PSTN is circuit switched. Which means that the circuit is built and used for the entire
connection. There is no switching done while the circuit is alive.
CO central office use SS7 in order to route and build the circuit.
The call can be build on Dialup , ISDN , or a TDM.
Each call is 64 kbps of bandwidth and is called a DS0
DS0=64 Kbps.
Old PBXs sit in the enterprise and will give you.
Extension dialing.
VoiceMail
transfers
conferencing
To connect to another site a company can set up a TIE line .
On the TIE line there are no charges to the enterprise.
However the TIE line itself costs money.
Alright, on net which is on the TIE lines.
Off NET to the PSTN.
This will be the same even in VOIP.
If you are using T1's you will call it On-Net
If you are having a call over the internet it will be off-net
PSTN requires you paying charges per each call.
While a Tie line has a fixed monthly cost.
T1 can carry only 24 calls.
24 * 64 = 1280 256 = 1536 Kbps
Now in most books they say T1 is 1.544 Mbps .
So where are the missing 8 bps.
Apparently those are used by the telco for synchronization.
So 23 B channels + 1 D channel = 24 Channel then you need to add 8 Kbps for synchronization.
In the case of Telco. This is ALL used for calls and cannot be used for Data.
Ok.
CO - central office.
This is a map of all the CO (central office) in the USA.
Notice how the west coast has less per mile.
So I drew this up. Since the CCDA one in the book looks pretty useless.
So Tie line is what I buy so I can connect two offices and not pay toll.
Tandem Trunk is what the PSTN provider uses to connect local CO (local exchanges is the correcter term)
Tandem trunks go to a Tandem Switch (class 4)
They will connect to a Class 3 switch which will connect to another Class 3.
Technically if you want to dial abroad then you need to reach a Class 1 switch.
Anyway from Switch 3 to Switch 3 it is and INTER TOLL trunk.
Co to PBX and PBX to Co is just the connection to the CO from the office.
As a note. When you dial in NY. you only need to dial 7 digits since it uses the TANDEM trunks.
When you dial to boston you need the FULL number. it goes on the TOll trunk.
Okay
FXS
Foreign Exchange Service.
We are Exchanging - ie TALKING
In VOIP you will use the same ports to connect OLD equipment to your VOIP network.
Like the ATA from Cisco gives you two FXS ports for the old devices.
So FXS ports
provide Dial tone
Power
Ring Voltage.
Now
FXO is the port that
Ah fuck it.
Just try this.
FXS point to the STATION.
FXO points to the central Office.
So on the phone you have an FXO port
you plug the cable to the FXS port on the PBX
The PBX has an FXO port
that you will plug a cable from the FXO port
to the FXS jack which is the POTS circuit to the Central OFFICE.
E&M ear and mouth - Earth and Magnet.
This is basically a PORT on a PBX. you run an Analog cable which will run to another PORT on PBX2
This allows you to send a signal. This is a TIE trunk for analog.
This has been replaced by BRI PRI digital.
Since we use T1/E1
T1 has 24 channels. It can work either.
CAS Channel associated Signalling - The signalling here is in each channel
In each channel a bit will be robbed for signalling. So 24 channels.
CCS - common channel signalling - This uses one channel for signalling so 23B+D
ISDN uses this and so does SS7
Signalling the state of the phone.
Supervisory signalling tell if it is on hook or off hook
Addressing sends the digits.
Informational sends you the BUSY
Loop start - residential CO to Phone. When you lift the handset the circuit is closed.
Ground Start - CO to phone signals to the switch that it is about to take the line.
helps prevent glaring which is when both take the line at the same time.
E&M - PBX to PBX Two wire - four wire adds more signalling
CAS T1 occurs in band
CSS T1 sets up a separate channel for the signalling
QSig Q.931 used for ISDN between PBX to PBX and Hybrid to CUCM
SS7 inter PSTN switches signalling used by the PhoneProvider.
Loop Start
the CO has the Power 48 DC.
That is why a phone does not need electricity.
When you lift the handset the circuit is closed (off hook) and the power flows all the way to
the phone and back to the CO which sends a dial tone.
Ground Start.
Uses TIP and RING
The PBX has a TIP detector
When the CO grounds the TIP
the PBX detects this and will ground the RING.
Now the CO power 48DC can flow and the arrival of the 48 DC will signal to the CO to send Dial tone.
If the PBX wants to ring.
It will ground the RING which will be detected by the CO.
The CO will ground the TIP
Now the CO power 48 can flow and when it reaches the CO it send Dial tone
E&M
type I and type II are in the USA
type III is everywhere.
Type V is outside the USA.
Immediate start wait 200ms and send
Wink - wait for on hook from other side and then sends.
Delay start - lift wait for 200ms check if on hook. if still off-hook then wait till on-hook.
on hook means
Off- hook means
SS7 signalling
So
Best I can do STP would be internal.
SCP for control - controls 800-900 and credit cards
Sending the Dialing Numbers
Pulse - rotary - sends 5 signals 1 1 1 1 1 = 5
DTMF - sends Two tones at the same time. X+Y = 5
NANP
North American - numbering plan
country-area-office-line
nxx-nxx-xxx-xxxx
Centrex - the CO has the PBX
VoiceMail -
Database or CDR keeps data on the calls. You can get reports.
