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
- Analog phone
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
- Line Cards
- 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
- CCS style signaling
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
- Reduced cost of communications
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)
Codecs
- 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
- Random port – 16,384 <> 32,767
- Separate session from RTP
- Devices send RTCP packet once every 5 seconds