Aaron's Worthless Words

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

• 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

!  Multicast Paging
!  When 1045 is dialed, ephone 2 is paged
R1(config)#ephone-dn 45
R1(config-ephone-dn)#number 1045
R1(config-ephone-dn)#paging ip 239.1.1.100 port 2000
R1(config-ephone-dn)#exit
R1(config)#ephone 2
R2(config)#paging-dn 45

!  Multiple Group Paging
!  When 1046 is dialed, both ephones 1 and 2 are dialed
R1(config)#ephone-dn 46
R1(config-ephone-dn)#number 1046
R1(config-ephone-dn)#paging group 44, 45

• There is a limit of 10 DNs in the paging group.

After-hours Call Blocking

• Allows you to configure time ranges and patterns that cannot be called during those ranges
• Three steps
  1. Defines days and/or hours that are considered after-hours
  2. Specify patterns to be blocked
  3. Create exemptions

R1(config)#telephony-service
R1(config-telephony)#after-hours day mon 18:00 07:00 <- afterhours = 6pm to 7am
R1(config-telephony)#after-hours day tue 18:00 07:00
R1(config-telephony)#after-hours day wed 18:00 07:00
R1(config-telephony)#after-hours day thu 18:00 07:00
R1(config-telephony)#after-hours day fri 18:00 07:00
…
R1(config-telephony)#after-hours date Dec 25 00:00 00:00 <- Christmas is after hours
…
R1(config-telephony)#after-hours block pattern 1 91900……. 7-24 <- Pattern index 1 blocks 900 numbers 7day/24hours
R1(config-telephony)#after-hours block pattern 2 91………. <- Pattern index 2 block all long distance after hours
…
R1(config-telephony)#login timeout 15 clear 18:00 <- Allows logins for entering a PIN for after-hours exemption; times out in 15 minutes and clears at 18:00
R1(config-telephony)#exit
R1(config)#ephone 1
R1(config-ephone)#after-hours exempt <- the boss's phone can call anywhere except the 7-24 patterns
R1(confg-ephone)#ephone 2
R1(config-ephone)#ping 1234 <- Your phone can log in with this PIN for after-hours access

• Phones have to be restarted or reset for the Login key to be enabled.

Call Accounting

• It's important to see who is calling international numbers every day at lunch.
• Call Detail Records (CDRs) record who called what number when for how long plus more stuff.
• CME logs CDRs to the logging buffer, syslog, or both.
• Logging buffers clear when a router loses power, but it may be better than nothing.  <- Don't do this ever!  Get a syslog server!

R1(config)#logging buffer 512000 <- Set the logging buffer size to 512000 bytes
R1(config)#dial-control-mib retain-timer 120 <- Roll records out in 120 minutes
R1(config)#dial-control-mib max-size 100 <- Only keep last 100 records

• Sending to syslog allows you to keep more records

R1(config)#gw-accounting syslog
R1(config)#logging 192.168.0.2 <- Log to this server

• Account codes are used for billing.
  • Each department or unit can enter a code that appears in the CDR for use later.
• Users press the Acct key when the call is ringing or connected to enter their code.

Music on Hold

• Do I have to explain what MoH is?
• WAV or AU file in flash
• Files must be G.711 or G.729
  • G.711 is recommended since it is of higher quality
• Can be delivered via unicast or multicast

R1(config-telephony)#moh piratedmusic.au <- Plays a local audio file as MoH
R1(config-telephony)#multicast moh 239.1.1.15 port 2001 <- multicast the MoH

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

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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

Call Forwarding

• Can be done by the user or through CLI
• User presses CFwdAll button, enters a number, and #; pressing CFwdAll again cancels forwarding.
• CLI forwarding is more flexible

R1(config-ephone-dn)#call-forward busy 1800
R1(config-ephone-dn)#call-forward noan 1800 timeout 25 <- if no answer after 25 seconds
R1(config-ephone-dn)#call-forward max-length 0 <- disabled forwarding
R1(config-ephone-dn)#call-forward max-length 4 <- restricts forwarded number to a length of 4 digits

