Example: CAC Deployment

The example shows a scenario with two sites, both of which have three phones connected to a VoIP gateway through a PBX. The two gateways are connected via an IP network. The network is designed for a maximum of two concurrent calls.

• IP network (WAN) is only designed for two concurrent voice calls.

• If CAC is not deployed, a third call can be set up, causing poor quality for all calls.

• After CAC is deployed, the third call is blocked.

Initially, CAC was not used. Whenever there were three active calls, all of them experienced severe voice quality issues.

CAC is deployed to avoid this problem. The gateways are configured to allow no more than two calls at the same time. When a third call is attempted, the call is blocked. With the new configuration using CAC, no voice quality problems should be experienced.

The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,

Summary

This topic summarizes the key points that were discussed in this lesson.

Summary

• Converged enterprise networks include components supporting VoIP, such as gateways, gatekeepers, Cisco Unified CallManager, and IP phones.

• Cisco ISRs provide voice capabilities, including gateway, call agent, and DSP functions.

• Cisco Unified CallManager provides call processing and signaling services and provides access to applications from IP phones.

• IP deployment models include single site, multisite (centralized and distributed), and clustering over WAN.

• Cisco IOS dial peers are used at the gateway to configure a local dial plan.

• CAC is a method that prevents bandwidth exhaustion caused by too many voice calls.

Module Summary

This topic summarizes the key points that were discussed in this module.

Module Summary

• VoIP networks are composed of multiple components, using either distributed or centralized call control methods.

• In VoIP networks, analog signals have to be converted into digital format. DSPs provide this conversion by sampling, quantization, encoding, and optional compression.

• Digitized voice is encapsulated into RTP, UDP, and IP headers. To reduce bandwidth requirements, these headers can be compressed on a link-by-link basis.

• The total bandwidth required for a VoIP call depends on the codec, packetization period, and encapsulation overhead.

• Based on the network topology and size, different IP telephony deployment models can be utilized.

VoIP networks include different components in order to be able to assure voice calls. First analog voice signals must be converted to digital format. DSP modules can be used to perform conversion by sampling, quantization and encoding. Optionally compression can be used too.

Payload of the voice packet is small and RTP header compression is used as voice headers are significant part of the packet. RTP header compression is comnpressing IP RTP and UDP part of the header to reduce bandwidth. Bandwidth must take into account layer 2 header too and optionally additional overhead is added in case of encryption. Total bandwidth depens on different codecs used too. IP telephony deployments can be using distributed or centralized call control methods and different deployment models can be utilized.

The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,

Module Self-Check

Use the questions here to review what you learned in this module. The correct answers and solutions are found in the Module Self-Check Answer Key.

Q1) Which is not a benefit of packet telephony networks? (Source: Introducing VoIP Networks)

A) lower transmission costs

B) access to new communication devices C) improved productivity

D) less packetization overhead

Q2) A _____ is used for mixing audio streams in conference calls. (Source: Introducing VoIP Networks)

Q3) Which stage of a call involves admission control? (Source: Introducing VoIP Networks)

A) call setup B) call blocking C) call teardown D) call maintenance

Q4) Which protocol is an example of centralized call control? (Source: Introducing VoIP Networks)

A) RSVP

B) H.235 C) H.323

D) MGCP

Q5) Cisco IOS voice gateways use the _____ interface when connecting to an analog phone. (Source: Introducing VoIP Networks)

Q6) Which two statements are true about digital interfaces? (Choose two.) (Source:

Introducing VoIP Networks)

A) T1 CAS has 24 clear 64-kbps voice channels.

B) E1 CCS has 30 voice channels.

C) E1 CAS has 30 voice channels.

D) T1 CCS has 24 clear 64-kbps voice channels.

E) T1 CCS has 23 voice channels with robbed-bit signaling.

F) E1 CCS has 31 voice channels.

Q7) Which is the correct order for analog-to-digital conversion? (Source: Digitizing and Packetizing Voice)

A) compression, sampling, encoding, quantization B) encoding, sampling, quantization, compression C) sampling, encoding, quantization, compression D) sampling, quantization, encoding, compression

Q8) During digital-to-analog conversion, what is reconstructed from the PAM signals?

