voip

Slide : 01

Yesterday’s Network YCircuit Switched Networks (VOICE)

Packet Switched Networks (DATA)

Separated Networks Separated Applications/Services

PBX PBX

COCO

CO

Router Router

Router Router

Router

Slide : 02

Converged Network CPBX CO

Router

PBX

RouterRouter Router

Router

Converged Networks Separated or Integrated Applications

Slide : 03

W

Cost Savings One backbone instead of two parallel ones.

No maintenance of proprietary switching systems.

Significant capital equipement cost reduction.

Simplification One infrastructure

Multi-vendor capable

Advanced ServicesUnified Messaging

Computer-telephony integration

Why converge ?

Slide : 04

OutlineO

Introduction History What is VoIP Why VoIP VoIP architecture Components of Voip VoIP Protocols VoIP & QOS Test Application Conclusion

( H.323 & SIP )

Slide : 05

IntroductionI

Increase in Broadband new applications with

enormous benefits to the users.

The prominent example of broadband new services is the

developement of VoIP (Voice over Internet Protocol).

Slide : 06

HistoryH

By 1980 : availibility of transmitting voice conversations

over the internet to end-users.

By 1998 : some entrepreneurs started to market PC-to-

phone and phone-to-phone VoIP solutions.

By the end of 1998 : VoIP calls had yet to total 1% of all

voice calls.

By 2000 : VoIP calls accounted for 3% and by 2003 that

number had jumped up to 25%.

RFC : http://www.joehallock.com/index.php

WSlide : 07

What is VoIP?

VoIP is a technology optimized for the transmission of voice

through the Internet.

VoIP allows high quality tow-way voice transmission over

broadband connections.

VoIP systems carry telephony signals as digital audio.

Slide : 08

H How convert signals?

Filter Sampler Quantifier Encoder

IP Protocol

Slide : 09

W Why VoIP ?

Cost reduction Save up to 40 % on local calls, and up to 90 % on international calls.

Operational ImprovementCommon network infrastructure.Simplification of Routing Administration.

Business Tool IntegrationVoice mail,email and fax integration.Web + Call.Mobility using IP.

Telephony

Voice instant messaging

Teleconferencing

Slide : 10

V VOIP Architecture

Figure 1. Different architectures of VoIP

RTC

Internet

Modem ADSL

Free Box

Classic Phone

Classic Phone

Gateway Operator

RTC / Internet

PC with Softphone

PC with Softphone

PC to PCPC to PhonePhone to Phone

Slide : 11

C Components of VoIP ?

Coding & Decoding of Analog Voice Analog-to-Digital and Digital-to-Analog conversions, Compression.

SignalingCall setup & tear down, Ressource & coding negotiation.

Transport of Bearer TrafficVoice packet transmission,Routing,Support of quality of service.

Numbering Phone number, IP address.

W What Protocols are required ?

Signaling Protocol : To establish presence, locate user, set up,

modify and tear down sessions (Sip,H.323,MGCP…).

Media Transport Protocols : To transmit packetized

audio/video signal (RTP/RTCP).

Supporting Protocols : Gateway Location, QOS, AAA, Address

Translation, etc.

Slide : 12

V VoIP Protocol Stacks

Slide : 13

H.245 Q.931 RAS SDP/SIP MGCP RTCP RTP

TCP UDP

IP

Link Layer Protocols

VOICE & VIDEOH.323

H.225

Call Control & Signaling Signaling & Gateway control

Media

H.225 : Call Control Singnaling

H.245 : Call Channel Singnaling,media control

MGCP : Media Gateway Control Protocol

Q.931 : ISDN Singnaling

RAS : Registration , Admission , Status

RTCP : RTP Control Protocol

RTP : Real Time Transport Protocol SDP : Session Discription Protocol

SIP : Session Initiation Protocol

TCP : Transport Control Protocol

IP : Internet Protocol

UDP : User Datagram Protocol

O OSI & VoIP

Table 1. Relation between the OSI model and the protocols used for VoIP

Slide : 12

OSI Level VoIP protocols

7 Application NetMeeting / GnomeMeeting / Applications

6 Presentation Codecs

5 Session H.323 / MGCP / SIP

4 Transport RTP / TCP / UDP

3 Network IP

2 Data Link Frame Relay, ATM, Ethernet, PPP, MLP, and others

VoIP Using H.323

H H.323 Protocol (1/5)

ITU-T standard, latest version v4.

Peer-to-peer protocol that supports terminals communicating over packet based networks.

Different functions of H.323 run over either TCP or UDP.

Definition

Slide : 15


Architecture

Slide : 16

Figure 2. H.323 Network Architecture


Protocol stack

Slide : 17

Figure 3. H.323 protocol stack

IP

UDPTCP

Gatekeeper

RegistrationAdministration

Status(RAS)

RTCP

RTP

H.261H.263G.7xx

T.120H.245H.225

Layer 3

Layer 4

Layer 5

ControlA/V CntlVideo Audio DataControl


Slide : 18

H.323 protocol exchange

Figure 4. Establishing connection using the H.323 protocol

H.225 / RAS Admission request

H.225 / RAS Admission confirm

H.225 / Q.931 Setup

H.225 / Q.931 Call Proceeeding

H.225 / RAS Admission request

H.225 / RAS Admission confirm

H.225 / Q.931 Alerting

H.225 / Q.931 Connect

H.245 Terminal Capability Set

H.245 Terminal Capability Set

H.245 Terminal Capability Set Acknowledge


H.245 Open Logical Channel

H.245 Open Logical Channel



Terminal H.323Terminal H.323 GateKeeper

H H.323 Protocol (5/5)H.323 exchange

Slide : 19

Figure 5. Release of connection using the H.323 protocol

H.225 / Q.931 Disconnect

H.225 / Q.931 Release

H.225 / Q.931 Release Complet

H.225 / RAS Disconnect RequestH.225 / RAS Disconnect Request

H.225 / RAS Disconnect ConfirmH.225 / RAS Disconnect Confirm

Terminal H.323Terminal H.323GateKeeper

VoIP Using SIP

S SIP Protocol (1/6)

IETF standard , RFC 3261.