• H.450.3: A voice gateway redirects the forward to another gateway instead of using the phone as a proxy
  • Direct path from originator to destination
  • Frees up network resources by keeping path direct
  • Keeps latency and jitter down by avoiding long looping paths and a hairpin turn at the phone
• Forwarding patterns can help restrict where calls can be forwarded

R1(config-telephony)#call-forward pattern 1… <- allows forwarding to a 4-digit number starting with 1

Call Transfer

• H.450.2: A voice gateway redirects transfers to another gateway instead of using the phone as a proxy.
  • The user doing the transfer is dropped from the conversation after transfer is complete.
• Generically, there are two types of forwarding.
  • Blind: sends the caller to the number blindly
  • Consult:  allows you to talk to the endpoint before transferring the call
• CME has three types of forwarding.
  • full-blind:  blind transfers using H.450.2 or SIP REFER
  • full-consult:  consult transfers using H.450.2 or SIP REFER if second line is available; if not, fall back to full-blind
  • local-consult:  Cisco-proprietary method for full-consult

R1(config-telephony)#transfer-system full-consult
- or -
R1(config-ephone-dn)#transfer-mode consult

• Transfer patterns work similarly to forwarding patterns

R1(config-telephony)#transfer-patter 1…

Call Park

• Call parking allows a user to retrieve a call from any phone by "parking" the call to an extension.
• The call can be picked up from any phone able to dial that extension.
• Park numbers can be assigned randomly or manually.

R1(config-ephone-dn)#park-slot <- makes this DN a park slot

• Call parking has several options.
  • reserved-for dn:  Only that DN can use this park-slot
  • timeout seconds:  Ring the phone phone that parked the call after that many seconds to remind them of the park
  • limit count:  After that many timeout intervals, drop the call.  Not good for customers.
  • notify dn [ only ]:  Notify that DN when a timeout is reached
  • recall:  Sends the call back to the original phone when the timeout is reached
  • transfer dn:  Sends the call to this DN when the timeout is reached
  • alternate dn:  If the DN in the transfer command is not available, go here
  • retry seconds:  Try to transfer again after this many seconds
• The phone must be reset for call parking to take effect.

Call Pickup

• Allows users to pick up other ringing phones
• Best to use pickup groups so the sales guys don't pick up support calls by accident

R1(config-ephone-dn)#pickup-group 5000

• There are three methods to pickup a call.
  • Directed pickup:  A user picks up a ringing phone by pressing PickUp followed by the target DN.
  • Local group pickup:  A user picks up a ringing phone in his pickup group by pressing GPickUp then *.
  • Other group pickup:  A user picks up a ringing phone in another pickup group by pressing GPickUp then the other group number.

Intercom

http://www.youtube.com/watch?v=3P2dbwrT_fQ

• Technically is a speed dial and auto-answer combination
• Intercom button is pressed, which dials a DN bound to another phone; that phone automatically answers on mute.
• The DNs involved usually (?) can't be dialed.
  • e.g., A101

R1(config)#ephone-dn 99
R1(config-ephone-dn)#number A99
R1(config-ephone-dn)#intercom A98 label "Boss"
R1(config-ephone-dn)#exit
R1(config)#ephone-dn 98
R1(config-ephone-dn)#number A98
R1(config-ephone-dn)#intercom A99 label "Lackey"
R1(config-ephone-dn)#exit
R1(config)#ephone 54
R1(config-ephone)#button 5:99
R1(config-ephone)#restart
R1(config)#ephone 73
R1(config-ephone)#button 5:98
R1(config-ephone)#restart

• Other options
  • barge-in:  Places existing calls on hold on the other end and barges n
  • no-auto-answer:  Rings instead of auto answers
  • no-mute:  Doesn't mute when auto answering.  Can you say spying?

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

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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

1. Phone connects to an Ethernet switch and gets power if needed
2. Switch tells the phone the correct voice VLAN through CDP
3. Phone sends DHCP request for its voice VLAN
4. DHCP offer includes the TFTP server from which to download the config
5. Phone downloads the config from the TFTP server
6. Phone contacts the call processing server as dictated in the config file

DHCP Settings on a Cisco Router or L3 Switch

R1(config)#ip dhcp pool MYPOOL
R1(dhcp-config)#network 192.168.0.0 255.255.255.0
R1(dhcp-config)#default-router 192.168.0.1
R1(dhcp-config)#dns-server 192.168.0.10
R1(dhcp-config)#option 150 ip 192.168.0.20  <– Tells the phone to download the config from this TFTP server
R1(dhcp-config)#exit
R1(config)#ip dhcp excluded-address 192.168.0.1 192.168.0.100  <– Don't use these IPs when handing out DHCP

NTP

Why should you use NTP for a CME setup?

• Phones display correct time
• Voicemails have the correct time
• CDRs are timestamped accurately
• Router logs are timestamped accurately
• Time-based access worked predictably

R1(config)#ntp server 1.1.1.1
R1(config)#clock timezone MYTZ -5  <– Sets the timezone to a zone called MYTZ that's 5 hours behind UTC

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

More Posts - Website

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

An ephone-dn is a directory number that can be assigned to one or more phone.  This is usually your extension and/or DID number.

Configuration in CME:

R1(config)#ephone-dn 18   <– Again, just a tag
R1(config-ephone-dn)#number 1000  <– the extension

Ephone-dns (i.e., extensions) are assigned to ephones through the button directive under the ephone setup.  You can have more than one assignment per button command.

Configuration in CME:

R1(config)#ephone 34
R1(config-ephone)#button 1:18   <– Assigns extension 1000 (through ephone-dn 18) to button 1

The colon (:) in the button line is a separator that means that this is a normal ring phone – when someone dials that extension, your phone rings and lights up.  There are other separator characters.

Configuration in CME:

R1(config)#ephone 34
R1(config-ephone)#button 3m15  <– Monitors ephone-dn on button 3
R1(config-ephone)#button 4s82  <– Assigns ephone-dn 82 to button 4 but nothing will ring
R1(config-ephone)#button 5f31  <– Assigns ephone-dn 31 to button 5 with a triple ring

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

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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
  • IEEE standard for PoE from 2003
  • Defines power classes so different devices can ask for different power levels
    • Class 0:  15.4W allocated
      • Used for el cheapo devices that just want power
    • Class 1:  4.0W
    • Class 2:  7.0W
    • Class 3:  15.4W
  • Uses all 4 pairs of wire, so works on gig links
  • Power procedure

  1. Small DC current is applied to the line
  2. If an 802.3af device is attached, it runs the current through a resistor
  3. The resistance is detected by the switch which can determine the class of power
  4. Power is applied to the device

• Cisco Inline Power
  • Cisco's version of PoE created in 2000 (before 802.3af)
  • Each device tells the switch what its power needs are
  • Power procedure

  1. PoE device connected to the switch
  2. Switch sends Fast Link Pulse (FLP)
  3. If FLP is received back, 6.3W of power are applied
  4. Device boots off of 6.3W and tells the switch what its real power requirements are via CDP

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

More Posts - Website

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
  • Infrastructure:  Switches, routers, firewalls, etc.
    • QoS!
  • Call processing:  Call signaling, routing, etc.
  • Applications:  Additional functionality like IM support and unified messaging
  • Endpoints:  Phones

• Cisco Call Processing
  • Unified Communications 500 (UC500): Standalone device with switch, router, firewall, voice processing, voice mail all built in
  • Communications Manager Express (CME):  Voice capabilities contained in ISR router
  • Communications Manager Business Edition:  Server solution with most voice capabilities integrated
  • Communications Manager (CM): Full server-cluster solution to support many thousands of phones

• Cisco Applications
  • Interactive Voice Response (IVR):  Those troublesome menus where you say your account number but it never understands you
  • Auto attendant:  Interactive interface where users direct themselves to the correct person/group/team/department by using touch tones.
  • Cisco Unified Contact Center:  Provides IVR, auto attendant, automatic call distribution (ACD), computer telephony integration (CTI), chat/web/email integration
  • Cisco Unity Express:  Linux-based appliance in a router for limited voice mail, IVR, and auto attendant
  • Cisco Unity Connection:  Server-based solution for more robust VM, IVR, and auto attendant
  • Cisco Unity:  Fully-integrated solution running on server clusters

