02. ++Analog+and+Digital+Voice+Connections
Transcript of 02. ++Analog+and+Digital+Voice+Connections
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Analog and Digital VoiceConnections
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Homes and Small OfficeConnections
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Local Loops
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Ty pes of Local- Loop Signaling
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Supervisor y Signaling
On-hook signal : When the phone is on-hook, theconnection between the tip and ring wires is broken and noelectrical signal passes between them.
Off-hook signal : When the phone is off-hook, the phoneconnects the tip and ring wires, completing the circuit andallowing electrical signal to pass.
Ringing : To cause an analog phone to ring, the phonecompany sends an alternating current (AC) signal downone of the wires, which the phone detects and generates aring signal.
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Informational Signaling
Dial toneBusyRingbackCongestionReorder
Receiver off-hookNo such number Confirmation
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A ddress Signaling
Dual-tone multifrequenc y (DTMF): The buttons on atelephone keypad use a pair of high and low electricalfrequencies (thus dual-tone) to generate a signal each
time a caller presses a digit. DTMF is the predominantsignal type used in the United States.
P ulse : The rotary-dial wheel of a phone connects anddisconnects the local loop circuit as it rotates around tosignal specific digits.
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Dual Tone Multifrequenc y
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P ulse Dialing
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Informational Signaling with Call P
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On-Hook
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Off Hook
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Ringing
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Ringing
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Trunks
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Foreign Exchange Trunks
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Ty pes of Trunk Signaling
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Loop-Start Signaling
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Loop-Start Signaling
Loop-start is the more common of the access signalingtechniques.When a handset is picked up (the telephone goes off-
hook), this action closes the 48V circuit that draws currentfrom the telephone company CO and indicates a change instatus, which signals the CO to provide a dial tone.
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G round-Start Signaling
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G round-Start Signaling
Ground-start signaling is another supervisory signalingtechnique, like loop-start, that provides a way to indicateon-hook and off-hook conditions in a voice network.
Ground-start signaling is used primarily in switch-to-switchconnections.The main difference between ground-start and loop-startsignaling is that groundstart requires ground detection tooccur in both ends of a connection before the tip andring loop can be closed.
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E&M Signaling
E&M is another signaling technique used mainly betweenPBXs or other network-tonetwork telephony switches. E&Msignaling supports tie-line type facilities or signals betweenvoice switches. Instead of superimposing both voice andsignaling on the same wire, E&M uses separate paths, or leads, for each.
There are six distinct physical configurations for the
signaling part of the interface. They are Types IV andSignaling System Direct Current No.5 (SSDC5). They usedifferent methods to signal on-hook or off-hook status, asshown Table 3-4. Cisco voice implementationsupports E&M Types I, II, III, and V.
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E&M Signaling
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E&M Ty pe I
Type I signaling is the most common E&M signalingmethod used in North America.
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E&M Ty pe V
Type V : Type V is the most common E&Msignaling form used outside of North America.
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Trunk Supervision Signaling Inmediate Start
Immediate-start, is the simplest method of E&M accesssignaling.. This signaling approach is used for E&M tie trunkinterfaces.
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Trunk Supervision Signaling - Dela y Start
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Facts about Sounds
The average human ear is able to hear frequenciesfrom 2020,000 Hz.
Human speech uses frequencies from 2009000 Hz.Telephone channels typically transmit frequencies from3003400 Hz.
The Nyquist theorem is able to reproduce frequenciesfrom 3004000 Hz.
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Digitizing A nalog Signals
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B asic Voice Encoding : Converting Digital
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N yquist Theorem
If you sample at twice the highest frequency on a voiceline, you achieve good-quality voice transmission.
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Q uantization
The process of converting the analog wave into digital,numeric values is known as quantization.
