VoIP (Voice over IP) Introduction Course

VoIP Technology TrainingVoIP Technology Training

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VoIP (Voice over IP)Introduction Course


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Course Outline

1. History of Telecomm

2. Circuit Switched Network

3. 7-layer communication model

4. IP Packet Switched Network

5. Packet Network Considerations


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1. History of TelecommOperator control

• Bell files telephone patent, Feb 14, 1876.

• Phones connected over wire “loops”to central operator.

• User cranks magneto to power the phone and signal operator.

• Operator controls circuit making connections using plugs and patch panel.

• Operator plug has a Tip, Ring and Ground/Sleeve conductors.


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1. History of TelecommElectromechanical Switching

• Almon Strowger patents Strowgerswitch, August 1888. Automates control using “step by step switch”.

• User “dials” to identify called party (431), the dial opens and closes circuit based on the digit dialed.

• Step by step, relays respond to “digits”to set-up circuit. Signal sent from one stepper switch to the next to complete a circuit through the matrix.

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4

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1

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1. History of TelecommElectronic Control

• 1960’s electronic control using micro-processors implemented.

• Micro-processors control analog switch matrix, path established through cross-point matrix.

• Faster switching, more reliable, reduced maintenance, lower cost .

• Shift to CCS (Clear Channel Signaling) with SS7 (Signaling System 7) for CO to CO.

• Tone (DTMF) introduced for user signaling, greatly improves switching speed.

A

B(A+B)

uPaddressbus

clockdatabus


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1. History of TelecommDigital Switching Systems

ts 1 ts 2 ts 3 ts 4

• Voice is digitized at network edge.• Multiple digital samples grouped.• Signal sent in time-slots through a TDM

(Time Division Multiplex) switch.• ISDN, Integrated Digital Services Network,

introduced in 1980’s in two forms:– BRI Basic Rate Interface, 2B+D– PRI Primary Rate Interface, 23B+D or

30B+2D

• B, bearer channel, 64 Kbps TDM switched.• D, data channel, 16 or 64 kbps signaling

channel, always on.


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• Payload (voice) encapsulated with sender/receiver IP address and header for packet.

• Packet networks promoted by IBM, CSMA/CA passing a “token” since ’70’s.

• Ethernet developed at Xerox ARC, CSMA/CD, became IEEE 802.1

• IP evolved from military experiment with distributed packet communication network, ARPANET.

• Business IP LAN/WAN grown rapidly, Ethernet networks everywhere, network data traffic exceeds voice traffic.– Corporate need for data– World Wide Web explosive growth

1. History of TelecommVoIP (Voice over IP networks), the future


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• Several types of PSTN analog circuits based on “signaling”:– Loop Start, basic FXS interface– Ground Start– E&M Tie Line

• Tone/Pulses interpreted by LEC’s CO equipment.

• Tones and Loop supervision used for other signaling, request for service, alerting, CID, disconnect, etc.

• Voice modulates “battery feed”current with mic, demodulates with speaker.

2. Circuit Switched NetworksAnalog PSTN

PSTNNetwork

FXS port(SubscriberInterface)

BatteryFeed

Switch

Ring Generator

dial

ringerhandset

FXO port(PSTN OfficeInterface)

Tip

Tip

Ring

Ring

Switch

Hook-switch

48V


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• Originally developed for Office to Office connection T for “trunking”, 1 for first level, T1.

• E1 a Euro based T1 like standard.• Voice digitized: Nyquist sample at 8,000 times

per second, each 8 bits for a 64 kbps signal, PCM (Pulse Code Modulation).

• Voice codec-T1: u-Law, E1: a-Law.• Broadband pipe to multiplex PCM signals to a

time-slot: T1 @1.544 MHz for 24 time slotsE1 @2.048 MHz for 32 time slots

2. Circuit Switched NetworksDigital PSTN, T1/E1, ISDN PRI

ts 1 ts 2 ts 3 ts 4


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• Line coding, Bipolar signal eliminates DC component and must vary continuously to assure Clock Sync.– AMI-Alternate Mark Inversion

• Every other mark bit is inverted.• Used in T1 & E1.

– HDB3-High Density Bipolar 3• Substitute 3 zeros, inserts bipolar violation, output

depends on odd/even 1’s since last substitution.• Used with E1.