• Phones
  • Entry-level
    • 3911
      • Inline power
      • Fixed buttons
      • Half-duplex speakerphone
    • 7906G/7911G
      • Inline power
      • Onscreen soft keys
      • Basic XML support
      • 7911G has built-in switch
    • 7931G
      • Inline power
      • Onscreen soft keys
      • Basic XML support
      • Built-in switch
  • Business-class
    • 7940G
      • Built-in switch
      • Inline power
      • Broader XML support
      • Onscreen soft keys
      • Full-duplex speaker
      • Headset support
    • 7941G = 7940G + better display with backlight
    • 7941G-GE = 7941G + 10/100/1000 switch
    • 7942G = 7941G + high-fidelity audio and Internet Low Bitrate Codec (ILBC)]
    • 7945G = 7941G-GE + 16-bit color display
    • 794X phones support 2 lines; the same 796X phones support 6 lines.
  • Touchscreen phones
    • 7970G:  7940G + touchscreen
    • 7971G-GE: 7941G-GE + touchscreen
    • 7975G: 7945G + touchscreen + 5″ display
  • Specialty phones
    • 7985G:  Video phone
    • 7921G:  Wireless VoIP phone
    • 7937G:  Conference station
    • ATA 186/188:  Converts analog phones to VoIP
    • Cisco IP Communicator:  Softphone
    • VT Advantage:  Integrates webcam and computer with phone
    • 7914/7915/7916:  Expansion modules for 796X and 797X phones

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

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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
  • Steps
    • Sampling: taking samples of the voice
      • Nyquist method:  sample rate = 2 x highest frequency
      • Human voices usually stay below 4000Hz, so a good sampling rate is 8000 samples/second.
      • Pulse-amplitude modulation (PAM)
    • Quantization:  assigning values to the sample
      • Assignment based on amplitude of the signal
      • Logarithmic scale for better accuracy at the more common amplitudes
    • Encoding:  converting quantization to binary
      • Pulse-code modulation (PCM)
      • 8 bits/sample * 8k samples/second = 64k bpbs
    • Compression:  optionally compress the binary information
  • Advantages
    • Transmitting numbers is less susceptible to attenuation
    • Multiple digital voice signals can use same pair
      • Time division multiplexing (TDM)

• Digital signaling
  • 24 channels * 8 bits/sample = 192 bits of voice
  • The T1 frame sends all 24 channels in one T1 frame with 1 bit for framing bit, so the T1 frame = 193 bits
  • 193 bits/frame * 8k frames/second = 1.544 Mbps
  • Channel associated signaling (CAS):  steals bits in a channel for signaling
    • The 8th bit of every 6th sample is stolen for signaling
    • Super frame (SF) uses 12 frames to synchronize a signal, so 12 samples are required to be received for synchronization (12/8000 second).
    • Extended super frame (ESF) uses 24 frames; 2000 bps for sync, 2000 bps for errors, 4000 bps for control and reporting
  • Common channel signaling (CAC):  uses a dedicated channel for signaling
    • Q.931 is a CAC signaling standard.

• The PSTN
  • Phone companies connect together using SS7 signaling (a CAC method), which is responsible for routing the call.
  • E.164 is an ITU standard for phone numbers.
    • Country code
    • National destination code
    • Subscriber number

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

More Posts - Website

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

• Phones – These include analog phone, digital phones, IP phones, softphones, etc.
• Gateways – These devices connect the different devices that cannot access the IP network.  For example, making a 911 call from your IP phone requires a gateway that switches and converts your VOIP conversation to the PSTN.
• Gatekeepers – These are devices that handle call routing (resolving an IP to an extension/phone number) and call admission control (CAC, grants permission to make the call).
• Multipoint control units (MCUs) – These are conference bridges that connect a bunch of streams together and present it to all participants.  Some can do video as well.
• Call agents – These are devices used in a centralized model that handle the call routing, address translation, call setup, call maintenance, and call termination.
• Application and database servers – These provide required and optional services to the packet telephony network and include TFTP servers for configuration and OS download and XML servers for application use.
• Digital signal processors (DSPs) – These guys converts signals from one form to another.  They convert analog to digital signals, digital to packetized data in the form of a codec, from codec to codec, etc.

Analog Interfaces

• Foreign Exchange Office (FXO) – These are interfaces that expect to connect to a CO or equivalent.  You connect these to your wall jack to get access to the PSTN.
• Foreign Exchange Station (FXS) – You connect your analog devices (phones, modems, faxes, etc.) to these guys to get dial tone.
• Ear and Mouth (E&M) – These are the old-school way to connect PBXes together.

