Andy's Cisco Exam Notes

List the four key quality issues with converged networks?

engineeringtop — Wed, 23 Jan 2008 21:59:37 +0000

available bandwidth
end to end delay
delay variation
packet loss

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?

engineeringtop — Wed, 23 Jan 2008 01:41:14 +0000

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.

What is CAC?

engineeringtop — Wed, 23 Jan 2008 01:09:10 +0000

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.

List at least 3 voice gateway functions on a Cisco router

engineeringtop — Wed, 23 Jan 2008 01:00:32 +0000

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)

List at least three important components of enterprise voice implementations.

engineeringtop — Tue, 22 Jan 2008 23:35:00 +0000

gateway
gatekeeper
Call Manager
Cisco IP Phones

What is VAD?

engineeringtop — Tue, 22 Jan 2008 01:20:56 +0000

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.

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

engineeringtop — Tue, 22 Jan 2008 01:00:27 +0000

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

Equation:

Total packet size = Total bandwidth

payload size nominal bandwidth

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

engineeringtop — Tue, 22 Jan 2008 00:02:02 +0000

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

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

engineeringtop — Mon, 21 Jan 2008 23:58:51 +0000

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

What is the relationship between packet rate and packetization period?

engineeringtop — Mon, 21 Jan 2008 23:52:07 +0000

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).

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

engineeringtop — Mon, 21 Jan 2008 23:47:30 +0000

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

*Packetization size and packet rate are inversely related*

What is cRTP?

engineeringtop — Mon, 21 Jan 2008 23:44:52 +0000

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).

What feature does RTP provide to complement UDP?

engineeringtop — Mon, 21 Jan 2008 23:41:04 +0000

RTP provides Sequence Numbering (Reordering) and Time Stamping.

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

engineeringtop — Mon, 21 Jan 2008 23:32:45 +0000

IP 20 bytes
UDP 8 bytes
RTP 12 bytes

What is a DSP?

engineeringtop — Mon, 21 Jan 2008 23:15:19 +0000

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

What is MOS?

engineeringtop — Mon, 21 Jan 2008 23:06:22 +0000

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.

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

engineeringtop — Mon, 21 Jan 2008 04:26:04 +0000

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

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

engineeringtop — Mon, 21 Jan 2008 01:11:36 +0000

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.

What are the two main quantization techniques?

engineeringtop — Mon, 21 Jan 2008 00:47:13 +0000

Linear quantization and logarithmic quantization

List the steps for converting digital signals to analog signals

engineeringtop — Mon, 21 Jan 2008 00:28:58 +0000

Decompression (optional)
Decoding and FILTERING
Reconstructing the analog signal

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

engineeringtop — Mon, 21 Jan 2008 00:20:42 +0000

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.

List the steps for converting analog signals to digital signals.

engineeringtop — Thu, 17 Jan 2008 21:28:21 +0000

Sampling
Quantization
Encoding
Compression (optional)

What are the two main models of call control?

engineeringtop — Thu, 17 Jan 2008 21:24:37 +0000

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.

List threes stages of a phone call.

engineeringtop — Thu, 17 Jan 2008 21:19:08 +0000

Call setup
Call maintenance
Callterdown

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

engineeringtop — Thu, 17 Jan 2008 21:17:46 +0000

BRI
T1 CAS
T1 CCS
E1 CAS
E1 CCS

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

engineeringtop — Wed, 16 Jan 2008 04:59:50 +0000

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

List at least three important components of a packet telephony network

engineeringtop — Wed, 16 Jan 2008 02:16:31 +0000

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

List at Least 3 benefits of packet telephony networks.

engineeringtop — Wed, 16 Jan 2008 01:20:58 +0000

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