1. Voice of IP Fundamentals
CHAPTER I+2
Overview of the PSTN and Comparisons to Voice over IP
Enterprise Telephony Today

2. Traditional Phone Connection

3. Traditional Phone Connection
Does Not Scale Well!
N x (N-1)/2

4. Traditional Phone Connection to Central Office

5. Traditional Phone Analog Waveform

6. Traditional Phone Analog Waveform Amplified with Noise

7. Digital Waveform Amplified with Noise

8. Human Speech

9. Human Speech:
Nyquist Theory (Dr. Harry Nyquist, Bell labs.)- To accurately recreate an electrical pulse the sampling rate must be twice the frequency of the original.
Human speech typically ranges up to 9000 Hz therefore the sampling rate must be 18,000 samples per second!

10. Converting Analog to Digital:
Sample the signal
Quantize the signal
Encode the quantized value into binary format:
Optionally compress the sample to save bandwidth.

11. Sample the Signal:
How often to Sample?
Nyquist – 18,000 Samples per second!
Realistically to recognize voice and mood 8,000 Samples per second.
Result less quality less bandwidth
Process referred to as Pulse Amplitude-Modulation (PAM)

13. Quantize the Signal:
How many Digits?
Known as Quantization
Divided into sixteen (16) segments. 0 through 7 positive and 0 through 7 negative
Values are not evenly spaced to allow for more accurate recreation of voice patterns

15. Encode the Quantized Signal:
How many Digits?
Each Quantized value is encoded into an eight bit (8) binary number.
Total bandwidth is equal to eight bits for each sample times eight thousand samples per second.
8 X 8000 = 64Kbps

16. Meshed Network of Central Office Switches

17. Hierarchical Network of Central Office Switches

18. Circuit-Switched Hierarchical Network of Central Office Switches

19. Public Switched Telephone Network (PSTN):
The Pieces:
Analog Telephone: Able to connect directly to the PSTN.
Local loop: Connection between the customer premises and the phone company central office.
Center Office (CO) Switch: Provides services to the devices on the local loop.

20. Public Switched Telephone Network (PSTN) continued:
The Pieces:
Trunk: Provides a connection between central office switches.
Private Switch (PBX): Allows a business to operate an “in-house” phone company.
Digital Telephone: Typically connects to a PBX converts audio into binary

21. DTMF Signaling

22. Address Signaling:
Dual-tone multifrequency (DTMF)-Each button on the keypad of a touch-tone pad or push-button telephone is associated with a pair of high and low frequencies. On the keypad, each row of keys is identified by a low-frequency tone and each column is associated with a high-frequency tone. The combination of both tones notifies the telephone company of the number being called, thus the term dual-tone multifrequency (DTMF).
Pulse-The large numeric dial-wheel on a rotary-dial telephone spins to send digits to place a call. These digits must be produced at a specific rate and within a certain level of tolerance. Each pulse consists of a “break” and a “make”, which are achieved by opening and closing the local loop circuit, The break segment is the time during which the circuit is open. The make segment is the time during which the circuit is closed. The break-and-make cycle must correspond to a ratio of 60 percent break to 40 percent make.

23. Multiple calls over a single line:
Time Division Multiplexing (TDM)
each call has a “time-slot”
T1 has twenty-four (24) time slots known as a Digital Signal. IE: Digital Signal 0 is DS0
E1 has thirty (30) DS0

24. ISDN

26. Signaling:
Channel Associated Signaling (CAS): Uses the same bandwidth as the voice. IE: In-band signaling, as in telnet. Because it uses bits of the voice for signaling it is referred to as “Robbed Bit Signaling” (RBS).
Common Channel Signaling (CCS): Uses a separate dedicated channel for signaling. IE: Out-of-band signaling as in a console connection or ISDN “D” channel.

27. Robbed Bit Signaling (RBS):
Uses the eighth (8th) bit on every sixth (6th) sample.
Uses the least significant bit (binary 1) to limit change in quality of voice transmission

29. T1 Frame:
Each T1 frame consists of:
Twenty-four (24) DS0’s of eight (8) bits
One framing bit
8 X 24 = = 193 bits
At 8000 frames per second (Nyquist)
Total is 193 X 8000 = Mbps

30. Super Frame (SF):
Each Super Frame sends twelve (12) T1 frames at a time.
Uses the twelve framing bits only for synchronization.