Peer-to-peer protocol for initiation, modification termination of communication sessions between users.

SIP is a protocol based on a client-server in text mode.

The simplicity, speed and lightness of use of SIP are all arguments that could allow SIP to convince investors.

Definition

Slide : 21

SSlide : 22

SIP Protocol (2/6)

Architecture

Slide : 23

S SIP Protocol (3/6)

Different SIP requests

Request DescriptionINVITE Invite the server to participate to a session.

OPTIONS Inquire capability and options.

BYE Terminate a session.

CANCEL Canccel any in-progress request.

ACK Accept th INVITE to participate, acknowledgement toINVITE.

REGISTER A client to register location information with a server.

S Slide : 24

SIP Protocol (4/6)

SIP protocol exchange

Figure 6. Establishment and release of connection using SIP

INVITE

100 TryingINVITE

100 Trying

180 Ringing180 Ringing

200 OK200 OK

Ack Ack

BYE BYE

200 OK200 OK

SIP PhoneA

SIP PhoneB

A CALL B

A Hang Up

B Pick Up

Comminication established

Server

Slide : 25

S

Benefits

Extremely simple protocol,

Flexible Protocol,

Simple to implement: messages written in clear,

An excellent interoperability ,

Very good possibility of mobility management,

Used for telephony 3G (UMTS).

SIP Protocol (5/6)

Slide : 26

S Disadvantages Based on IP addresses, clients behind NAT is unreachable directly,

Problems in the management of presence and instant messaging,

Low number of users : This protocol is still unknown but it tends to replace H.323,

Uses centralized Registrars so they can become overloaded and thus collapse.

SIP Protocol (6/6)

RTP & RTCP

Slide : 28

T Transport Protocol : RTP /RTCP

A session consists of an RTP/RTCP pair of channels,

Usually works over UDP/IP,

End-to-end protocol,

No QoS guarantees,

No guarantee of packet delivery,

suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services.

Slide : 29

Format Header RTP/RTCP

V=2 P X CC M content type

sequence number

TIME MARKER

Synchronization source identifier « SSRC »

Defined by the profile Length

DATA

Source identifier conributive « CSRC »

T Transport Protocol : RTP /RTCP

0 2 4 6 8 10 12 14 16 18 20 22 24 26 28 30

VoIP QOS

Slide : 31

P Problems of Quality of Service

Jitter variability in the

arrival times of the datagrams at the receiver.

Delay Network delay Accumulation

delay. Processing delay

Losing Packets

Datagrams that are lost generally can't be recovered, so they appear as momentary gaps in the conversation.

Slide : 32

QFive components defined by UIT-T G114 affect the voice quality :

QOS Considerations - Voice Quality

Codec used

Packet Loss - Voice can tolerate some packet loss(<0,001). - Use packet loss concealment to improve quality.Packet Transfer Delay

- End-to-end delay must be <150 milliseconds. - Use DiffServ and priority queuing for voice packet transport.

Jitter - Too much jitter degrades voice quality. - Use jitter buffer to reduce jitter.

Echo

- Different codecs use different

compression algorithms.

Slide : 33

B Depends on codec used, packetized delay and protocol overhead.

Packetized delay : Time required to collect voice in the packet Payload.

Payload length = (codec bit rate)* (packetization delay).

Packet overhead : RTP , UDP , IP.

Bandwith required : (overall length in bits)/(packetization delay).

Bandwith Required

Slide : 34

Q QOS criteria

Good Average Bad

Delay D < 150 ms 150 ms < D < 400 ms

Jitter G < 20 ms 20 ms < G < 50 ms

Packet Loss P < 1% 1% < P < 3% 3% < P

Test Application Implementation of Asterisk server

Slide : 36

I Implementation of Asterisk server

Stages of a SIP call

Asterisk Processus

1. connection request

2. C

heck

s si

p.co

nf

3. fo

und

the

user

4. Communication

1. We compose the Num

3

.Ret

urn

user

app

2.Ch

ecks

ext

en.c

onf

4. Routing call XLITE

XLITE

Slide : 37


Creating user accounts:

User 1

User 2

# vi /etc/asterisk/sip.conf

SIP.conf

Slide : 38


Registering extension:

#vi /etc/asterisk/extensions.conf

User 1

User 2

Extensions.conf

Slide : 39

C Conclusion

VoIP challenges :

Available Bandwidth,Network Latency,Packet Loss,Jitter ,Echo,Security,Reliability .

That is why the integration of voice over IP is just one step towards EoIP: Everything over IP

THANK UFOR YOUR ATTENTION

voip

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