1. VoIP for development Authors: Alberto Escudero-Pascual, Louise Berthilson (cc) Creative CommonsAttribute Non-Commercial Share-Alike 2.5 Based on: VoIP-4D Primer Building voice infrastructure in developing regions Unit16

2. Objective

To understand the basic concepts related to VoIP.

To introduce the benefits of Asterisk and software based solutions in implementing VoIP networks.

To present the great challenges in developing regions

To present a practical case study of introducing VoIP services.

3. Motivation

When living in Tanzania in 2004

Two big challenges:

• Technical knowledge is not available in the local languages

• The absence of low-cost IP infrastructure (voice and data)

The proprietary solutions were not flexible enough

4. VoIP guide for development

40 pages of introduction to VoIP

"Do it yourself" approach

Pedagogical approach vs. a list of commands

The guide wants to serve both the technical and general public

Aimed at developing regions and their specific problems

5. VoIP guide for development (2)

The document is available in four languages (en, es, fr, ar)

Licenced underCreative Commons Non-Commercial Share-Alike

Now included in the second Spanish edition of the bookWNDW

The chosen distribution channel is Internet

Funded byIDRC (Acacia initiative)

6. Table of contents

PART 1

• Introduction to VoIP

• VoIP basic foundations

• Equipment, hardware

PART 2

• How can I create my PBX (more information in the guide and practical section)

PART 3

• A case study

7. Evidence of VoIP explosion

Telecommunications deregulationallowedthe emergence of new operators:

• MCI (www.mci.com)

• Qwest (www.qwest.com)

• Level3 (www.level3.net)

• Vonage ( www.vonage.com )

• • >42 million lines in service, March 2006

• Skype ( www.skype.com )

• • 200 milliondownloads , November 2005

• • >5 million simultaneous users, January 2006

8.

The traditional suppliers buy data companies. IP divisions are created.

• Tradtionaltelecomservices suppliers

• • • Siemens, Alcatel, Ericsson

• IP equipment suppliers

• • • Cisco , 3Com, Nortel Networks

VoIP services appear

• http://www.pulver.com/products/sip/

Convergence 9. The magic potion

VoIP

• Carrying telephone conversations as IP packets

Open standards

• Allow everyone to implement compatible communication systems: interoperability

Free and open source software

• Learn from existing experiences and share our results

10. Our magic potion

We have access to bothsoftwareandhardwarethat allow us to exchange calls

We have access to anopenandpublicnetwork (Internet)

We are able toadaptandmodifytechnology to meet our needs

11. A typical question

Why not useSkype, or Google Talk?

12. The short answer

Flexibility

Appropriation

Opportunity

Sustainability

13. The recipe in detail (Contents)

PBX(the base)

PSTN (P ublic Switched Telephone Network )

• • Comparison between IP and PSTN signalling

VoIP equipment ( the terminals )

Quality of Service

• • Codecs, Latency and Jitter

14. What is aPBX?

P rivate (Automatic)B ranchE x change.

Definition for the layperson:

• It allows sharing one or more telephone lines with multiple users

• Routing of incoming and outgoing calls

• The (personal) owner of the system takes routing decisions and decides how to share the external phone lines with the users

15. PBX advantages

Value-added services

• Call Transfer

• Three way calling

• Voice mail

• Interactive Voice Response (IVR) services

16. What isAsterisk?

A free implementation of a telephonic switch (Central office or exchange)

It allows associated phones to establish calls among them and connect to any othertelephone subnet

17. What isAsterisk?

Created byMark Spencer(Digium)

• Based on previous work of Jim Dixon (Zapata Telephony Project)

Runs betterunder GNU/Linux

18. PSTN

P ublicS witchedT elephoneN etwork

• The global network of circuit-switched telephones

• The amalgamation of all circuit-switched telephone subnets in the world

• The network that will become obsolete :-)

19. PSTN vs. Internet

Flow of information

• Channel vs. individual datagrams

Data processing

• Inside the netwok vsthe edges

Standards setting organizations

• ITUvs. IETF

Routing mechanisms

• Telephone numbers vs. IP addresses

20. Signalling in traditional telephony

signalling anddata are separated into different channels

signalling:

• Is responsible for the establishment and status of the call

• Is used in coordination with the billing systems

21. Signalling inPSTN

PBXs are the PSTN routers.

