I passed the IIUC yesterday, so now I’m a CCNA Voice.  It’s kind of belittling to get a CCNA-level certification at this point in my career, but I didn’t want to be completely left behind, so I figured I should move into some voice stuff before I’m left in the dust.

The exam was probably the best Cisco exam I’ve ever taken.  Of all the exams I’ve taken in the last few years, this is the only one that didn’t have questions with huge misspellings or grammatical errors.  I was really taken aback at that since a good portion of the questions from some of my recent CCNP exams were plain unreadable.  I think I remember leaving a comment on one IIUC question about the word “an” being left out of a description, but that was no big deal.  I’m not that obesessive-compulsive.

As always, feel free to correct anything that needs correcting or add anything that needs adding.  There is a lot more to the full definition of wildcards, but these are the basics.  Note to *nix guys:  This isn’t regex as you understand it.  Yes, the use of curly braces would be nice, but we don’t get that here.

T:  Represents anywhere from 0 to 32 digits

destination-patter 9T  <- matches a 9 followed by 0 - 32 other digits

I don’t usually cover news from Cisco, but they’ve changed some certification stuff around again, and I thought I would bring it up.  This time they’ve changed the CCNA Voice, CCVP, and CCSP, so, if you’ve on those tracks, be careful what you’re studying!

CCNA Voice

Circle 28 February 2011 on your calendars.  That’s when the CCNA Voice track gets a shakeup.  The IIUC (640-460) exam will be no more, and passing CVOICE (642-436) will no longer be a valid way to get the cert.  After the big day, you’ll have to take ICOMM (640-461).  This seems to be a much broader exam instead of having the enterprise and commercial focuses in CVOICE and IIUC, respectively.  Look out for both CME- and CUCM-based topics including a troubleshooting section. 

I tried something like this earlier this year with STP.  It got rave reviews (from my mother), so I figured I try it again.  

Below is a list of requirements for configuring a router as a call processor.  In a lab or in your head, configure the router to support the features as listed.  This isn’t a contest or anything like that.  If you get it right, a virtual thumbs up is all I can afford to give you.  There are some licensing issues for running this stuff in GNS3/dynamips, so I can’t help you out on that.  I’ll just hint that GNS3 and dynamips will bind to real networks and that copies of a compatible IP softphone are available.

More of my IIUC study notes.  As always, feel free to correct.  I really need to have a real post, don’t I?

show voice port summary

• Shows the voice ports available for use

R1#show voice port summary
                                          IN       OUT
PORT           CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
============== == ============ ===== ==== ======== ======== ==
50/0/1         1      efxs     up    up   on-hook  idle     y
50/0/1         2      efxs     up    up   on-hook  idle     y
50/0/2         1      efxs     up    up   on-hook  idle     y
50/0/2         2      efxs     up    up   on-hook  idle     y
50/0/3         1      efxs     up    up   on-hook  idle     y
50/0/4         1      efxs     up    up   on-hook  idle     y
50/0/5         1      efxs     up    up   on-hook  idle     y

• An ephone-dn shows up as efxs, so all these are ephone-dns.
• Channels are numbered 0-23; timeslots are numbered 1-24

FXS Ports

Here are some more notes from my IIUC studies.  As always, corrections requested.

Paging

• Broadcasts messages to a group for a one-way communication
• Paging groups are used to limit which phones get the broadcast
• Paging can be unicast or multicast
  • Unicast groups limited to 10 members
  • Multicast requires mcast support on the network
• Paging configurations can be unicast, multicast, or multiple-group

!  Unicast Paging
!  When 1044 is dialed, ephone 1 is paged
R1(config)#ephone-dn 44
R1(config-ephone-dn)#number 1044
R1(config-ephone-dn)#paging
R1(config-ephone-dn)#exit
R1(config)#ephone 1
R1(config-ephone)#paging-dn 44

Here are some more notes from my IIUC studies.  As always, corrections requested.