– B8ZS-Bipolar 8 Zeros Substitution• 8 consecutive null bits are recoded with bipolar violation.• Used with T1.



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• Framing, needed to distinguish start/end of channels:– D4, ESF for T1– CRC4 & NOCRC4 for E1

• T1 framing added to end of frame as bit 193.• Framing, channel 0 for E1.• Signaling for T1, bit robbing, LSB every 8th sample CAS

(Channel Associated Signaling).• E1 uses channel 16 for signaling.• Simulate analog trunks Loop/Grd/E&M• R2 signaling used for enhanced service and call control.


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• ISDN PRI uses T1/E1 for connection.• ISDN employs Message based signaling• CCS (Clear Channel Signaling) using separate time-

slot for signaling. T1: channel 24, E1: channel 16.



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• International Standards Organization’s Open System Interconnection 7-layer Reference Model.

• Defines functional layers needed for communication between 2 or more network end-points.

• Message delivery requires each function layer. 7 Application Interactive Voice (VoIP)6 Presentation Codec5 Session RTP, RTCP4 Transport TCP & UDP3 Network IP, Diffserv2 Data Link Ethernet1 Physical UTP Cat 5

3. 7-layer Communication ModelISO’s OSI 7-layer model


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• VoIP standards define protocol messages and packet structure/content for each layer.

• “Protocol Stack” sum of protocol layers in OSI model,– User message must go down the stack at the sender

(Layer 7 to layer 1),– At receiver, message must traverse up the stack (Layer

1 to Layer 7).

• Several VoIP standards:– IEEE developed H.323– IETF developed SIP– MGCP used by DOCSIS (Cable TV), disassociated Gtwy– IEEE/IETF Megaco/H.248 IETF/IEEE disassociated Gtwy– Net2Phone proprietary

4. Packet NetworksVoIP Concepts


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• Defines functions for sharing physical medium and media access.

• Ethernet, 10/100 Base T LAN IEEE 802.• Uses Layer 1 as UTP Cat 5 cable.• MAC (Media Access Control) 6 byte address

unique for each device.• Sometimes called MAC layer

4. Packet NetworksVoIP Concepts, Layer 2 Data Link


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• IP (Internet Protocol) actually lower layer neutral.• IP Packet Network, IP defines packet content

w/Sender, Receiver address and IPv4 header.• IPv4 Packet Header 20 bytes (five 32-bit words)

include:– Type of Service byte (Diffserv code point)– Header & Packet size– Sequence– Fragmentation flag, Offset– Type of transport (TCP/UDP)– Check sums and DLL CRC

• Router protocols exist at this layer.

4. Packet NetworksVoIP Concepts, Layer 3 Network


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• Layer 4 defines controls reliability and other route characteristics.

• For VoIP TCP (Transport Control Protocol) & UDP (Uniform Datagram Protocol) employed.

• TCP “reliable” includes retransmit, used for VoIP network call signaling.

• UDP relies on application for transport reliability, Real-time nature of voice suggests use of unreliable transport, UDP for voice packets.

4. Packet NetworksVoIP Concepts, Layer 4 Transport


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• Each session (call or request) carries a unique id.• RTP (real-time protocol) & RTCP (real-time control

protocol), session statistics.• Receiver requests, application processes, e.g. for

excessive delay, app could request change codec.• Call (session) control signaling messages:

requestalertingconnectrelease

• ReSerVation Protocol (RSVP), Quality of Service protocol.

4. Packet NetworksVoIP Concepts, Layer 5 Session


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4. Packet NetworksVoIP Concepts, Layer 6 Presentation

• To conserve LAN/WAN bandwidth, codec applies compression at encoding, common Codecs:– G.711, PCM codec, 64 kbps– G.726, ADPCM, 16 to 32 kbps– G.723.1, MP-MLQ/ACELP 5.3/6.4 kbps

• Multi-phase-Max Likely Quantization & Algebraic Code Excited Linear Predictive

– G.729A, CS-ACELP, 8 kbps• Codec is selected from “Capability” message, may

be predefined.• Silence suppression using VAD (Voice Activity

Detection) and CNG (Comfort Noise Generation).


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• Inter-active Voice communications:– IP Phone call– Office to Office FAX– IP Conference Calling– Voice Mail– Unified Messaging, TTS “reads” e-mails

4. Packet NetworksVoIP Concepts, Layer 7 Application


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• Multimedia communication over an unreliable network, aimed at IP and UDP.

• Application Layer control protocol.• An umbrella standard employs many other protocols by

reference and Annex.• H.225 Call Signaling - q.931• H.245 Call Control – capabilities, media, logical channels• H.235 Security• H.450 Supplementary Services

– Forward– Hold– DND– Etc.

4. Packet Switched NetworksVoIP Protocols, IEEE H.323


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• Four “entities”:– Gatekeeper

Zone controller, RAS channel (register, status, admit)

– GatewayProvides access between disparate networks

– Terminal (IP Phone)End-point, user interface device

– MCU (Multi-point Control Unit)Conference controller & audio processor


Internet

Zone

GatewaysTerminals

Gatekeeper W/MCU

PSTN PSTN


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4. Packet Switched NetworksVoIP Protocols, IEEE H.323 Protocol Architecture

RTP

G.711G.729

G.723.1

H.261H.263

AudioApps

VideoApps

H.225.0Call

signaling

TCPUDP

IP

Link Layer 802.3

RTCPH.225.0

RASH.245

Controlsignaling

T.120Data

Terminal control and management

H.323 Stack


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• RAS Channel messages:request to GK, response from GK (confirm/reject):» GRQ - GCF/GRJ: GK Discovery request-response» RRQ - RCF/RRJ: Endpoint Registration request-response» URQ - UCF/URJ: Endpoint Unregistration request-response» LRQ - LCF/LRJ: Endpoint Location request-response» ARQ - ACF/ARJ: Admission request-response» BRQ - BCF/BRJ: Bandwidth request-response » DRQ - DCF/DRJ: Disconnect request-response» IRQ - IRR: Information request-response


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• GateKeeper must provide:Address translationAdmission controlZone Management & Bandwidth Control

• Terminal, Gateway, MCU must use GK once registered.• GK can manage calls at many levels:

v2 has Fast StartDirect call signaling & control modelGK routed call signaling & control

• v2 added Fast start, even Direct is complex.



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• connect

• setup

• alerting


• media type

• capabilities

• RAS• RAS

• audio codec


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4. Packet Switched NetworksVoIP Protocols, IEEE H.323 -Fast Start

H.3

23

en

dp

oin

t H.3

23

en

dp

oin

t

Gate

Keep

er

setup

ARQACF

ARQ

ACFalerting

connect

end sessionend session

Release complete

• Endpoint gets GK permission (ARQ/ACF).

• ARQ request for Fast start to GK with destination & media.

• Setup request between endpoints, includes media and codec.

• Other messages (alerting, connect) then open audio channel.

• Close audio channel (either endpoint)• Disconnect, return to idle advise to GK.

RAS = blackH.225 = blueH.245 = green

Audio

DRQDCF

DRQ

DCF


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• SIP: Establish multi-media communications sessions IETF’s RFC-2543.

• Text based protocol, similar to HTTP, SIP URLs SIP:[email protected].

• Low layer neutral, Reliable (TCP) or Unreliable (UDP), packet or byte network.

• Client/Server Application layer control protocol.• Default IP port 5060.• SIP Entities include User Agents and proxy, registrar,

location servers, redirect server.• Session has unique Call ID (call leg id) with CSeq

(command sequence) for each transaction controlled by a Call Agent.

4. Packet Switched NetworksVoIP Protocols, SIP (Session Initiation Protocol)


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4. Packet Switched NetworksVoIP Protocols, SIP Protocol Architecture

RTP/RTCP

G.711G.729

G.723.1

H.261H.263

AudioApps

VideoApps

UDP

IP

Link Layer, 802.3

SIP

Terminal control and management

SIP Stack

TCP

SAP/Q.931SDP


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• Requests from the User Client include:– INVITE– ACK– CANCEL– BYE– OPTION– REGISTER

4. Packet Switched NetworksVoIP Protocols, SIP

• Response message has 3-digit “Status Code”:– 1xx Information– 2xx Success– 3xx Redirect– 4xx Client error– 5xx Server error– 6xx Global failure

• SIP messages:

–Client to Server request from “User Agent, Client”.

–Server to Client response from “User Agent, Server”.


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• User Agent A, client requests service with INVITE msg.