Digital Interfaces

• Basic Rate ISDN (BRI) – These give you 2 64kbps channels (bearer channels) to run voice over.  It also includes a 16kbps D (delta) channel with 48kbps of framing overhead to give you 192kbps.

• T1 (North America) – This is a channelized T1 or a Primary Rate ISDN (PRI).
  • Common Channel Signaling (CCS) – The D channel is dedicated to signaling, giving you 23 64kbps channels.
  • Channel Associated Signaling (CAS)  – There is no D channel, but every bearer channel dedicates a few data bits for its own signaling.

• • E1 (North America) – This is a channelized E1 or a Primary Rate ISDN (PRI).
    • Common Channel Signaling (CCS) – The D channel is dedicated to signaling, giving you 30 64kbps channels.
    • Channel Associated Signaling (CAS)  – There is still a dedicated D channel, so you still have 30 64kbps channels to use.

VOIP Signaling

• H323. – ITU Standard that uses a whole mess of RFCs; distributed model
• Media Gateway Control Protocol (MGCP) – IETF RFC 3435; centralized model
• Session Initiation Protocol (SIP) – IETF standard; distributed model

Phone Call Stages

• Call setup – connects the call between the endpoints
  • Call routing – figures out where the call is going
  • CAC (optional) – Do you have enough resources (i.e., an available channel or bandwidth) to make the call?
  • Call negotiation – negotiates the source and destination IPs, source and destination UDP ports, and codec.
• Call maintenance – collects call statistics for on-demand or historical use
• Call teardown – hanging up and terminating the connection

Digitizing Analog Signals

• Sampling – Periodic capturing and recording of voice resulting in a pulse amplitude modulation (PAM) signal
• Quantization – Assigning numerical values to the PAM signal
• Encoding – Converting the quantization to binary
• Compression (optional) – compressing the binary stream
• Pulse code modulation (PCM) converts analog to digital, but it doesn’t use compression.  It takes 8000 samples per second and converts each sample to an 8-bit number, giving 64kbps of capacity.

Digital to Analog

• Decompression (optional)
• Decoding and filtering – binary is converted back to a PAM signal; filtering removes any noise from the conversion
• Reconstructing the analog signal

The Nyquist Theorem

• The number of samples required to accurately encode (and decode) a signal is twice the highest frequency of the signal.
• Since telephone lines can only transmit up to 3400 Hz (4000 Hz for simplicity), the sample rate should be 8000 samples/second.

Measuring Compression Qualities

• Mean opinion score (MOS) – ITU standard technique for measuring quality of codec; subjective score from 1 to 5
• Perceptual speech quality measurement (PSQM) – Another ITU standard technique for measuring quality of codec; test equipment score from 0. to 6.5
• Perceptual analysis measurement system (PAMS) – Developed by BT; predictive system
• Perceptual evaluation of speech quality (PESQ) – Another ITU standard; combines PSQM and PAMS; objective measurement of factors including subjective values

Digital Signal Processors (DSPs)

• Provide 3 major services – voice termination, transcoding, conferencing
• Also performs compression (codec), echo cancellation, voice activity detection (VAD), comfort noise generation (CNG), and jitter handling
• Conferencing among participants with the same codec is called a single-mode conference.
• Conferencing among participants with different codecs is called a mixed-mode conference.

Protocols

• VOIP calls run over Real Time Protocol (RTP).
• RTP provides sequence reordering, time-stamping, and multiplexing
• Rides on UDP ports 16384-32767
• Voice does not need the reliability (retransmission) of TCP since retransmitted data is no longer useful (I already said that).
• VOIP packets headers:
  • IP – 20 bytes
  • UDP – 8 bytes
  • RTP – 12 bytes
  • L2 headers vary depending on technology (Ethernet = 12 bytes, MPLS, etc.)
• 2 10-ms packages are usually in one packet (20ms of voice)
• G.711 (64kbps) produces 160 bytes from 20 ms of voice.
• G.729 (8kbps) produces 20 bytes from 20 ms of voice.