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Q uantization Techniques
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Q uantization Techniques
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CodecsCodecs provide the coding and decoding translationbetween analog and digital facilities. Each codec typedefines the method of voice coding and the compressionmechanism that is used to convert the voice stream.The PSTN uses TDM to carry each voice call. Each voicechannel reserves 64 kbps of bandwidth and uses the G.711codec to convert an analog voice wave to a 64-kbpsdigitized voice stream.In VoIP design, codecs might compress voice beyond the64-kbps voice stream to allow more efficient use of network
resources.The most widely used codec in the WAN environment isG.729, which compresses the voice stream to 8 kbps.
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Example : Waveform Compression
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Example : Source Compression
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G.729 and G.729A Comparison
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G.729 and G.729A Comparison
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Voice-Compression Techniques
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Compression B andwidth Requirements
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Mean Opinion Score
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Mean Opinion Score
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Medium to Large Office Connections
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Digital Trunks
Digital trunks are used to connect to the PSTN, to aPBX, or to the WAN and are widely available worldwide.
Digital voice ports are found at the intersection of apacket voice network and a digital, circuit-switchedtelephone network
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Ty pes of Digital Trunks
T1: Uses Time Division Multiplexing (TDM) to transmitdigital data over 24 voice channels using CAS.
E1: Uses TDM to transmit digital data over 30 voicechannels using either CAS or CCSISDN: A circuit-switched telephone network system usingCCS. Variations of ISDN circuits include the following:
BRI: 2 B (Bearer) channels and 1 D (Delta) channelT1 PRI: 23 B channels and 1 D channelE1 PRI: 30 B channels and 1 D channel
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Signaling
Digital technology solves the problems of signaldegradation and the inability to send multiple calls over a single line that occur in analog technology, it creates a
new issue: signaling .With analog circuits, supervisory signals were passedby connecting the tip and ring wires together. The phonecompany generated informational and address signalsthrough specific frequencies of electricity.
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Signaling
To solve this, two primary styles of signaling were createdfor digital circuits:
Channel associated signaling (CA
S ): Signalinginformation is transmitted using the same bandwidth as thevoice.
Common channel signaling (CCS ): Signaling informationis transmitted using a separate,dedicated signalingchannel.
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Channel A ssociated Signaling (C A S )
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T1 Digital Signal Format
SF: Super Frame
ESF: Extended Super Frame
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Robbed- B it Signaling
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Robbed- B it Signaling
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DS0 Signaling B its in a Single T1 Extended Super Frame
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E1 R2 C AS
One big difference between an E1 and a T1 is that an E1bundles 32 time slots instead of 24. This results in abandwidth of 2.048 Mbps. With an E1, one time slot is usedfor framing and one is used for signaling. This leaves 30 timeslots available for user data.
A multiframe consists of 16 consecutive 256-bit frames. Eachframe carries 32 time slots. The first time slot is usedexclusively for frame synchronization. Time slots 2 to 16 and18 to 32 carry the actual voice traffic, and time slot 17 is used
for R2 signaling.The first frame in an E1 multiframe includes the multiframeformat information in time slot 17. Frames 2 to 16 include thesignaling information, each frame containing the signaling for two voice time slots.
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E1 R2 C AS
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T1/ E1 CAS
You are able to support 24 DS0 audio channels out of a T1 interface (used primarily in the United States,Japan, and Korea). E1 interfaces (typically usedoutside of the United States, Japan, and Korea)support 30 channels.
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Common-Channel Signaling (CCS )
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Framing and Signaling
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Common-Channel Signaling (CCS )
When using CCS configurations with T1 lines, the
24th time slot is always the signaling channel. Whenusing CCS configurations with E1 lines, the 17th time slotis always the signaling channel.
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ISD N (Integrated Service Digital N etwork )
ISDN is a circuit-switched telephone network systemdesigned to allow digital transmission of voice and dataover ordinary telephone copper wires, resulting in better quality and higher speeds than is available with the PSTNsystem.
There are 2 types ISDN BRI and ISDN PRI
Both media types use B channels and D channels. The B
channels carry user data. The D channel, in its role assignal carrier for the B channels,
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ISD N (Integrated Service Digital N etwork )
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ISD N (Integrated Service Digital N etwork )
Its an example of CCS!!
