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VoIP > MIDTERM Weeks-1-4 > Flashcards

MIDTERM Weeks-1-4 Flashcards

(44 cards)

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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?

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A

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

19
Q

Signalling Protocols…

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A

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

20
Q

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

21
Q

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

Study These Flashcards
A

CDP - Cisco Discovery Protocol

LLDP - Link Layer Discovery Protocol

22
Q

S.R.T.P

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A

Secure Real Time Protocol

23
Q

POE delivery

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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

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A

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

25
CUCM Redundancy
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
What is the Gatekeeper responsible for?
1. ) Call Setup 2. ) Call Maintenance 3. ) Call Tear-down
27
What protocol carries the voice data DIRECTLY between IP phones?
RTP - Real-time Transport Protocol
28
What will give VOICE VLAN information?
CDP - Cisco Discovery Protocol
29
SIP - Session Initiation Protocol Steps
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
Linear Quantization
signal noise ratio (SNR) lower SNR for small signals ( worse voice quality) higher SNR for larger signals (better voice quality)
31
Nyquist Theorem
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
Total Bandwidth Calculation Procedure
Total Packet Size Total Bandwidth Requirement -------------------------- = -------------------------------------------- Payload Size Nominal Bandwidth Requirement Layer 2 Header IP + UDP + RTP Headers Payload Size Nominal Bandwidth
33
How is Analog Voice converted to Binary?
1. ) Sample analog signal 2. ) Quantize sample 3. ) Encode digital signal (1010110) 4. ) Compress to reduce bandwidth
34
BRI-ISDN
only 2 B-channels | 64kbps
35
PRI-ISDN
up to 23 B-channels 1 D-channel 64kbps
36
What is a PBX
Private Branch Exchange routes calls WITHIN a business exclusively connect to outside via PSTN PSTN uses POTS/ISDN
37
PSTN
Public Switched Telephone Network circuit-switched (no reroute mid-call!) call establishment is allocated for analog signal runs are kept short! more repeaters
38
VoIP Dial Peers
packet-switched network transmits ALL digits in the called number by default match calls to 'Session Targets'
39
P.OT.S Dial Peers
-remove or strip any outbound digits that match destination pattern match calls to Voice Ports uses traditional phone network
40
Dial Peers
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
Call Flow Steps
matched to an INBOUND dial peer translation rules or profiles applied from dial peer outbound dial peer is matched via called number config
42
Call Types
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
Supervisory Signalling (Loop-Start)
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
PCM
Pulse Code Modulation G.711/G.722/G.729 - the main 3! base for digital telephone 64kbps

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