Two components according to the role

• FXO = Foreign Exchange Office

• FXS = Foreign Exchange Station

22. Foreign Exchange Office (FXO)

Any device behaving as a telephone

Accepts signalling

• on-hook/off-hook

• busy

Starts and receives phone calls

23. Foreign Exchange Station (FXS)

Generates dialand ring tones.

In analogue lines:

• Generates calling pulses

• Provides DC voltage to telephone terminals

24. Do not forget...

An FXS connects to an FXO and viceversa

• • In the same way as a phone line (FXS)connects to a phone (FXS)

An FXS is an active element that feeds a passive element (FXO)

25. FXO, FXS in a PBX

The PBXs that have an FXO and an FXS can connect to the PSTN and to terminals

The telephone lines coming from the operator must be connectedto the FXO interface of the PBX

Your office phones must be connected to the FXS interfaces of the PBX

26. FXO and FXS

An analogue phone is an FXO device connected to a telephone line (PSTN) acting as an FXS

27. FXO and FXS

• An Analogue Telephony Adapter, orATA ,acts as an FXS.

28. FXO and FXS

• APBXcan be fitted with either FXS or FXO interfaces

29. Analogue signalling

The signals transmitted between FXS and FXO are:

Dial and busy tones

Ring tone

On-hook and off-hook

30. Analogue signalling (2)

Signalling methods vary from place to place

Two of the most common methods are loop start and ground start

The PSTN(AT&T, ITU),traditionally uses SS7

31. Analogue signalling (3)

In the PSTN, voice and data are separated

• One circuit is for the voice (the conversation)

• A second circuit is for supervisory andadministrative signalling (SS7)

These information circuitsdo not have to use thesame physical channel

32. Signalling in IP telephony

Signalling and conversations are separated(as in the PSTN)

Each signalling mechanism represents a cultof followers

33. Signalling in IP telephony

Dozens of protocols and their cults:

• H.323(Telco)

• SIP (Internet) Session Initiation Protocol

• IAX2 (Community) Inter- Asterisk eXchange

34. SIP

A protocol developed by IETF

Responsible for:

• Setting up the calls and other signalling tasks

• Authentication

• Negotiating the quality of the phone call

• Handling the port numbers and IP addresses involved in voice flow

35. SIP and mobility

SIP Proxy servers

• facilitate the establishment of phone calls

• acts as an intermediary that knows how to find a certain phone number in the network(where the userwas initially registered )

IP telephony allows to physically move the phone numbers

36. SIP proxy servers 37. Phone calls and NATs

Calls (voice) are transmitted using a protocol called RTP (Real-time Transport Protocol

In a network with a Network Address Translator (NAT) a set of machines share a routable IP address

The NATs are the big enemies of RTP

38. RTP y NAT

Pros

• NATs are easy to implement

• They connect machines without requiring more network resources

• Great acceptance and products

39. RTP and NAT(2)

Cons

• Limitations on the real traffic routing

• It is difficult to create services within a NAT

• They create "audio" problems with VoIP networks (e.g.: listening only to the party within the NAT who initiates the call)

• Unfortunately public IP addresses are a scarce resource in developing regions

40. IAX2

Created as part of the development of the PBX Asterisk

It uses a bidirectional flow to send the voice (SIP uses two independent flows)

It works much better (always) in the presence of NATs

It allows merging conversations taking place at the same time, thus saving bandwidth. Trunking

41. Why isIAX2 better than SIP?

It minimizes thebandwidth used percall

It incorporates native support of NATs and it is easier to integrate with firewalls

It further minimizes the use of bandwidth when making many simultaneous calls

42. VoIP Equipment

The base

• PBX

The terminals

• VoIP telephones

• Soft phones

• Analogue Telephone Adaptors (ATA)

Connection to PSTN

• PSTN interface cards

43. PBX

Components:

Motherboard: VIA Mini-ITX Epia M10000

Chassis: Morex Mini-ITX Chassis Cubid 2688

Hard drive: 40 GB IDE UDMA133

Memory: 512 MB DDR PC3200 400MHz

Today price: 1000 USD

Expected: 100-150 USD (IP04, 2008)

44. PBX 45. VoIP telephone

Dedicated VoIP equipment

When buying a VoIPphone do not forget:

• • • 1) Support for high-compression codecs

• • • 2) A good administrator interface

• • • 3) A good audio output

• • Price today: USD 100-120

• • Expected: