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VoIP > MIDTERM Weeks-1-4 > Flashcards

MIDTERM Weeks-1-4 Flashcards

1
Q

ISDN

A

Integrated Services Digital Network

Pair of copper wires using TDM
binary resides on Bearer channels
2 types of ISDN: BRI/PRI
virtual out of band control is a separate private channel for call-tear downs

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2
Q

DTMF

A

Dual-Tone-Multi-Frequency

touch tone (1963)
generates frequencies [1209,1336,1477]
45m/s quickest time between dials
3sec longest duration between dials

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3
Q

What fields are NOT included in UDP?

A

Sequence Numbers

Time Stamps

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4
Q

P.O.T.S

A

Plain Old Telephone Service

voice grade
analog over copper wires

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5
Q

C.A.S

A

Channel Associated Signalling

Extended Superframe (ESF)
- TI framing standard
BiPolar 8-zero Substitution (B8ZS)
- Encoding method on T1 curcuits

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6
Q

Which command will display ALL the TFTP files created by CME?

A

show voice register tftp bind

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7
Q

What is the maximum number of configurable Directory Numbers (DN)?

A

400

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8
Q

What is the maximum number of configurable Pools?

A

110

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9
Q

Digital Voice Encoding

A

Each sample encoded with 8 bits

1 bit = POLARITY
3 bits = SEGMENT BITS
4 bits = STEP BITS

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10
Q

V.E.O

A

Voice Encapsulation Overhead

voice sent in small packets at high rates
IP, UDP, RTP overheads are HUGE
G.729 headers are 2x size of payload
G.711 headers are 1/4 size of payload

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11
Q

DSP

A

Digital Signal Processing

app specific integrated circuit (ASIC)
voice end points 
echo cancellation & jitter removal 
audio conferencing 
transcoding between codecs 
installed on ISR
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12
Q

IP Phone Startup Process

A
  1. ) PoE from switch
  2. ) Leans/provides VLAN info to phone (CDP/LLDP)
  3. ) Requests IP from DHCP server
  4. ) Receives DHCP IP config
    i) IP address of TFTP server
    ii) where to download cnf.xml files
  5. ) Download device config file
  6. ) Device registers with Gatekeeper/Call Agent
  7. ) Gatekeeper responds with 200 OK
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13
Q

FXO

A 
Foreign Exchange Office 
**used by end-device side of FXS->FXO connex**
supervisory signal type 
answers defined ring number 
relay caller-id information
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14
Q

FXS

A 
Foreign Exchange Station 
**PSTN/PBX side of FXS->FXO connex**
dial tone and power 
supervisory signal type 
process address signaling 
CPTone & Station-ID
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15
Q

What is Glare and how do you fix it?

A

Glare: when an incoming call happens at the sametime as outgoing line is requested and become connected

Ground Signaling fixes with ‘signal groundstart’

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16
Q

How many unique SEP.cnf.xml config files are shared by TFTP on router?

A

2

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17
Q

Voice Hunt Group - Parallel

A

Allows incoming call simultaneously ring all numbers in hunt group

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18
Q

What are Hunt Groups?

A

main phone # (pilot number)
answered by 1-7 extensions in turns
next call depends on hunt group list cmd

19
Q

Signalling Protocols…

A

H.320 (circuit switched)
H.323 (packet switched)
SCCP (skinny)
MGCP

20
Q

What will extend the ‘Trust Boundary’ to include IP phones?

A

CDP/LLDP

21
Q

How do IP phones gather VLAN information and QoS Trust settings?

A

CDP - Cisco Discovery Protocol

LLDP - Link Layer Discovery Protocol

22
Q

S.R.T.P

A

Secure Real Time Protocol

23
Q

POE delivery

A

802.f (standard complaint)
Method A: currently used today (gigabit)
uses data pairs 1-2 / 3-6
Method B: legacy mode (10/100mbps)
uses data pairs 4-5 / 7-9
data separated from power

24
Q

TFTP Server Role

A

TFTP houses config + firmware files
located ON CME or Call Manager
IP phone config NOT done on device

25
Q

CUCM Redundancy

A

2:1 Scheme - cost efficient,degraded service during upgrades
1:1 Scheme - high-availability during upgrades
publishers/subscribers handle mass volume of control (load balancing)

26
Q

What is the Gatekeeper responsible for?

A
  1. ) Call Setup
  2. ) Call Maintenance
  3. ) Call Tear-down
27
Q

What protocol carries the voice data DIRECTLY between IP phones?

A

RTP - Real-time Transport Protocol

28
Q

What will give VOICE VLAN information?

A

CDP - Cisco Discovery Protocol

29
Q

SIP - Session Initiation Protocol Steps

A
  1. ) INVITE - establish media session
  2. ) BYE - terminates session
  3. ) REGISTER - registration of user agent
  4. ) CANCEL - terminate non-established session
  5. ) ACK - acknowledge final response from invite
  6. ) OPTIONS - query user agent or proxy server re: capabilities
30
Q

Linear Quantization

A

signal noise ratio (SNR)
lower SNR for small signals ( worse voice quality)
higher SNR for larger signals (better voice quality)

31
Q

Nyquist Theorem

A

determines minimum sampling rate of analog
requires sampling rate be 2x max FHq
human speech 200-9000Hz
human ear 20-20,000Hz
sampling rate for digitizing was set to 8000

32
Q

Total Bandwidth Calculation Procedure

A

Total Packet Size Total Bandwidth Requirement
————————– = ——————————————–
Payload Size Nominal Bandwidth Requirement

Layer 2 Header
IP + UDP + RTP Headers
Payload Size
Nominal Bandwidth

33
Q

How is Analog Voice converted to Binary?

A
  1. ) Sample analog signal
  2. ) Quantize sample
  3. ) Encode digital signal (1010110)
  4. ) Compress to reduce bandwidth
34
Q

BRI-ISDN

A

only 2 B-channels

64kbps

35
Q

PRI-ISDN

A

up to 23 B-channels
1 D-channel
64kbps

36
Q

What is a PBX

A

Private Branch Exchange

routes calls WITHIN a business exclusively
connect to outside via PSTN
PSTN uses POTS/ISDN

37
Q

PSTN

A

Public Switched Telephone Network

circuit-switched (no reroute mid-call!)
call establishment is allocated for analog signal
runs are kept short!
more repeaters

38
Q

VoIP Dial Peers

A

packet-switched network
transmits ALL digits in the called number by default
match calls to ‘Session Targets’

39
Q

P.OT.S Dial Peers

A

-remove or strip any outbound digits that match destination pattern
match calls to Voice Ports
uses traditional phone network

40
Q

Dial Peers

A

matches when call comes INTO or placed from a CME
dial peers are matched based upon calling or called number
if NO info avail will use -> port number

41
Q

Call Flow Steps

A

matched to an INBOUND dial peer
translation rules or profiles applied from dial peer
outbound dial peer is matched via called number config

42
Q

Call Types

A

Local: does NOT traverse WAN or PSTN
On-Net: between phones on same network/WAN
Off-Net: when you dial access code to use PSTN
Plar: automatically connect 1 phone to another

43
Q

Supervisory Signalling (Loop-Start)

A
  1. ) Endpoint initiates outgoing call by connecting the Ring Lead -> Tip Lead
  2. ) Notify destination phone -> CO rings via ring-voltage
  3. ) Receiver picks up phone Tip/Ring leads connect
44
Q

PCM

A

Pulse Code Modulation
G.711/G.722/G.729 - the main 3!
base for digital telephone 64kbps

VoIP (1 decks)

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