31. Extended Super Frame (ESF):
Sends groups of twenty-four (24) T1 frames at a time.
Of the 8000 framing bits sent every second:
Two-thousand (2000) are used for framing.
Two-thousand (2000) are used for error checking.
Four-thousand (4000) are used as a supervisory channel (Out-of-band)

33. DS0 Robbed Bits (RBS):
Four bits (One per every six (6) DS0) per twenty-four (24) frames (ESF)
A pattern of 1111 signals ringing
A pattern of 0101 signals off-hook

34. Traditional Call Set-up

35. Call setup, routing, and teardowns Call Supervision
Switching systems provide three primary functions;
Call setup, routing, and teardowns
Call Supervision
Customer ID and telephone numbers

36. Supervisory signaling
Address signaling
Informational signaling

37. Supervisory Signaling:
On-hook Signal: When the phone is on-hook there is no connection between tip and ring.
Off-hook Signal: When the phone is off-hook the connection between tip and ring is made and electrical current (signal) is present.
Ringing: To cause a phone (on-hook) to ring an AC (Alternating Current) signal is sent.

38. Informational Signaling:
Dial Tone: Indicates the phone company is ready to receive digits.
Busy: Indicates the remote phone is in use.
Ringback: Indicates to the originator that the receiving phone is ringing.
Congestion: Indicate the long distance network is not able to complete the call.
Reorder: Indicates the local network is not able to complete the call.
Receiver 0ff-hook: Indicates the local phone has been off-hook for an extended period of time.

39. Informational Signaling (Continued):
No Such Number: Indicates the dialed number is invalid.
Confirmation: Indicates the telephone company is attempting to complete the call.

40. Glare (Loop Start Signaling, Most common in Home):
When a user attempts to dial an outgoing call at the same time an incoming call is received, the two connect without ring or dial-tone.
More frequent in business where multiple incoming calls are received and multiple outgoing calls are made

41. Telephone Services: Call Waiting Call Forwarding Three-way Calling
Display
Call Blocking
Calling Line ID Blocking
Automatic Callback
Call Return
Circuit Switched Long Distance
Calling Cards
800/888/877 Numbers
Virtual Private Networks
Private Leased Lines
Virtual Circuits

42. VPN

43. Key Systems:
Geared to small business environments where the individual phones will have multiple PSTN lines and ability to share lines

44. Key-System

45. PSTN Call Through a PBX

46. Number Translation Through a PBX

47. Tie-Line

48. Tie-Line Cost

49. PSTN Numbering Plans (SS7) E.164:
Limited to fifteen (15) Digits
Country Code
National Destination Code
Subscriber Code

50. North American Numbering Plan (NANP) E.164:
Country Code
Area Code
Central Office or Exchange Code
Station Code

51. E.164:
Example:
Country Code = 1 (USA)
Area Code = 401 (Rhode Island)
Central Office Code = 825 (Warwick)
Station Code = 1000 (CCRI)

53. International Telecommunications Union (ITU) accepted in 1996.
H.323:
International Telecommunications Union (ITU) accepted in 1996.
Designed to carry multimedia over Integrated Services Digital Network (ISDN)
Based or modeled on the Q.931 protocol
Cryptic messages based in binary
Designed as a peer-to-peer protocol so each station functions independently
More configuration tasks
Each gateway needs a full knowledge of the system
Can configure a single H.323 Gatekeeper that has all system information
Each end system can contact the gatekeeper before making a connection
Gatekeeper can perform Call Admission Control (CAC) to determine if resources are available before a call is accepted
Gatekeeper and Gateway can be the same device

56. SIP was designed by the IETF as an alternative to H.323
SIP is a single protocol whereas H.323 is a suite of protocols as FTP is a single protocol within the TCP/IP protocol suite
SIP is designed to set up connections between multimedia endpoints
Uses other protocols (UDP, RTP, TCP….) to transfer voice or video data
Messaging is in clear ASCII text
Vendors can create their own “add-ons” to the SIP protocol
SIP is still evolving
SIP is destined to become the only voice and video protocol

58. IETF standard with developmental aid from Cisco
MGCP:
IETF standard with developmental aid from Cisco
All devices under a central control
Voice gateway becomes a dumb terminal
Allows minimal local configuration
Single point of failure
Uses UDP port 2427

60. Often called “skinny” protocol
SCCP:
Often called “skinny” protocol
Cisco proprietary
Similar to MGCP in that it is a stimulus/response protocol
Allows Cisco to deploy new features in their phones
Cisco phones must work with Cisco systems (CME, CUCM,CUCME…)
Cisco phones can also use other protocols such as SIP or MGCP with downloaded firmware

61. End of Chapter 1 & 2