VoIP Theory

7/28/2019 VoIP Theory

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Panda KD Ontime

May-07

VoIP Theory

and

Knowledge


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Objects list

Basic of Telephone Internet Telephony/VoIP

Codecs

Packetization

Jitter Delay

H.323

Routing

Network Topology

configuration options


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Basic of Telephone - Voice

It is a subclass of acoustics category

It is applied to the frequency area between 20 to 20,000Hz

The cognized and understood frequency between 300 to 3300Hz

The voice can hold with mistake but not delay and resonance

If we have done nothing to change itwe can only gain one voice

conversation gateway on a piece of line / loop / truck / circuitry. Even if we

extend the bandwidth, it cant be changed

Circuitry exchange set up a fixed end to end path to put up the in-phase

continue transmission link

Trusty

Top-quality

Can set up continue time and voice transmission delay Low-use rate for resource (silence, bandwidth)


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Digital voice

The voice whose frequency is between 300 to 3300 Hz can carry through digital

coding at the less loss , and become the 8000* 8bit/s sampling frequency Digitalsignal

Most telephone network are data network except for local loop

The digital coding and decoding commonly take place at the beginning of

exchange and the local loop joint.

In view of quality tolerance64Kb PCM coding(aka G.711) is widely applied to

public network

Measure errorthe reason of loss fidelity is that using the limited digital scale to

carry through continue frequent spectrum coding.

Compandinghow to progress digitally coding

Try hard to get the less loss and mistake at the low frequency.

G.711 mu-law (North America/Japan)

G.711 A-law (elsewhere)

The new and low bandwidth coding project was given

Special circuitry networkADPCM, CVSD

Data pack exchangeG.711, G.723, G.729, elemedia SX7300,


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Data pack transmission

The transmission delay is ambulatory and uncertain at the data pack exchange;

The network cant guarantee the data pack transmission s order and at the end Data pack exchange allows band mistake inspection and checkout exchange in

the possible delay

Most data application can tolerance delay but not false transmission

In theory,every pack can absolute transfer through different route

Data pack network will settle the congestion problem by the way ofabandoning separate data pack

Data pack transmission can be benefit from high bandwidth for every task Technique and demands push the drop of processor cost and improve the

disposal ability. Speed routing and exchanging

High capacity

Technique and demands promote the drop of transfers cost and improve itsability.

Fiber optics improves the bandwidth. Wavelength divides up WDM


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Technical term

ingress - entry point to the network (which one?)

egress - exit point from the network PSTN - Public Switched Telephone Network (but often includesprivate as well)

SCN - Switched Circuit Network (public or private)

ANI - Automatic Number Identification (caller ID)

DNIS - Dialed Number Identification Service (destination number)

ISDN - Integrated Services Digital Network multiple channels of voice/data over a single facility (B-channel, bearer)

separate signaling channel (D-channel)

user and network sides

POTS - Plain Old Telephone Service (usually analog service, no features) can carry voice, fax (via modem), or data (via modem)

framing - identifying the boundaries between one chunk of info and the next

DTMF - Dual Tone Multi-Frequency (touch-tones)

MF - Multi-Frequency (tones, but NOT touch-tones)

CDR - Call Detail Records (information usable to generate usage accounting/bills)


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Technical term II

T1 - facility capable of carrying 24 channels of voice/data

ISDN - 23B+D CAS (channel associated signaling) 24 channels

64 Kbps channels

used primarily in North America and Japan

signaling choices: RBS - robbed bit signaling - cannot use all 64Kbps

clear channel - can use all 64Kbps

note: a voice T1 is not like a data T1 that has no channels and no framing

E1 - facility capable of carrying 30 channels of voice/data

used elsewhere

MFC-R2 (multi-frequency compelled) - provides 30, 64Kbps channels

ISDN PRI

SIT - Special Information Tone - audible message played to caller

indicating a problem; begins with a tone sequence that identifies thecondition


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Technical term III

local loop

the facilities between the residence/business and the first switch in the

carriers network

comfort noise

pure silence on a voice connection can be interpreted by a person as a

broken connection. Comfort Noise is a low level of noise transmitted

instead of silence to assure the user that the call is still connected. echo

arises from imperfect analog components, crosstalk) between wires, and

acoustics (speakerphones)

