Exam

Well,

failed by 1 questions.
I guess I'll take again in a few months.

Questions were very poorly worded but what can you do.

Network management

Alright last big chapter.

Network management is done at the last two stages.
Operate,   then based on the data accumulated the network will be
optimized...


Statistics will be grabbed on
link utilization -  to see if the link needs upgrading
CPU utilization -  to see if the device will need upgrading
interface utilization - to see traffic patterns

Cisco Works can centralize configuration changes.

FCAPS is the acronym for this chapter.
Fault management
Configuration management
Accounting Management
Performance management
Security management

This is all done using.
NMS - Network management Systems.  This is not a product but a framework for Network management.
In order to run this management the NMS will need.

Network management protocols and standards.  - These will be SNMP, RMON,

The NMS will manage devices  -  these will be the     "managed devices"

Inside each device that is being managed there will be a diplomat a "Management Agent".
Snmp Agent
RMON Agent








Netflow can be used instead of RMON.
Syslog can accumulate data from all the devices instead of storing the logs/data locally on each.



SNMP RFC1157
Runs over UDP.

The data in each device is stored in it.
The storage is arranged in a  TRee format.
MIB.
The MIB can be queried for the data in the cell/branch by referencing its locations
either by name  or number.

Name
Syntax   - interger or string
Encoding

Normal Tree would give you basic data like the interface packets.
Cisco has some "private" MIB which will give you the small, med , larg packets.


SNMPv1
Request and Respond mainly.
Get request -        let's see get CPU usage              -  (get response 60%)

Get Request    -     let's see get route table                  -(get response 10.0.0.0/24 next-hop 15.15.15.1)
Get Next  Request                                                    -  get response 20.0.0.0/24 next-hop 16.16.16.1) 

when the agent   responds    to the   requests    he will send a      Get response 

okay so far we have been reading the fields
Set Request   -      will enable you to write to a field.     set request mib3.3.3.4.5  contactname Saar
pretty useless, I can't find any real implementations of this.

Trap -   this is a setting on the agent.  When a certain item on the device happens it will try to send an
alert about it to the NMS.
For example on linkdown  send   to NMS.



SNMPv2
added getbulk       that way you don't have to repeat the getnext requests
added inform request        basically an improved trap  with Confirmation.


SNMPv3
This one finally adds  Security.
noAuthnoPriv     no authentication at all    and   no privacy  (which means no encryption)
authNoPriv         ok, authentication is ok    but no privacy. (no encryption)
AuthPriv         authentication and Privacy.
Great!!!!

Authentication is MDAC
Encryption is   DES, 3DES, AES


RMON
Network Nodes are needed for this.
Can't avoid it.
They are expensive. I think the leader in this is  NetScout.
Looks at MAC Layer 2 data.

RMONv1  grabs data from the Layer 1-2
RMONv2  grabs it from the layer 3-7

Netflow is a higher level of the above.
It grabs.
Accounting is the data grabbing
Collectors  will grab the data physically.
Analazyers will give you the reporting and GUI for this.

The netflows can be used for billing.
Network planning
Planning for user actions
or Application actions.


CDP
this is a Cisco protocol.
Cisco Discovery Protoocl.
It is helpful for troubleshooting
It runs on Layer 2 level.

syslog allows you to get information from multiple sources.
You can accumulate it all on the Syslog device and then use that to get data.
The levels go from the lowest
Which is the most critical.
0  Emergency
to 6 infomrational
7 debug

Voice and Video

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.

IPv4

IPv4

Version
0100  = 4  so IPv4
0110 = 6  so IPv6

IHL  internet header length.
How long is it in Bytes   (IPv4 changes size) (IPv6 is fixed)

TOS  Type of service   which is your QoS  DSCP coloring marking.

Total Length  of the packet including the data. Useful for determining if you need to fragment the packet

Identification  -  identifies the Fragments.

Flags
0  Fragment
1  do not fragment

Fragment offset    this is fragment 1 of 40

TTL   time to live    each hop cuts 1

protocol      8 bits   used by IANA
1  ICMP    (ping)
2  IGMP  (multicast)
6    TCP
17    UDP
50   ESP
51   AH
88   EIGRp
89   OSPF
103  PIM   (multicast)
112   VRRP

Header Checksum   - used to see if the packet is still ok after transport. Changes every header change.

Source address 32 bits
Destination address 32 bits

IP options   not in use.  Used for security , route record and similar.

Padding   so the packet ends on a 32 bit  boundary.



TOS
used for the QoS.

Voip is 101
nothing is 000
PIFFCIN    priority imediate flash flash Critical in network

TOS itself is 4 bits.
It can be used to select a route based on.

