IETF framework for multimedia protocols.

SS7

Allows voice and network calls to be routed and controlled by central call controllers. Permits modern consumer telephone services. Protocol used in the PSTN.

PSTN

Public Switched Telephone Network.

DTMF

Dual-Tone Multifrequency dialing.

PBX

Private Branch Exchange.

GoS

Grade of service. The probability that a call will be blocked when attempting to seize a circuit.

Centrex

With Centrex services, the CO acts as the company's voice switch, giving the appearance that the company has its own PBX.

IVR

Interactive Voice Response systems provide recorded announcements, prompt callers for key options, and provide information.

ACD

Automatic Call Distribution systems route calls to a group of agents.

Table 15-9 summarizes the different types of codecs used for voice coding. Table 15-9 Codec Standards

Codec

Bit Rate

MOS

Description

G.711u

64 kbps

4.1

PCM. Mu-law version used in North America and Japan. Samples speech 8000 times per second, represented in 8 bits.

G.711a

64 kbps

4.1

PCM. A-law used in Europe and international routes.

G.723.1

6.3 kbps

3.9

Multipulse Excitation-Maximum Likelihood Quantization (MPE-MLQ).

G.723.1

5.3 kbps

3.65

Algebraic Code-Excited Linear Prediction (ACELP).

G.726

16/24/32/ 40 kbps

3.85

Adaptive Differential Pulse-Code Modulation (AD-PCM).

G.728

16 kbps

3.61

Low-Delay CELP (LDCELP).

G.729

8 kbps

3.92

Conjugate Structure ACELP (CS-ACELP).

Table 15-10 summarizes the IPT functional areas. Table 15-10 IPT Functional Areas

IPT Functional Area

Description

Service applications

Unity, IVR, TAPI interface

Call processing

Cisco CM

Client Endpoints

IP phones, digital and analog gateways

Voice Enabled Infrastructure

Layer 2 and Layer 3 switches and routers

Table 15-11 summarizes protocols used in VoIP networks.

Table 15-11 Significant Protocols in VoIP Networks

Protocol

Description

DHCP

Dynamic Host Control Protocol. Provides IP address, mask, gateway, DNS address, and TFTP address.

DNS

Domain Name System. Provides the IP address of the TFTP server.

TFTP

Trivial File Transfer Protocol. Provides the IP phone configuration and operating system.

SSCP

Skinny Station Control Protocol. Establishes calls between IP phones and CM.

RTP

Real-time Transport Protocol. Carries codec voice streams.

RTCP

Real-time Transport Control Protocol. Controls RTP streams.

H.323

ITU framework standard. Used to control Cisco IOS gateways.

SIP

Session Initiation Protocol. An IETF replacement for H.323.

Table 15-12 summarizes the different schemes used for QoS.

Table 15-12 QoS Scheme Summary

QoS Scheme

Description

CRTP

RTP header compression. Reduces header overhead from 40 bytes to 2 to 4 bytes.

LFI

Link Fragmentation and Interleaving. Fragments large data packets and interleaves VoIP packets between them.

PQ-WFQ

Also known as IP RTP priority. Uses a single strict queue for RTP traffic. All other traffic in WFQ.

continues continues

Table 15-12 QoS Scheme Summary (Continued)

QoS Scheme

Description

LLC

Also known as PQ-CBWFQ. Uses a single strict queue for RTP traffic. Differentiated CoS available for all other traffic.

CAC

Call Admission Control. Reroutes voice calls to the PSTN.

Q&A

As mentioned in the Introduction, you have two choices for review questions: here in the book or the exam questions on the CD-ROM. The answers to these questions appear in Appendix A.

For more practice with exam format questions, use the exam engine on the CD-ROM.

1. True or false: LLQ is recommended for VoIP networks.

2. True or false: H.323 is an IETF standard, and SIP is an ITU standard for multimedia protocols.

3. True or false: An Erlang is a unit that describes the number of calls in an hour.

4. What do you implement to stop packets from being transmitted when there is silence in a voice conversation?

5. The variable delay of received VoIP packets is corrected with what kind of buffers?

6. True or false: Common Channel Signaling uses a separate channel for signaling.

7. True or false: FXO ports are used for phones, and FXS ports connect to the PSTN.

8. True or false: SS7 provides mechanisms for exchanging control and routing messages in the PSTN.

9. An organization uses what kind of system to gather and provide information for the customer before transferring her to an agent?