undetectable when delay is small

VoIP delays make this more noticeable and it must be addressed

hunt group

a collection of lines/trunks that are interchangeable from the perspective of

the caller (e.g., a modem pool, a call center queue, or a set of gateway T1s)

and are dialed using the same phone number


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(TDM) - eg., ISDN, T1, E1

On a single equipment or line, to distribute fixed time slice for each independent

list and form into distributing multi-bit single list Time slice is fixed and unaltered in spite of any signal is important or not.

The data pack exchange hold equipment or route channel on a digitally list butonly when it requires transmission

1 0 1

0 1 1

0 1 0

0 0 1 1 1 0 0 1 1

DS0s CompositeInput Channels

3 2 1 3 2 1 3 2 1 Time Slots

Framing Bits

3 2 1 Frames

MUX

1

2

3


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Loop exchange

Phone

Circuit Switched Network

CO CO

Local Loop

Dedicated

Circuit

Local Loop

Analog* to Digital -------------------------------> Digital ----------------------------------> Digital to Analog

Varying Distances (a few miles or a few thousand)

Loop exchange

depart a end to end resource on a wholecalling process Tradition telecom technique( analog or data)

e.g.TDM

*In the case of ISDN,

voice is digital in the local loop as well.


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Internet telephone/ VoIP

VoIP transfers voice and other real time media on IP

routing pack exchange network.

Concluding :Internet, Intranets, extranets

VON - Voice Over (data) Networks concluding

ATM, frame relay, and IP

Voice and data can share a same network

Can integrate multi-media on a same network

Can improve the efficiency of network using and

running

gradually depressed leverage by using the data network

fee

With the data voice gradually improvingcan became

the eximious middle transmission


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Internet Telephony/VoIP (2)

VoIPs challenge

VoIP is designed in order to improving the using rate of

bandwidth

VoIP cant guarantee

Transmission speed

Delay speciality Packages reaching

Pack resulting in delay

The new challenge

The voice not only can be benefit from package but also

identify preferential using bandwidth or booking bandwidthby PRI

Tone package tends to smaller than data package


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ISO VoIP model

L1

L2

L3 - IP

L4 - TCP L4 - UDP

RTP/RTCP

coding/framing

digitized voice

LWP/ALP H.225/H.245 RAS

Call signaling/Call routing/User Authentication


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Coders

G.711 64Kbps project standard

64Kbps has the higher demands than existing POTS

modems (33Kbps, 56Kbps) transmission data.

Low bandwidth coders is feasible.

The benefit of Coders POTS modems transmission is feasible.

Other ingress media bandwidth requirement

DSL, cable, LAN, WAN

Master media bandwidth requirement

LAN, WAN

Coders challenge:

Need the more strong Coding ability

the extra delay in transfers.


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Coders (2)

coder bandwidth applications

G.711 64K PSTN voice, MMCX

G.723 5.3K, 6.4K remote access, ITS-SP, NetMeeting, [MMCX]

G.728 16K H.320 ISDN videoconferencing

G.729a 8K VoIP, VTGW4.0, GW-1000 - send silenceG.729b 8K suppress silence


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Packing

Exchanging the consecutive digitally list to

discontinuous data

The delay produce because the integrity package would

wait all the digital package reach, assembled.

Can get rid of time space of transmission silence

Trusty TCP v. UDP

TCP is acted to the transmission signal information by setting

up and back out.

TCPs package head is so large for the tone package data but

UDP is adapted to transfer tone package.


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Frame is not only frame

The tone packages frame

One unit of Sampling codefixed tong continue time

10, 15, or 30 msec.

One or more frames can pack a single IP package to transfer.