Local Directory

• Allows users to look up names
• Allows names to show up when dialing or receiving a call
• Most phones have a directory button; some have a menu options for the directory

R1(config)#ephone-dn 1
R1(config-ephone-dn)#name Roger Smith

• Directory entries can be added manually

R1(config-telephony)#directory entry 1 1700 Corporate Fax
R1(config-telephony)#directory entry 2 1701 HR Fax

• By default, sorting is done alphabetically by first name.
• Sorting can be changed

R1(config-telephony)#directory last-name-first

More study notes.  Correct if wrong, though I hope I get some of it right since I already since I’m an R&S guy.  :$

**Switchport Configuration
**

• switchport mode access:  This config makes the port an access port that carries the primary and voice VLAN traffic
• switchport mode trunk:  This config akes the port a trunk unconditionally, but it will still send DTP messages
• switchport nonegotiate:  This config keeps the port from sending DTP messages.
• switchport mode dynamic auto:  If the port receives DTP messages, it will become a trunk.  If not, it will be an access port.
• switchport mode dynamic desirable:  The port actively sends DTP messages trying to become a trunk.  This is the default configuration on a Cisco switch.

Cisco IP Phone Boot Process

These are some of my notes on my IIUC studies.  Since I am a novice as voice stuff, please let me know what I get wrong.

An ephone is a representation of a phone.  It’s basically a structure of features that a phone will have. 

Configuration in CME:

R1(config)#ephone 34  <– This is just a tag and has nothing to do with an extension or phone
R1(config-ephone)#mac-address 1111.2222.3333    <– Assigns this ephone to the phone with that MAC address

Feel free to correct anything that is wrong or incomplete.

• Power over Ethernet (PoE)

  • Can provide power to a Cisco phone, access point, security camera, etc., through the network cabling, eliminating the need to plug the phone into the wall for power.
  • Generic term for providing power on the Ethernet cable
  • Provides centralized power that can be put on a UPS
  • Allows devices to be located away from power outlets
  • Removes cabling clutter at the user’s desk
  • Can be provided through PoE-enabled switches, power panels or inline couplers (power injectors)
  • Oversubscription is common
    • If every device on a switch asks for full power, the switch may not be able to handle the load.
  • Of course, devices can be powered with a power brick at the desk

• 802.3af

Feel free to correct.  No need to sugar-coat it; I’m pretty new at this stuff.  :)

• Advantages of VoIP

  • Reduces costs of communications:  Eliminates/reduces long distance and international call tolls
  • Reduces costs of cabling:  No need for second network of phone lines
  • Integrates all voice into one large network:  All your remote offices can be implemented/maintained/controlled centrally
  • Provides mobility:  Moves, adds, and changes (MACs) are (nearly) eliminated since your phone is just a network node
  • Allows use of IP Softphones
  • Unifies emails, voice mails, and faxes:  All these can be treated as a single box for user messages
  • Increases productivity:  Ringing multiple devices at the same time eliminates phone tag.   <— pushing it, eh?
  • Enhances communications:  Applications can be launched/updated from a voice call through application servers
  • Provides open, compatible standards:  You can connect different vendor devices into the same VoIP network.   <— I’ve never seen that happen

• Cisco VoIP Structure

These are the notes I’ve taken as I read through the study materials.  Feel free to correct anything you see.

