CCNA VOIP Ch 6

FXS

• ds0-group 1 timeslots 1-6 type fxo-loop-start
• (note: ds0-group is for CAS signaling)

• Note:   for CCS signaling, t1, PSTN use
• pri-group timeslots 1-6
• CCS signaling router assumes ISDN PRI signaling.

t1 lines have 24 timeslots.

verify with show voice port summary

9

digit stripping rule for pots.

G.729

a and c

To a translation profile.

• B.
•  
• a b and d are all pots digit manipulation methods

prefix puts the defined number in front of the number.

forward-digits tells the ftf how many (right-justified) digits to send   (note: right justified counts from right to left)

digit-strip is built in and strips any explicitly defined number.  use no digit-strip to negate.

dial peer 0

C.  

you can't control distance between routers.

end devices.  analog phone, modem, fax machine

show voice port summary

ground start

loop start  (default)

any trunk type

• PBX system
• PSTN
• CO

ds0-group 1 timeslots 1-24 type fxo-loop-start

verify:  show voice port summary

pri-group timeslots 1-24

router assumes ISDN pri for signaling.

T1 interface, channel 23 (timeslot 24) is used for signaling.  timeslot 1-24 for North America.

E1, channel 16 (timeslot 17) is used for signaling.  timeslot 1-30 for EU.

POTS, anything traditional, FXS, FXO, E&M, digital voice port.

VOIP,  anything with an IP address.

a call leg represents a connection to or from (or both) a voice gateway from a POTS or VOIP source. 

It's like a routing table for phone numbers.

• config t
• dial-peer voice 1101 pots
• destination-pattern 1101
• port 0/0/0

• dial-peer voice 1102 pots
• destination-pattern 1102
• port 0/0/1

• verify:  sh dial-peer voice summ
• debug: debug voip dialpeer

POTS,  rtr automatically strips any explicitly defined digit from a POTS dial peer before forwarding the call.

21

negate with no digit strip

• config t
• dial-peer voice 2000 voip
• destination-pattern 2...
• session target ipv4:10.1.1.2
• codec g711ulaw

note: codecs must match on both ends.

default codec for voip dial peers is g729

matches 0-9 and *  ex. 20..  matches 2000 thru 2099

• matches one or more instance of preceding digit
• ex. 5+23  matches 5523  55523  555523 etc.

matches range of digits

^ means does not match

• ex. [1-3]22 matches 122 222 322
• ex. [^1-3]22 matches 022 422 522 622 722 822 922 and *22
•  

matches any number of dialed digits  0-32 in length

problem is that after you dial your string, it is still waiting for the next digit for 10 seconds  (t302 timer)

inserts a 1 second pause between dialed digits

7 digit dialing area

10 digit dialing area

11 digit long-distance dialing

service numbers

international dialing

• conf t
• voice-port 0/0/0
• connection plar 1102

• as soon as user lifts handset, ext 1102 is dialed.
• Private Line Automatic Ringdown

Dialed Number Information Service    number dialed.  (used on digital calls, incoming, not used with PSTN)

Automatic Number Identification   number sending the call.  (sent on analog calls, incoming.  Not useful to alert rtr where call should go)

You can setup PLAR to route all inbound calls to the receptionist.

• conf t
• voice-port 2/0/0
• connection plar 1500
• exit
• voice-port 2/0/1
• connection plar 1500
• exit

Most specific destination pattern always wins.

When a match is found, the rtr immediately processes the call.

• sh dialplan number **num**
• rtr lists more specific match first.

• 1. match DNIS using incoming called-number config
• 2. match ANI using answer-address config
• 3. match caller ID info using ANI using destination-pattern config
• 4. match incoming POTS using port dial-peer config
• 5. use dial peer 0

• 1. DC            DNIS   Called-num 
• 2. AA            ANI     Answer-address
• 3. AD            ANI     Dest-pattern
• 4. PP             POTS   Port dial-peer
• 5. Dial peer 0

• any voice codec
• no DTMF relay
• IP Prec 0 (no QOS)
• VAD enabled
• no RSVP  (QOS)
• Fax-rate voice
• no application support (no IVR)
• no DID support

• Mode: prefix digits   POTS dial peer, 
• adds digits to the front of a number

• mode: POTS dial peer,
• specifies number of digits to forward, (right justified)

mode:  POTS dial peer,  enables/disables digit strip.

POTS strips any explicitly defined digits.

• mode:  global,   transforms any defined number into another number.
• ex.  num-exp 4... 5...
• ex.  num-exp 0 5000

• mode: global ----  for both  POTS or VOIP dial peer,  
• you can configure up to 15 rules, you set it up in global mode, and apply it to either POTS or VOIP dial peer.

Like an access list.  

Use Preference commands. 

preference 0 is primary path.

preerence 1 is secondary path.

Rule 1:  If there is no outgonig COR list applied, the call is always routed.

• Rule 2:  If there is no incoming COR list applied, the call is always router.
•  

Lack of bandwidth

• Delay: 
• Fixed delay 
• Variable delay 
• Jitter (delay variation)

Packet loss  

B. 150 ms or less

b. 30 ms or less

c. 1% or less

Best effort - default on every network.  First come first serve.

Integrated Services (IntServ) - uses reservations, not scalable, uses RSVP, (Resource Reservation Protocol)

Differentiated Services (DiffServ) - you define what network traffic goes into classes, each class gets a piece of the bandwidth.

Classification and Marking - usually ACL based and marked at L2 or L3.

Congestion Management - All QOS queuing strategies fall into this category. RUles are set up that kick in when congestion occurs.

Congestion Avoidance - This basically drops enough non essential packets so congestion does not occur in the first place.

Policing and Shaping - Policing limits a defined type of traffic to a certain bandwidth number and drops the excess. Shaping queues excess traffic and tries to send it later.

Link Efficiency - Compression



Note for CCNA Voice, only two categories are covered.  Link Efficiency and queuing algorithms.

Payload Compression - compresses application data.

Header Compression - for small payload packets, but bunches of them.  Header traffic becomes significant. cRTP (compressed real time protocol) can take 40 byte header down to 2-4 bytes.

Link Fragmentation and Interleaving (LFI) - addresses serialization delays by chopping large packets into smaller ones. 

Weighted Fair Queuing (WFQ) - (default) - Balances available BW among all senders.

Class-Based Weighted Fair Queuing (CBWFQ) - Allows you to specify what traffic gets what % of the BW.

Low Latency Queuing (LLQ) - also referred to as PQ-CBWFQ (Priority Queuing).  same as CBWFQ with PQ added. This lets the HI priority (voice) traffic to not only get guaranteed BW but first BW. 

• Classification
• Marking
• Policing

• Marking
• Policing
• Shaping
• Congestion Management
• Congestion Avoidance
• Compression
• Fragmentation and Interleaving

auto qos voip - RTR or L3sw - no trust, uses NBAR (Network Based Application Recognition) to id traffic.

auto qos voip trust - rtr or L3sw - device explicitly trusts markings, does not rely on CDP to verify IP phone is at other end.

auto qos voip cisco-phone - sw - trusts markings from IP phone if verified by CDP.

• auto qos voip cisco-softphone - sw - trusts markings only if CDP can verify softphone.
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