Can be searched by coders

Frames means that each second communication needs lesspackage and less consume, but it will increase delay.

Frames other term use

TDPseparating one time intervals bits from the next

ISO layer 2, e.g., Ethernet frames carrying IP packets insubnets


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Jitter

The reaching time of tone package is different.

The package unstable transmission delay will be takenpace some problem as followed,

Resource contention

The traffic difference of network transmission

Different package will have the different transmission path.

It will result in clearance, click,poop, interregnum and etcat the target tone bunch

Solution Voice digitally buffer can make the long time delays package

arrested

serial contraction buffer or append

At the same time Voice cushion result in delay


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Jitter without buffer

321

321

Packetized &

Compressed Voice

Transport Through

Packet Network

321Voice Packets Received

with Delay and Jitter

321

Regenerated Voice signal

without Jitter Buffering

largest delay

Voice signal


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Jitter with buffer or append

321

321

321

Jitter Buffering321

delay


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Delay

Delay from packet

codec

Network element

Network transmission

jitter (build out)

result The appreciable condition of listener

Easy to be apperceived when it exceed 250ms

solution fast PCs, Servers

Special equipments (gateways, DSPs)

Adjusting network (optimizing equip, engineered networks)

Real time monitor network (QoS) - DiffServ, MPLS, RSVP...


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Call routing vs. IP routing

Call routing

Gatekeeper provide service

Which network resource can be used form one end of IP network to the

other end?

After gaining ingress calling on a network gateway, it will decide which

gateway can provide egress phone and which one is best.

Gatekeeper use routing list to decide routing.

IP routing

Form a known IP address found a path to reach another known

IP address hubs, switches, and routers can do it.


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Network function module

gateway

Connect to PSTN and deal with inbound call , outbound call

and outbound real-time Pots tones

Connect to PSTN and deal with fax inbound call and

outbound real-time fax

Connect to PSTN and deal with PC inbound call Connect to package interchange network for transfer tone in

IP network

Interchange tone with other gateways

Not progress IP routing and switch

Progressing transfer circuit-package interchange

Progressing pack and unpack

Send IVR, gain keyboards number

Interchange routing and singnal with Gatekeeper


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Network function module (2)

gatekeeper

Connect with package interchange network

Manage gateway zone

Not carry IP route and switch

Authenticate gateway

Authenticate customer

Decide which zone can call outbound (interzone routing)

Decide which gateway can call outbound (intrazone routing)

Send call route choices to inbound gateway

Deal with call setup and divide off signal transfer in packageinterchange network

Not dealing tone transfer

Deal PCs H.323 proxy services


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Network function module (3)

Network manager

Connect to package interchange network

Not taking part in real-time call function

Use to manage and console gateway and

gatekeeper


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What is call

Connect in two or more endpoints

In QEA networkone POTS to POTS call need three

steps

1) the original call POTS to inbound gateway (circuits interchange)

2) inbound gateway to outbound gateway (package interchange)

3) outbound gateway to destination POTS (circuits interchange)


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H.323 standard

In no Qos network use multi-media (tone/image/data)

The first version use tree standard in June 1999

Defined at physical layer and transfer layer

Design base the interchange H.320but need gateway

For network design and not excluding Internet

Strong industry support - Microsoft, Intel, PictureTel, Lucent... The second version in 1998


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H.323 term

Terminal - H.323 end user device (PCs, workstations)

Endpoint - H.323 terminal, gateway, gatekeeper, MCU, etc.

Gatekeeper - H.323 endpoint that performs address resolution, admission

control and endpoint registration.

Gateway - H.323 endpoint that provides the service that allows H.323

terminals/MCUs to interoperate with other non-H.323 devices (e.g., H.320,PSTN, H.324, etc.)

Zone - Administrative domain controlled by a Gatekeeper, which includes

other H.323 elements

direct routed model - gatekeeper provides addresses of endpoints /

gateways but does not participate in signaling to them

gatekeeper routed model - gatekeepertandems signaling between

endpoints / gateways


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H.323 signal protocol

RAS - Registration, Admission and Status

Protocol used to by endpoints to register with a gatekeeper, find otherH.323 endpoints, ask permission from the gatekeeper to make calls and

provide status information between H.323 endpoints.