Money
reliable
throughput
delay

Assured Forwarding (AF) Behavior Group

	Class 1 (lowest)	Class 2	Class 3	Class 4 (highest)
Low Drop	AF11 (DSCP 10)	AF21 (DSCP 18)	AF31 (DSCP 26)	AF41 (DSCP 34)
Med Drop	AF12 (DSCP 12)	AF22 (DSCP 20)	AF32 (DSCP 28)	AF42 (DSCP 36)
High Drop	AF13 (DSCP 14)	AF23 (DSCP 22)	AF33 (DSCP 30)	AF43 (DSCP 38)

With DSCP they dropped the TOS which nobody used.
The way it works now is Class 4 will have a higher priority.
and if there is congestion the high Drop will be dropped first.


Precedence               AF  1           AF 2       AF 3         AF  4
Low drop precedence 001010 010010 011010 100010
Medium drop precedence 001100 010100 011100 100100
High drop precedence 001110 010110 011110 100110

So a 1 at the beginning is better.  1xx
A 010 will not be dropped.

MTU ethernet 1518
LAN jumbo frames
TCp will retransmit
UDP wont'

Class A 0xxxxxxx   so 0 to 127
Class B  1xxxxxxx   so 128 to 191
Class C  11xxxxx    so 192 223
class D   111xxxxx  so 224 239        multicast
Class E   1111xxxx   so  240 to 255   experimental.

Unicast
Broadcast
Multicast

Private are not routed  10/8  172.16/12     192.168/16
1 class A    16 Class B    256  Class C

You can subnet the addresses above.

Static nat is ONE to ONE   private to Public.

Dynamic  NAT   overloading    is PAT  port Address translation.
Dynamic Overloading    is an internal pool  to an external one.

Inside Local  is the IP of my PC.
Inside Global  is     the Public IP   I get on the web

Outside Global   is the IP of a device on the WWW.
Outside  Local    is    his IP when he is in my STUB/LAN



BOOTP
get IP and gateway  using UDP  replaced

DHCP
Manual is to map a MAC to an IP address.
Automatic    does not expire
Dynamic is from a Pool and expires.

DHCPDiscover.
Router can relay this
DHCP Offer
DHCP request
DHCP acknowledge

DHCPNAK  not acknowledge I am out of addresses.

DHCP should be in the server farm  / datacenter

Internal DNS campus
Edge External
remote datacenter   BOTH



ARP

IPV6

128 Bits instead of 32 bits per address.
Each IP is globally unique
Header is fixed at 40 Bytes
Header will reference options   so it is a fixed size.
Addresses can autoconfigure if required.
IPSEC is built in
MTU discovery
Multiple IPv6 addresses
No broadcast replaced by multicast

Version 0110   IPV6  = 6
Traffic class  8 bits   = TOS
Flow 20 bits    for ordering the flow.
Next Header   to add more
Hop limit  = TTL
source
destination

6
17 udp
50 esp
51 ah
88 eigrp
89 opsf

ipv4 compatible  000000000x.x.x.x

FF multicast
FE  link local
FC  private addressing

Global is routable
64 bits netowrk    64  bits host  (made up from the MAC 48 bits)

To convert a MAC 48 to 64  you add  two  FF FF   in the middle.


FE  is link local can be auto configured

FC  is private addressing    Unique

Globally aggregetable  = aggregate of the IPV6

Anycast is to the nearest.

FF:01 1  all nodes
FF: 01 2  all routers
FF:02  5  OSPF
FF:02 6  OSPF designated
FF:02  9   RIPnG
FF:02 A   EIGRP
FF:02 C   DHCP

ICMPv6 discovers MTu

IPv6  ND  neighbor discovery

IPv6 DNS   AAAAAAAAAAAA
Use the same DNS server.

Stateless link local
Stateless global
Stateful DHCP


Global
Talk to router and gets the prefix
Prefix  +  MAC = address

EIGRP  for  IPv6
RIPnG
OSPFv3
BGP4
ISIS for   IPv6

Dual Stack is both IPv4 and IPv6
Tunneling  IPv6  into an IPv4 tunnel
Translate   IPv6 to IPv4

Dual Stack  - if DNS sends AAAAA it uses IPv6

Automatic Tunnel
IPv4 compatible
6 to 4   the destination has an IPV4 in it    which is used as the tunnel envelope
6 over 4   Multicast over Multicast

ISATAP - Greek, Chinese

Daul stack can use PAT  or NAT-PT

Ciscio 6PE over MPLS

Service Block service the translations.