IVR - interactive for accounting press 5 for Jamil press 2
ACD - used for call centers - how long the calls + statistics.
Grade of service.
P.02 GOS = means 2% of calls won't make it
P.01 GOS - means 1% won't make it and will be blocked.
Earlang B.
You have 24 channels
if one is taken for a whole hour = 1 Earlang.
You measure it at the busy hours.
30 calls * 10 minute per call = 300 minutes 300/60 = 5 Earlangs.
So you run Earlang with the GOS to calculate the number of lines you need.
Earlang B extended
Same thing but takes into account some people will retry to call.
Earlang C
Same thing but instead of blocking calls will put them on HOLD.
Used for C callcenter.
Call second = 1 second of calls.
Centrum = cent 100 seconds
1 Centrum * 36 = 1 Earlng
Busy Hour also called Peak Hour
Busy Hour traffic.
average call duration (20 minutes) * number of calls (6) = 120 minutes/60 = 2 Earlangs.
If you have the length of the calls in seconds.
Then you need to divide it in seconds. So instead of 60 minutes you use 3600 seconds.
So an example.
each call is 300 seconds * 400 calls = 120,000 seconds / 3600 seconds = 33.333 earlangs.
Eventually you will have blocked calls. They are the GOS you plan for.
CDR
call detail records done by the PBX
ACD distributes calls to agents.
Cisco Unified Network
Network is at the base.
Call routing done by the CUCM / SRST dial plans and PST gateway
Call Control on off to the LDAP
Applications and services like chat or voicemail or contact center
Operations and Service Quality
VoFR - over frame relay
VoATM - over ATM
VOIP is now the leader.
Voice over IP.
PSTN is not flexible
Data networks are
You can also save money by adding it all on the DATA network.
SRST - backup goes to PSTN . Overflow of traffic goes to PSTN.
Cisco IPT IP telephony.
Network is the base.
Gateways convert from VOIP to analog
CUCM is the brains of the dial plan and routers VOIP
Single Site - one CUCM at HQ
Multi site CENTRALIZED - CUCM at HQ and SRST at branches
Multi site Distributed - CUCM at each location .
In a multi-site distributed.
Each Cluster up to 30000 users.
To connect clusters use Inter-cluster trunk
GateKeepers will keep them in sync and enforce CAC
CAC is call access control. Which prevents too many calls on the line. So the call quality is kept up.
Call manager express is one appliance that does the mailbox, pstn and voip
Video
Telepresence which is the Cisco Video conferencing is many to many and takes 4-12 Mbps HD
Desktop Video to Video - many to many less demand
cameras to hq - many to few many cameras send data to few(hq)
Streaming Video Few to Many - the VOD to many users.
Access for users
Transport for sending the data
Bridging for converting it.
session provide signalling.
Storage store content
Codecs
G711 8000 samples per second * 8bits per sample 64000 = 64 Kbps = DS0
Analog to Digital
Filter the range you want to record. anything above 4000hz will be dropped
sample the speaking 8000 times per second
Digitize it into 0 1 01 1 01010 1 also called PCM pulse to Code modulation
G711 U for USA 64 Kbps
G711 a for international 64Kbps
G729 8 kbps
G728 16 Kbps
G726 16-40 Kbps
G723.1 6.3 5.3 Kbps
711 is the best then 729 728 726 723.
If you have the bandwidth keep it at 711
if you want to compress and not lose quality 729
Call control
Q931 for ISDN
H225 for the rest.
This is done over the TCP
UDP
G7xx is the Audio over UDP
H.26x H for Hvideo
or RTP real time Protocol does them both.
RTCP is control of the Video/audio
RAS is control
SCCP Cisco proprietary VOIP call cotrol..... IT only sets up the Control
RTP for voip streaming. this is the actual data call stream.
MGCP media Gateway Control Protocol
The HQ Gateway controls everything.
SIP - voip networks for non-cisco proprietary.
IP = 20 bytes
UDP = 8 bytes
RTP = 12 bytes
cRTP compresses the 40 Bytes to 2-4 Bytes
Hop by Hop for small 768 kbps links
MGCP allows the CUCM to control Gateways that go to PSTN
the CUCM is the Call agent endpoints are the phones.
H.323
Terminals are the clients
MCU mixes streams
Gateway converts to PSTn
Gatekeeper - Dial plan + CAC used for multisite distributed CUCM
Gatekeeper works like a OSPF DR and holds the Dial plan.
SIP proxy manages the SIP clients
VAD supresses silence
Propagation delay is the travel time.
Processing delay is the time to convert it to digital
Serailization is how long to put it on the interface. -
Queuing delay is waiting ebcause of other packets. - LFI and QoS helps
Jitter is the change in the delays - use dejitter buffers
Echo delay of 15ms and above must be cancelled.
Classify MATCH
Mark it with the color THEN
Congestion avoidance by using WRED or DWRED to drop tails.
Traffic Policing
Traffic Shaping by buffering and releasing slowly.
AUTO-QoS
Marks
does 802.1Q
LLQ
CBWFQ for control traffic
P>S.
VOICE sucks.
Wait till you have to use the Cisco tools to sell someone a Callmanager.
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