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

Traditional Phone Connection

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

Traditional Phone Connection to Central Office

Traditional Phone Analog Waveform

Traditional Phone Analog Waveform Amplified with Noise

Digital Waveform Amplified with Noise

Human Speech

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!

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

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)

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

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

Meshed Network of Central Office Switches

Hierarchical Network of Central Office Switches

Circuit-Switched Hierarchical Network of Central Office Switches

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.

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

DTMF Signaling

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.

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

ISDN

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.

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

T1 Frame: Each T1 frame consists of: Twenty-four (24) DS0’s of eight (8) bits One framing bit 8 X 24 = 192 + 1 = 193 bits At 8000 frames per second (Nyquist) Total is 193 X 8000 = 1.544 Mbps

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

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)

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

Traditional Call Set-up

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

Supervisory signaling Address signaling Informational signaling

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.

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.

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

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

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

VPN

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

Key-System

PSTN Call Through a PBX

Number Translation Through a PBX

Tie-Line

Tie-Line Cost

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

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

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

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

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

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

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

End of Chapter 1 & 2