cRTP

• Compressed RTP (cRTP) reduces the headers
• After the first packet lands, the IP, UDP, and RTP headers won’t change, so why send them again?
• The headers are reduced to a hash.
• cRTP reduces the headers to 4 bytes with a UDP checksum and 2 bytes without a UDP checksum.
• Slow links only
• Processing overhead
• Finite delay in packetization

Packet Size Effect on Bandwidth

• The size of a voice frame depends on:
  • Packet rate and packetization size – rate is inversely proporational to size
  • IP overhead – RTP, UDP, IP, cRTP overhead
  • L2 overhead -
  • Tunneling overhead – IPSec, GRP, MPLS, etc.
• Codecs have different bandwidth
  • G.711 (PCM) – 8000 samples per second @ 8 bits per sample = 64 kbps
  • G.726 (Adaptive Differencial PCM – ADPCM) – Variable bit rate of 32 kbps, 24 kbps, or 16 kbps
  • G.722 (Wideband Speech Encoding) – 2 subbands using modified ADPCM of 64 kpbs, 56kbps, or 48 kbps
  • G.728
  • G.729 – 10 samples per 10-bit code = 8 kbps

Calculating Total Bandwidth

• Step 1 – Determine codec and packetization period: What does the codec require in bandwidth?  How many samples per packet (usually 2)?
• Step 2 – Determine link-specific overhead:  Encapsulation?  cRTP?
• Step 3 – Calculate packetization size:  Size of voice payload; codec bandwidth * packetization period / 8 = voice payload in bytes
• Step 4 – Calculate total frame size: IP + UDP + RTP + Tunneling + data link + packetization size
• Step 5 – Calculate packet rate: 1 / packetization period (ex., 20ms packetization period is 1/0.020 = 50 packets per second)
• Step 6 – Calculate total bandwidth:  Total frame size * packet rate

VAD and Bandwidth

• Common for 1/3 of conversation to be silence
• VAD bandwidth savings depends on:
  • Type of audio: regular phone call (two-way), conf call (one-way), music on hold (MOH)
  • Background noise: noise may be detected as voice
  • Other factors:  language, culture may influence amount of silence

Enterprise VOIP Implementations

• Consists of gateways, gatekeepers, Cisco Unified CallManagers (CCM), Cisco IP Phones
• Routers can provide the voice gateway function by connecting the IP network to the WAN (and other gateways), PSTN, PBXes, etc.
• Survivable Remote Site Telephony (SRST) allows local calling and use of PSTN while services are down

Functions of CCM

• Call processing – routing, signaling, accounting
• Dial plan administration -  call routing
• Signaling and device control – configuration and instruction in case of events
• Phone feature administration – button programming, profiles, etc.
• Directory and XML
• API for interface – allows custom programming for IP phones

Enterprise Deployment Models

• Single-site: You have one site, and everything is there.
• Multisite with centralized call processing: You have multiple sites, but the main site has the CCM cluster.
• Multisite with distributed call processing: You have multiple sites, and each site has its own CCM cluster.
• Clustering over WAN: You have multiple sites, and each site has a part of one big CCM cluster.

IOS Voice Commands

----- R1 -----
! FXS on 1/1/2
Dial-peer voice 1 POTS
 destination-pattern 120
 port 1/1/2

! Extension 230 is on R2
Dial-peer voice 2 R2
 destination-pattern 230
 session target ipv4:10.1.1.2

----- R2 -----
! FXS on 2/2/1
Dial-peer voice 1 POTS
 destination-pattern 230
 port 2/2/1

! Extension
Dial-peer voice 2 R2
 destination-pattern 120
 session target ipv4:10.1.1.1

Call Admission Control (CAC)

• QoS can guarantee bandwidth but can only reserve so much (say, for 2 simultaneous calls).
• CAC make sure that resources are available (denies a new call if 2 calls are already placed).
• Dropped packets affect every call – not just the new ones

—–

Additional Reading

1. H.323 Sources on Wikipedia
2. MGCP – RFC 3435
3. SIP – RFC 3261
4. Nyquist Theorem on Wikipedia
5. MPLS on Wikipedia

Aaron Conaway

I like to lean my head to the left, hit it with the palm of my right hand, and document what knowledge falls out.

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