CCNA Collaboration – Notes, Chapter 1

These notes are based on reading the official certification guide “CCNA Collaboration CICD 210-060 Official Cert Guide“.


CICD – Ch1 – Traditional Voice vs. Unified Voice

Analog Voice

Analog Signal

  • Electrical voltage, frequency, current and charge used to convey properties of voice

Analog Phone Lines

  • Uses properties of electricity to convey changes in voice over cabling
  • Must convey signaling
    • Dial tone, dialed digits, busy

Analog Circuit

  • Pair of wires
    • Ground (positive) – TIP
    • Negative (battery) – RING
    • Where concept of TIP and RING comes from

Loop Start Signaling

  • 2 wire phone with 48v DC current
  • Typically found in home environments
  • Problem – Glare
    • Occurs when going to pick up phone off hook and at the same time a call comes in before the phone can ring

Key System – PBX

  • Have chance of glare occurring

Modern PBX

  • Found in larger corporate environments
  • Ground start signaling
    • Originated by the pay phone
    • Allows PBX to separate an answering phone from incoming phone line

Supervisory Signaling

  • On hook, off hook, ringing
  • Informational signaling
    • Dial tone, busy, ring back
  • DTMF – Dual Tone Multi Frequency

Analog Signal Problems

  • Degraded over distances, signal loss
  • Repeaters installed to increase distance
    • Regenerated the signal
    • Could not differentiate between voice on wire vs. line noise
    • Line noise would be amplified
  • Number of wires the phone company needed to run and maintain
    • Each phone required 2 wires
    • Bundles of wires hard to maintain

Digitizing Voice

  • Process of changing analog voice signal into a series of numbers
  • Digital transmission eliminated need for individual pair of wires required for analog connectors

Traditional Digital

  • TDM, Time Division Multiplexing
  • Digital encode multiple conversations over a single 4 wire path
  • Numeric value transmitted in specific time slots
    • Different conversations

T1 Circuit (US, Canada, Japan)

  • 24 separate 64 kbps channels
    • DS0
  • Each channel supported 1 voice call

Channel Associated Signaling – CAS

  • Signaling information is transmitted in the same channel as voice
  • Robber bit signaling – RBS
    • Steals a bit from voice channel to transfer signaling information
  • T1 uses 8th bit of every 6th sample / frame

Common Channel Signaling – CCS

  • Dedicated one DS0 from T1/E1 for signaling
  • Out of Band (OOB) signaling
    • Signal is completely separate from voice traffic
  • T1 has 23 usable DS0’s for voice
  • Signaling protocol – Q.931
    • Used on ISDN circuits
    • SS7 signaling for CCS between CO’s
  • Allows PBX vendors to communicate proprietary messages and features between PBX systems using ISDN
  • T1 uses 24th slot
  • E1 uses 17th slot

PSTN – Public Switches Telephony Network

  • Establish world wide pathways to allow people to easily connect, converse and disconnect
  • Components
    • Analog phone
      • converts audio to electrical and vice versa
      • Connects to PSTN
    • Local Loop
      • Link between customer and service provider
    • CO Switch
      • Provides services to devices on local loop
      • Signaling, digit collection, call routing, setup and teardown
    • Trunk
      • Connection between CO Switches
    • Private Switch
      • Miniature PSTN inside company
      • PBX?
    • Digital Phone
      • Connects to PBX
      • Converts audio into binary

PBX – Private Branch Exchange

  • Manage in-house phones (business)
  • Allow internal calls without using PSTN resources
  • Connects internal phones and connects to PSTN
  • Components
    • Line Cards
      • Provides connection between phones and PBX
    • Trunk Card
      • Provides connection from PBX to PSTN or other PBX’s
    • Control Card
      • Intelligence of PBX
      • All call setup, routing and management functionality
  • Key System
    • Geared for smaller environments (50 users or less)
    • Fewer features, more of a shared line feel
  • Connections to PSTN
    • CCS style signaling
    • Call setup, routing, billing, informational messages
    • Call > CO > # lookup > forward to destination

PSTN Number Plans

  • Must use valid E.164 standard address
  • E.164 is international number plan created by International Telecommunications Union (ITU)
    • Limited to max of 15 digits
  • Components
    • Country Code
    • National Destination Code
    • Subscriber number

VOIP – Voice over IP

  • Send voice traffic over data network
  • Concerns
    • Ensuring packets get to destination intact and quickly (QoS)
    • Coding / Decoding (Codec)
    • Security, not snooped or changed in transit (Encryption)
  • Benefits
    • Reduced cost of communications
      • Use existing WAN/Internet for calls
      • No toll charges
    • Reduced cabling
      • Run single ethernet to desk
    • Seamless Voice Networks
      • Uses business network, not PSTN
      • Central control of all voice devices
      • 4 digit dialing across the world
    • Reduced cost of MACD
    • Softphone
    • Unified email, voicemail and fax
    • Increased productivity
      • Ring all devices
    • Feature rich
    • Open standards

Converting voice to packets

  • Dr Harry Nyquist – created mathematical foundation to convert analog signals (flowing) waveforms into digital format (binary)
  • 3 Step Process (optional 4th)
    • Samping
    • Quantization
    • Encoding
    • (optional) Compression
  • Audio Frequencies
    • Human ear can hear > 20 – 20,000 Hz
    • Human speech > 200 – 9,000 Hz
    • Traditional phone transmits > 300 – 3,400 Hz
    • Standard equipment to digitize human speech > 300 – 4,000 Hz
  • Nyquist
    • Sample twice the highest frequency (2×4000)=8000
    • Sample = 1 byte, 8 bits
    • 8000 samples per second, times 8 bits for each sample
      • Product = 64,000 bits per second
      • This is uncompressed audio
      • G.711 – 64kbps
    • Once sampling assigns a numeric value to all analog signals traffic gets encapsulated
      • UDP and RTP (real-time transport Protocol)


  • 2 main codecs on all Cisco IP Phones
  • Compression, reduce amount of bandwidth required for a call
  • MOS – Mean Opinion Score
    • Quality of various voice codecs
  • G.711
    • Common on all VOIP devices
    • 64 kbps per call
  • G.729
    • Compressed audio
    • 8 kbps
    • 2 Variants
      • G.729a (annex A) – sacrifices audio quality for better processor efficient coding
      • G.729b (annex B) – introduces VAD (voice activity detection), makes voice transmission more efficient
  • G.722
    • Default on new Cisco phones
    • Wideband codec
      • Reproduces a wider range of frequencies
      • Better audio
    • 64 kbps

Digital Signal Processor – DSP

  • Offload processing for voice related tasks from the routers processor
  • Chip that performs all sampling, encoding and compression functions on audio / video coming into the router
  • Packet Voice DSP Module (PVDM)
    • Memory stick
    • Comes in different sizes
  • Codecs consume DSP resources
    • Some consume more than others
    • DSP resources can handle (approx) double the number of medium complexity calls per DSP than high complexity

Real-time Transport Protocol – RTP

  • Operates at transport level of OSI model
    • UDP based traffic, does not require acknowledgement
  • UDP provides port numbers and header checksum
  • RTP adds timestamps and sequence numbers to header information
  • Random port – 16,384 <> 32,767
    • Always even number
  • Devices setup point to point RTP stream, one in each direction

Real-time Transports Control Protocol – RTCP

  • Reports statistics between 2 devices in the call
    • Packet count
    • Packet delay
    • Packet loss
    • Jitter (delay variations)
  • UDP based traffic
    • Random port – 16,384 <> 32,767
      • Always odd number
  • Separate session from RTP
  • Devices send RTCP packet once every 5 seconds

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