H.245

Protocol specifies how endpoints negotiate to determine which audio,

video and data coding schemes they are capable of and which will be

used.

It is then used to make connections for the exchange of the agreed upon

audio, video and data streams.

It also has some miscellaneous functions like notifying endpoints when

parties join or leave conferences.

H.225

call setup, features, RAS

includes Q.931 (the ISDN mechanism )


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H.323 information transfer protocol

RTP

Protocol describes how a sender should break up audio or video

streams in chunks, order and time stamp them and how the

receiver should perform the reverse operations.

RTCP - Real Time Control Protocol

The overlay protocol that reports on delay, lost packets, packets

received out of order, etc. for RTP streams


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Internet protocol

Transfer control protocol (TCP) Call connect from each gateway to gateway

Call signal (connect and hang up)

Call control

Translate DTMF tone (cant include in some coder)

User data protocol (UDP)

Transfer fax or compress tone

RTP

RTCP

RAS


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Phone-to-phone and fax-to -fax

IP

Network

Fax

Switch

Gatekeeper

Phone

Gateway

Network Manager

To Provisioning System

To Billing System

To QoS Mgmt

Fax

Switch

Phone

Gateway

Debit/Credit Calls

Phone to phone call scenario


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Phone-to-phone call scenario(two step)

1. Inbound gateway phone (one step)

2. PSTN send call to inbound gateway 3. If ID and PIN are authenticated

Inbound gateway prompt input ID PINand receive ID and PIN

Inbound gateway contact its gatekeeper and apply forauthenticating ID and PIN

Gatekeeper authenticate ID and PIN then send it to inbound

gateway 4. Inbound gateway prompt typing the destination phone

number (two step)

5. Inbound gateway send destination phone number to itsgatekeeper

6. Inbound gateways gatekeeper reference its INTERzone routelist and decide which zone can dial out to destination phonenumber



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Phone-to-phone call scenario(two step)

7. inbound gateways gatekeeper can dial out to destination

phone numbers gateway according prior sequence

8. Gatekeeper send the gateway list to inbound gateway

9.inbound gateway contact outbound gateway and dialing

out according the gateway list in turn

10. When connect setupoutbound gateway setup connectto destination phone use PSTN

11. Inbound and outbound gateway transfer tone package

12. If inbound gateway cant find the usable outbound

gateway, then hang up



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Phone-to-phone call scenario(one step dialing)

(1) dialing destination phone number

PSTN must send ANI and DNIS

PSTN must use DNIS check gateway phone number

(ex. take 1+, or 1010XXX+)

(2) not prompting type ID and PIN

use ANI as customers ID; PIN asguest

(3) not prompting type destination phone number

use DNIS as destination phone number


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Dialing and Dialing Dlan

Routing number

QEA try to find the outbound GWs number

PSTN international number (not including estimate access number)

VPN Not explaining by PSTN, perhaps including private area code

Describers dialing format How to explain the typing serials which do by caller

Capture by inbound GWs PSTN (one step dialing)

Capture by QTGW (two step dialing)

The number format of terminal gateway Gateway how to produce the dialing serial to outbound PSTN

Two kinds of number formats

Gateway how to connect all kinds of PSTN switch

PSTN including remote and international dialing

VPN interior and PSTN dialing

Three formats (for outbound):

Fixed format

dialing number always fixed Transformable format change dialing format by the caller source

User-defined format dialing by different rules base your requires


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Number

Income GW(NYC)

CO

Outbound GW

(Sao Paulo)

PBX

(1) User Dail

011551187654321

(5) Dest. GW Number

*987654321

Dest. GK

Original GK

(2) Route Number

551187654321

(3)Route Number

551187654321

(4)Route Number

551187654321

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VoIP Theory

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