MIDTERM Weeks-1-4 Flashcards
ISDN
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
DTMF
Dual-Tone-Multi-Frequency
touch tone (1963)
generates frequencies [1209,1336,1477]
45m/s quickest time between dials
3sec longest duration between dials
What fields are NOT included in UDP?
Sequence Numbers
Time Stamps
P.O.T.S
Plain Old Telephone Service
voice grade
analog over copper wires
C.A.S
Channel Associated Signalling
Extended Superframe (ESF)
- TI framing standard
BiPolar 8-zero Substitution (B8ZS)
- Encoding method on T1 curcuits
Which command will display ALL the TFTP files created by CME?
show voice register tftp bind
What is the maximum number of configurable Directory Numbers (DN)?
400
What is the maximum number of configurable Pools?
110
Digital Voice Encoding
Each sample encoded with 8 bits
1 bit = POLARITY
3 bits = SEGMENT BITS
4 bits = STEP BITS
V.E.O
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
DSP
Digital Signal Processing
app specific integrated circuit (ASIC) voice end points echo cancellation & jitter removal audio conferencing transcoding between codecs installed on ISR
IP Phone Startup Process
- ) PoE from switch
- ) Leans/provides VLAN info to phone (CDP/LLDP)
- ) Requests IP from DHCP server
- ) Receives DHCP IP config
i) IP address of TFTP server
ii) where to download cnf.xml files - ) Download device config file
- ) Device registers with Gatekeeper/Call Agent
- ) Gatekeeper responds with 200 OK
FXO
Foreign Exchange Office **used by end-device side of FXS->FXO connex** supervisory signal type answers defined ring number relay caller-id information
FXS
Foreign Exchange Station **PSTN/PBX side of FXS->FXO connex** dial tone and power supervisory signal type process address signaling CPTone & Station-ID
What is Glare and how do you fix it?
Glare: when an incoming call happens at the sametime as outgoing line is requested and become connected
Ground Signaling fixes with ‘signal groundstart’
How many unique SEP.cnf.xml config files are shared by TFTP on router?
2
Voice Hunt Group - Parallel
Allows incoming call simultaneously ring all numbers in hunt group
What are Hunt Groups?
main phone # (pilot number)
answered by 1-7 extensions in turns
next call depends on hunt group list cmd
Signalling Protocols…
H.320 (circuit switched)
H.323 (packet switched)
SCCP (skinny)
MGCP
What will extend the ‘Trust Boundary’ to include IP phones?
CDP/LLDP
How do IP phones gather VLAN information and QoS Trust settings?
CDP - Cisco Discovery Protocol
LLDP - Link Layer Discovery Protocol
S.R.T.P
Secure Real Time Protocol
POE delivery
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
TFTP Server Role
TFTP houses config + firmware files
located ON CME or Call Manager
IP phone config NOT done on device
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)
What is the Gatekeeper responsible for?
- ) Call Setup
- ) Call Maintenance
- ) Call Tear-down
What protocol carries the voice data DIRECTLY between IP phones?
RTP - Real-time Transport Protocol
What will give VOICE VLAN information?
CDP - Cisco Discovery Protocol
SIP - Session Initiation Protocol Steps
- ) INVITE - establish media session
- ) BYE - terminates session
- ) REGISTER - registration of user agent
- ) CANCEL - terminate non-established session
- ) ACK - acknowledge final response from invite
- ) OPTIONS - query user agent or proxy server re: capabilities
Linear Quantization
signal noise ratio (SNR)
lower SNR for small signals ( worse voice quality)
higher SNR for larger signals (better voice quality)
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
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
How is Analog Voice converted to Binary?
- ) Sample analog signal
- ) Quantize sample
- ) Encode digital signal (1010110)
- ) Compress to reduce bandwidth
BRI-ISDN
only 2 B-channels
64kbps
PRI-ISDN
up to 23 B-channels
1 D-channel
64kbps
What is a PBX
Private Branch Exchange
routes calls WITHIN a business exclusively
connect to outside via PSTN
PSTN uses POTS/ISDN
PSTN
Public Switched Telephone Network
circuit-switched (no reroute mid-call!)
call establishment is allocated for analog signal
runs are kept short!
more repeaters
VoIP Dial Peers
packet-switched network
transmits ALL digits in the called number by default
match calls to ‘Session Targets’
P.OT.S Dial Peers
-remove or strip any outbound digits that match destination pattern
match calls to Voice Ports
uses traditional phone network
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
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
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
Supervisory Signalling (Loop-Start)
- ) Endpoint initiates outgoing call by connecting the Ring Lead -> Tip Lead
- ) Notify destination phone -> CO rings via ring-voltage
- ) Receiver picks up phone Tip/Ring leads connect
PCM
Pulse Code Modulation
G.711/G.722/G.729 - the main 3!
base for digital telephone 64kbps
