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
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
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
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 (6 th ) 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) DS0s 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)
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
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: 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
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
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