10. An organization uses what kind of system to route calls to agents based on the agent skill group or call statistics?

11. In addition to codec selection, both_and_can be used to reduce the bandwidth of VoIP calls.

12. Label each of the following delays as fixed or variable:

a. Processing b. Dejitter buffer c. Serialization d. Queuing e. Propagation

13. How can you reduce serialization delay?

14. Which two queuing techniques use a strict priority queue for RTP traffic?

15. True or false: The maximum one-way delay in the G.114 recommendation for acceptable voice is 200 ms.

16. True or false: FRF.12 is an LFI standard used in networks with VoFR and VoIP over Frame Relay.

17. An assessment of a network determines that the average round-trip time between two sites is 250 ms. Can an IPT solution be implemented between the sites?

18. Match each protocol with its description:

i. DHCP

ii. SSCP

iii. RTP

v. TFTP

a. Transports coded voice streams b. Controls Cisco IOS gateways c. Provides call signaling between Cisco IP phones and CM

d. Provides IP address e. Provides phone configuration

19. Match each CM deployment model with its description:

i. Single-site deployment ii. Distributed WAN

iii. Centralized WAN

a. Single CM cluster with SRST at remote sites b. Single CM cluster implemented in a large building c. Multiple CM clusters

20. Match each component with its Cisco IPT functional area:

ii. Layer 3 switch iii. Digital gateway iv. Unity a. Service applications b. Call processing c. Client Endpoint d. Infrastructure

21. Which standard establishes specifications for call setup and packet formats for VoFR?

22. Which protocol is preferred for inter-PBX trunks?

d. DTMF

23. CRTP compresses the IP/UDP/RTP header to what size?

a. 2 or 4 bytes b. 2 or 5 bytes c. 40 bytes d. It compresses the RTP header only

24. The steps of converting an analog signal to digital format occur in which order?

a. Sampling, filtering, digitizing b. Filtering, sampling, digitizing c. Digitizing, filtering, sampling d. Sampling, digitizing, filtering

25. Digitizing is divided into which two processes?

a. Filtering and sampling b. Expanding and filtering c. Companding, and quantizing and coding d. Sampling, and quantizing and coding

26. Which of the following are goals of IP telephony?

a. Use the existing IP infrastructure b. Provide lower cost of ownership c. Provide greater flexibility in voice communications d. All of the above

27. An analysis of a 384-kbps WAN link shows IPT calls being delayed when large file transfers take place. The circuit is running at 45 percent utilization. What QoS scheme(s) should be implemented to alleviate this?

a. CQ and cRTP

b. LFI and cRTP

d. All of the above

28. Which codec is recommended for use in WAN links?

29. Which technology reduces the amount of bandwidth used? (Choose all that apply.)

c. cRTP

30. Which of the following statements is true?

a. CAC prevents voice calls from affecting other voice calls.

b. CAC prevents voice calls from affecting data bandwidth.

c. CAC prevents data from affecting voice calls.

d. CAC prevents data from affecting other data traffic.

31. What IPT component contains the dial plan and is used to register IP phones?

a. Gateway b. Unity server c. Gatekeeper d. Cisco Unified CallManager

Use both the scenario described in the following paragraph and Figure 15-18 to answer the following questions.

The client has an existing Frame Relay network, as shown in Figure 15-18. The network has a large site and 50 small remote sites. The client wants a design for a VoIP network. The client wants to provide differentiated CoS for the voice, Systems Network Architecture (SNA), FTP, and other traffic.

Figure 15-18 Client's Current Frame Relay Network

Remote Sites

Figure 15-18 Client's Current Frame Relay Network

Remote Sites

32. Based on the current network diagram, which Cisco IPT deployment model should you recommend?

33. What feature should you recommend to provide call processing in the event of a WAN failure?

34. Which queuing technique should you recommend?

35. For Site 1, the current data traffic is 512 kbps, and video traffic is 0. What is the minimum bandwidth required to support four concurrent VoIP G.729 calls plus the data traffic to the site?

36. Should you implement a CallManager cluster?

37. What feature can you use to reduce bandwidth over the WAN links?

38. Which LFI technique should you use to reduce the serialization delay?

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