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VoIP Theory and Knowledge

Panda KD Ontime

Objects list
Basic of Telephone Internet Telephony/VoIP Codecs Packetization Jitter Delay H.323 Routing Network Topology configuration options

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)

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 Digital signal 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,

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 of abandoning 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 its ability.
Fiber optics improves the bandwidth. Wavelength divides up WDM

Technical term
ingress - entry point to the network (which one?) egress - exit point from the network PSTN - Public Switched Telephone Network (but often includes private 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 can carry voice, fax (via modem), or data (via modem)

POTS - Plain Old Telephone Service (usually analog service, no features)

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)

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 the condition

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

(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 but only when it requires transmission
Input Channels



3 2 1

Framing Bits Frames









1 Time Slots

Loop exchange
Loop exchange depart a end to end resource on a whole
calling process
Tradition telecom technique( analog or data) e.g.TDM
Analog* to Digital -------------------------------> Digital ----------------------------------> Digital to Analog
*In the case of ISDN, voice is digital in the local loop as well.

Circuit Switched Network

Local Loop Local Loop



Varying Distances (a few miles or a few thousand)

Dedicated Circuit

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

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 bandwidth by PRI Tone package tends to smaller than data package

ISO VoIP model

digitized voice Call signaling/Call routing/User Authentication coding/framing


H.225/H.245 L4 - TCP



L3 - IP L2 L1

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


Coders challenge:
Need the more strong Coding ability the extra delay in transfers.

Coders (2)
G.711 G.723 G.728 G.729a G.729b

bandwidth applications
64K 5.3K, 6.4K 16K 8K 8K PSTN voice, MMCX remote access, ITS-SP, NetMeeting, [MMCX] H.320 ISDN videoconferencing VoIP, VTGW4.0, GW-1000 - send silence suppress silence

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.

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 less package 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 in subnets

The reaching time of tone package is different. The package unstable transmission delay will be taken pace 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 etc at 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

Jitter without buffer

1 2 3 Voice signal Packetized & Compressed Voice 1 2 3

Transport Through Packet Network

Voice Packets Received with Delay and Jitter

largest delay

Regenerated Voice signal without Jitter Buffering

Jitter with buffer or append

1 2 3


Jitter Buffering

Delay from
packet codec Network element Network transmission jitter (build out)

The appreciable condition of listener Easy to be apperceived when it exceed 250ms

fast PCs, Servers Special equipments (gateways, DSPs) Adjusting network (optimizing equip, engineered networks) Real time monitor network (QoS) - DiffServ, MPLS, RSVP...

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.

Network function module

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

Network function module (2)

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 package interchange network Not dealing tone transfer Deal PCs H.323 proxy services

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

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)

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

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 - gatekeeper tandems signaling between

endpoints / gateways

H.323 signal protocol

RAS - Registration, Admission and Status
Protocol used to by endpoints to register with a gatekeeper, find other H.323 endpoints, ask permission from the gatekeeper to make calls and provide status information between H.323 endpoints.

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.

call setup, features, RAS includes Q.931 (the ISDN mechanism )

H.323 information transfer protocol

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

Other protocols
H.323 v1 was first attempt at defining signaling for these networks
inefficiencies: signaling overhead and delay omissions: large systems, security, interoperability between vendors, interoperability between carriers

H.323 v2 and beyond are addressing the problems

implemented in ITS-SP R3 provides compatibility with standards and with future Lucent products

H.323 used for H.323 terminals (e.g., Internet Phone)

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) Transfer fax or compress tone RTP RTCP RAS

User data protocol (UDP)

Phone-to-phone and fax-to -fax

Network Manager
To Billing System To QoS Mgmt To Provisioning System

Gatekeeper Debit/Credit Calls

IP Network
Switch Fax Gateway Gateway Switch Fax



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 PIN and receive ID and PIN Inbound gateway contact its gatekeeper and apply for authenticating 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 its gatekeeper 6. Inbound gateways gatekeeper reference its INTERzone route list and decide which zone can dial out to destination phone number

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 connect to 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

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

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 How to explain the typing serials which do by caller
Capture by inbound GWs PSTN (one step dialing) Capture by QTGW (two step dialing)

Describers dialing format

The number format of terminal gateway

Gateway how to produce the dialing serial to outbound PSTN Gateway how to connect all kinds of PSTN switch PSTN including remote and international dialing VPN interior and PSTN dialing 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

Two kinds of number formats

Three formats (for outbound):

(1) User Dail 011551187654321

(2) Route Number 551187654321

Income GW (NYC)
Original GK

(3)Route Number 551187654321

(4)Route Number 551187654321

(5) Dest. GW Number *987654321


Dest. GK

Outbound GW (Sao Paulo)