(Source: Digitizing and Packetizing Voice)

A) PCM

B) analog signal C) digital signal D) codeword

Q9) According to the Nyquist theorem, what minimum rate is needed to sample analog frequencies up to 9000 Hz? (Source: Digitizing and Packetizing Voice)

A) 8000 Hz B) 8 kHz C) 18 kHz D) 4500 Hz

Q10) During encoding, how many bits are used to indicate the segment number? (Source:

Digitizing and Packetizing Voice)

Q11) Which statement is true about codec bit rates? (Source: Digitizing and Packetizing Voice)

A) G.729 and G.728 both use 8 kbps.

B) G.729A uses 8 kbps, and G.711 uses 16, 24, or 32 kbps.

C) G.728 uses 16 kbps.

D) G.711 and G.276 both use 64 kbps.

Q12) A DSP used for a _____-mode conference cannot mix different codecs. (Source:

Digitizing and Packetizing Voice)

Q13) Which is not a characteristic of VoIP packet delivery? (Source: Encapsulating Voice Packets for Transport)

A) Packets can arrive in incorrect order.

B) Packets experience varying delays.

C) VoIP packets are sent over a dedicated circuit of 64 kbps.

D) The total bandwidth of a link is shared by all IP packets.

Q14) Which header is not used for VoIP media packets? (Source: Encapsulating Voice Packets for Transport)

A) TCP

B) IP

C) UDP

D) RTP

Q15) Which headers does an RTP header compression reduce to 2 or 4 bytes? (Source:

Encapsulating Voice Packets for Transport) A) IP, UDP, and RTP

B) IP and TCP

C) RTP

D) UDP and RTP

The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,

Q16) Which two statements are true of packetization of voice? (Choose two.) (Source:

Calculating Bandwidth Requirements)

A) The shorter the packetization period, the smaller the IP packet size.

B) The longer the packetization period, the smaller the IP packet size.

C) The longer the packetization period, the lower the packet rate.

D) The shorter the packetization period, the lower the packet rate.

E) The longer the packetization period, the lower the codec bandwidth.

Q17) Which data-link protocol adds 18 bytes during encapsulation to the VoIP packet?

(Source: Calculating Bandwidth Requirements)

Q18) Which is the approximate amount of overhead added to a 40-byte VoIP packet by IPsec tunnel mode? (Source: Calculating Bandwidth Requirements)

A) more than 300 percent B) more than 35 percent C) more than 100 percent D) up to 35 percent

Q19) Which is not a factor for the calculation of total bandwidth? (Source: Calculating Bandwidth Requirements)

A) packetization period B) packetization size C) codec bandwidth D) encapsulation overhead E) buffer overhead

Q20) _____ can save bandwidth by suppressing transmission during periods of silence.

(Source: Calculating Bandwidth Requirements)

Q21) Which is not a component of an enterprise voice network? (Source: Implementing Voice Support in an Enterprise Network)

A) gateways

B) central office switch C) IP phones

D) Cisco Unified CallManager

Q22) Which telephony feature supports fallback scenarios for remote IP phones when the IP WAN is down? (Source: Implementing Voice Support in an Enterprise Network)

A) VAD

B) AAR

C) SRST

D) cRTP

Q23) Which is not a Cisco Unified CallManager function? (Source: Implementing Voice Support in an Enterprise Network)

A) providing DSPs for transcoding B) call processing

C) signaling and device control D) phone feature administration

Q24) Which Cisco Unified CallManager deployment model causes all signaling traffic to cross the IP WAN? (Source: Implementing Voice Support in an Enterprise Network) A) single site

B) multisite centralized call processing C) multisite distributed call processing D) clustering over WAN

Q25) Which command is not used when configuring a POTS dial peer? (Source:

Implementing Voice Support in an Enterprise Network) A) dial-peer

B) port

C) session target D) destination-pattern

Q26) CAC limits the _____ in a VoIP environment. (Source: Implementing Voice Support in an Enterprise Network)

The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,

Module Self-Check Answer Key

Q1) D

Q2) multipoint control unit Q3) A

Q4) D Q5) FXS Q6) B, C Q7) D Q8) B Q9) C Q10) three Q11) C Q12) single Q13) C Q14) A Q15) A Q16) B, C Q17) Ethernet Q18) C Q19) E Q20) VAD Q21) B Q22) C Q23) A Q24) B Q25) C

Q26) number of calls

Module 3

In document Optimizing Converged Cisco Networks (Page 119-127)