• SDP (Session Description Protocol) defines type of media, codec, capabilities, etc.

• SDP messages included in SIP INVITE or ACK msg.

• User Call Agent B, server responds with called parties accept, redirect, reject response.

• User Call Agent A, client sends ACK after acceptance response.

• RTP Session (audio) established• Either User Agent BYE request ends

call

4. Packet Switched NetworksVoIP Protocols, SIP

INVITE+SDP

200+SDP

ACK

RTP/RTCP

BYE

200

Use

r A

gen

t A U

se r A

gen

t B

Proxy


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• Proprietary IP based protocol

4. Packet Switched NetworksVoIP Protocols, net2phone

Net2Phone Network

DoormanCall

ControllerOPALServer

Net2Phone Servers

PC Clients

MAX GW MAX GW


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• IP a “best effort” network, packet may be delayed or discarded at router/switch.

• Integrated (voice & data) services network must support Elastic Apps, Real-time Tolerant apps and Real-time intolerant apps.

• Interactive voice is RTI app, delay > 150 msecdistracting/service affecting.

• Excess Jitter > 30 ms, Network Congestion & Collision cause loss of relevance for time sensitive voice packet content.

• > 2% packet loss results in “poor” quality audio.

5. Packet Network ConsiderationsQoS (Quality of Service)


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• Two QoS mechanisms, reserve bandwidth or prioritize packets in queue (traffic shaping & queuing).

• RSVP layer 4/5 bandwidth ReSerVation Protocol, PATH from sender & RESV request from receiver.

• Diffserv prioritizes packets at layer 3 with TOS byte.

TOS byte code word:

BE(best effort)F

A(assured)F

E(expedited)F

• IEEE’s 802.1 p/Q, priority & queuing at layer 2, with TOS bit in Ethernet packet.



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• Jitter (delay variation) needs buffer to re-sequence voice packets.

• Echo with long delays aggravates quality, G.168 EchoCancellation quite acceptable.

• Codec, highly compressed voice requires significant processing power and time, DSPs common.– 723.1 delay 37.5 msec (30 msec frame + 7.5 msec look-ahead)

– 729A delay 15 msec (10 msec frame + 5 msec look ahead)

• UDP packet w/voice payloads are short, Repeating Hubs and long cable runs can affect collision, use Switch Hubs.

• Bandwidth management, ftp type jumbo-grams cause delay, protocols available at Router for fragmentation.



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• WAN bandwidth resources must be available for voice, dependent on codec and protocol.

• Bandwidth Required = (Packet header + payload).• Payload is codec output.

G.723.1 5.3/6.4 kbps 30 msec frameG.711 64 kbps 10 msecG.729A 8 kbps 10 msec frame

• IP/UDP/RTP packet headers = 40 bytes, at 30 ms Voice frames.– 1 frame/packet 40 * 33.3 = 1332 Bps: 10.656 kbps– 3 frame/packet 40 * 11.1 = 444 Bps: 3.552 kbps– 6 frame/packet 40 * 5.55 = 222 Bps: 1.776 kbps

• Adding header & payload, 723.1 needs 17Kbps @ 30 ms frame.• Note WAN generally Full duplex, LAN probably half duplex.



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• NAT (Network Address Translation) server controls use of scarce Public IP addresses.

• NAT generally not compatible with standard VoIP protocols.

• H.323– NAT assigns public IP address and port for outgoing

request changing address & port in original packet– Far end sends H.245 request to open a logical second

port on same IP address. NAT does not recognize the request on a different port and discards packet as illegal.

• SIP– Similar issue with NAT in opening audio path.

• DMZ hosting or port-forwarding by routers– Partial solution to VOIP-NAT problem

5. Packet Network ConsiderationsOther Packet issues


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• Firewalls– Prevent incoming access from WAN to LAN.– Prevent outgoing access to WAN from LAN.– Typically use Allow/Deny Table of address/port.– IP address can be allowed use of defined ports.– IP address can be allowed full I/O access.– Certain apps, ftp, telenet, etc. use fixed ports.

• Call set-up result, reject or in audio path set-up, minimizes security issue for pure VoIP device.

• Most Firewalls support H.323/SIP traffic

5. Packet Network ConsiderationsOther Packet issues


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Thank-you

VoIP (Voice over IP) Introduction Course

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