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ISD N N etwork A rchitecture
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La yer 3 ( Q.9 30 /9 31) Messages
ISDN uses Q.931 as its Layer 3 signaling protocol.
This protocols supports user-to-user, circuit-switched (the Bchannels), and packet-switched (the D channel) connections.
A variety of call-establishment, call-termination, information,and miscellaneous messages are specified, including SETUP,CONNECT, RELEASE, USER INFORMATION, CANCEL,STATUS, and DISCONNECT.
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La yer 3 ( Q.9 30 /9 31) Messages
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ISD N Signaling Messages (Example )
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Q S IG P rotocol
QSIG is based on ISDN Q.931 SignalingVendor independant Protocol to allow PBX-to-PBX Communication
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Signaling S ystem 7
In order for all the telephony providers of the world tocommunicate together, a common signaling protocol mustbe used, similar to the way TCP/IP operates in the datarealm.The voice signaling protocol used around the world isS ignaling Sys tem 7 ( SS 7).
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Signaling S ystem 7
SS7 is an out-of-band (CCS-style) signaling method used tocommunicate call setup, routing, billing, and informationalmessages between telephone company COs around the world.
When a user makes a call, the first CO to receive the callperforms an SS7 lookup to locate the number. Once thedestination is found, SS7 is responsible for routing the callthrough the voice network to the destination and providing allinformational signaling (such as ring back) to the calling device.
N ote : SS7 is primarily a telephony service provider technology.You will not typically interface directly with the SS7 protocolfrom a telephony customer perspective
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Signaling S ystem 7
Primaily used between COs, but may integrated with somePBX systems
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Signaling S ystem 7
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Sigtran
The SIGTRAN protocols are an extension of the SS7protocol family. It supports the same application andcall management paradigms as SS7 but uses an IPtransport called Stream Control Transmission Protocol(SCTP), which is used to carry PSTN signaling over IP.
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Signal Conversion Example
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The problem of faxes and modems
Much more sensitive to loss than voiceVoIP compression algorithms designed for voice3 Methods to handle Fax/Modem in VoIP:
Fax/Modem RelayFax/Modem Pass-ThroughStore and Forward
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Fax Rela y
2 Methods:Cisco Fax Relay: Oldest, Default, Supported since 11.3 IOST.38 Fax Relay: ITU-T Standard-Router receives and demodulates fax tones-Fax transmited between routers using T.38 or Cisco
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Fax Rela y
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Fax /Modem Rela y
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Fax P ass- Through
Very Simple Technique - Code Fax/Modem in G.711Excessive Bandwidth use compared to Fax RelayNo VAD (Voice Activity Detection)No Echo Cancelation
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Fax /Modem P ass- Through
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Fax Store and Forward
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Fax Store and Forward
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Echo
Echo is a result of electrical impedance mismatches in thetransmission path.
Echo is always present, even in traditional telephonynetworks, but at a level that cannot be detected by thehuman ear.
The two components that affect echo are amplitude(loudness of the echo) and delay (the time between thespoken voice and the echoed sound). You can controlecho using suppressors or cancellers.
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Echo is alwa ys present
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Echo Cancellation
The term echo cancellation is used in telephony todescribe the process of removing echo from a voicecommunication in order to improve voice quality on atelephone call.
In addition to improving subjective quality, this processincreases the capacity achieved through silencesuppression by preventing echo from traveling across anetwork.
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Echo Cancellation
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4 -Wire Conversion and Echo
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VA D: Voice activit y detection
VAD is a technique used in speech processing in which thepresence or absence of human speech is detected. Themain uses of VAD are in speech coding and speech
recognition. It can facilitate speech processing, and canalso be used to deactivate some processes during non-speech section of an audio session: it can avoidunnecessary coding/transmission of silence packets inVoice over Internet Protocol applications, saving oncomputation and on network bandwidth.
Voice activity detection is usually language independent.
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