• Analog phone signaling

  • Misc
    • Ground = positive = tip
    • Battery = negative = ring
    • Signaling uses specific frequencies for specific events
  • Loop start signaling
    • When a circuit in the phone is completed (i.e., you take it off-hook), the CO detects it and provides services.
    • Susceptible to glare, where the phone requests dialtone at the same time that the CO sends a call.
      • Can connect two different calls if in a business with multiple lines
  • Ground start signaling
    • The circuit is temporarily completed to signal the CO for services
    • Doesn’t connect any call to any phone directly
    • Used in PBXes.
  • Supervisory signaling
    • On-hook:  Circuit is open
    • Off-hook:  Circuit is completed
    • Ringing:  AC current generated by CO to tell the phone to ring
  • Informational signaling
    • Gives information for the caller to use
    • Dial tone
    • Busy
    • Ringback: the ring you hear when you call
    • Confirmation:  the call is being attempted
    • Congestion:  no lines available to make the call
    • Receiver off-hook
    • Reorder:  can’t make the call
    • No such number:  can’t find the endpoint
  • Address signaling
    • Used to send digits
    • Dual-tone multifrequency (DTMF):  uses two electrical signals to indicate a digit; touch tone
    • Pulse:  flashes the circuit to indicate a digit; rotary dial
  • Disadvantages of analog signaling
    • Attenuation
    • Repeaters can’t differentiate between call and noise
    • One cable pair for each call; think about a pair for each call taking place in Manhattan right now

• Digitizing voice

Here are some more notes from my studies.  Of course, no one cares about them but me, but it’s my blog.  I’m sure someone will find it useful.  Please help to correct dumbass mistakes.

• Congestion

  • Speed mismatch - traffic leaves a lower-bandwidth interface than the one it came in on
  • Aggregation problem - lots of links with one egress of equal bandwidth
  • Confluence problem - a bunch of traffic needs to egress out of the same interface

• Queuing

Here’s another set of notes from my ONT studies.  I’m sure someone will find it useful.  Please help to correct dumbass mistakes.

• Classification is done with traffic desriptors

  • Ingress interface
  • CoS value on ISL or 802.1P frames
  • Source/destination IP address
  • IP Precedence or DSCP value
  • MPLS EXP
  • Application type

• Layer 3 QoS

  • Type of Service (ToS) is 8-bit field.
  • First 3 bits of ToS are the IP precedence.
  • First 6 bits of ToS are the DSCP value.
  • Last 2 bits of ToS are explicit congestion notification (ECN).

• Layer 2 QoS

I’ll try to keep it a little shorter this time.

Major issues for converged enterprise networks

• Available bandwidth: competition among applications
  • Fixes
    • Increase bandwidth: More power!
    • Properly queue based on classification and marking: QoS
    • Compress: cRTP, TCP header compression, etc.
• Delay: Lead time to get a packet to the destination
  • Types of delay
    • Processing delay: routing, switch delay
    • Queuing delay: how long a frame stays in an output queue
    • Serialization delay:  how long to put the frame on the wire
    • Propagation delay: the time to cross the physical medium
• Jitter (delay variation): Variation is the delay
  • Different delays mean different arrival times
  • De-jitter buffers save up packets to reduce jitter (like the old CD writers)
  • Fixes
    • More bandwidth
    • Prioritize sensitive data and forward first
    • Remark (reclassify) packets based on sensitivity
    • Enable L2 payload compression: make sure compression delay isn’t worse than the jitter
    • Use header compression
• Packet loss: Packets are lost in the network somewhere
  • Fixes
    • More bandwidth
    • Increase buffers space: more room for the queue on the interface
    • Provide guaranteed bandwidth: Queuing and QoS
    • Congestion avoidance
      • Random Early Detection (RED) and weighted RED (WRED) drop packets before the queue is full
      • Selective dropping is better than FIFO or LIFO dropping

QoS History

Here are some of the notes I’ve been taking while reading over the ONT book. I hope it benefits somebody.  Feel free to correct any stupid mistakes as a paraphrase to avoid a lawsuit.

There’s way too much info here.  I’ll refine the process a little better for the next topics.

Benefits of Packet Telephony Networks

• More efficient use of bandwidth and equipment - Packet telephony networks don’t dedicate channels or a static bandwidth to a call; it’s just another network application.
• Consolidate network expense - The common infrastructure (IP-based networks) keeps you from having to support another distinct network for voice like in traditional PBX implementations.
• Improved employee productivity - The phone can be used for more than just phone calls by utilizing the XML interface to run applications or provide content from the network.
• Access to new communications devices - IP phones can communicate with computers, network gear, PDAs, etc., and not just the PBX.

Packet Telephony Components