VoIP for development Authors: Alberto Escudero-Pascual, Louise Berthilson (cc) Creative Commons Attribute Non-Commercial Share-Alike 2.5 Based on: VoIP-4D Primer Building voice infrastructure in developing regions Unit 16

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.

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.

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

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

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

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

VoIP guide for development (2) The document is available in four languages (en, es, fr, ar) Licenced under Creative Commons Non-Commercial Share-Alike Now included in the second Spanish edition of the book WNDW The chosen distribution channel is Internet Funded by IDRC (Acacia initiative)

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

Licenced under Creative Commons Non-Commercial Share-Alike

Now included in the second Spanish edition of the book WNDW

The chosen distribution channel is Internet

Funded by IDRC (Acacia initiative)

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

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

Evidence of VoIP explosion Telecommunications deregulation allowed the 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 million downloads , November 2005 >5 million simultaneous users, January 2006

Telecommunications deregulation allowed the 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 million downloads , November 2005

>5 million simultaneous users, January 2006

The traditional suppliers buy “data” companies. IP divisions are created. Tradtional telecom services suppliers Siemens, Alcatel, Ericsson IP equipment suppliers Cisco , 3Com, Nortel Networks VoIP services appear http://www.pulver.com/products/sip/ Convergence

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

Tradtional telecom services suppliers

Siemens, Alcatel, Ericsson

IP equipment suppliers

Cisco , 3Com, Nortel Networks

VoIP services appear

http://www.pulver.com/products/sip/

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

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

Our magic potion We have access to both software and hardware that allow us to exchange calls We have access to an open and public network (Internet) We are able to adapt and modify technology to meet our needs

We have access to both software and hardware that allow us to exchange calls

We have access to an open and public network (Internet)

We are able to adapt and modify technology to meet our needs

A typical question Why not use Skype, or Google Talk?

Why not use Skype, or Google Talk?

The short answer Flexibility Appropriation Opportunity Sustainability

Flexibility

Appropriation

Opportunity

Sustainability

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

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

What is a PBX? P rivate (Automatic) B ranch E 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

P rivate (Automatic) B ranch E 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

PBX advantages Value-added services Call Transfer Three way calling Voice mail Interactive Voice Response (IVR) services

Value-added services

Call Transfer

Three way calling

Voice mail

Interactive Voice Response (IVR) services

What is Asterisk? A free implementation of a telephonic switch (Central office or exchange) It allows associated “phones” to establish calls among them and connect to any other telephone subnet

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

It allows associated “phones” to establish calls among them and connect to any other telephone subnet

What is Asterisk? Created by Mark Spencer (Digium) Based on previous work of Jim Dixon (Zapata Telephony Project) Runs better under GNU/Linux

Created by Mark Spencer (Digium)

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

Runs better under GNU/Linux

PSTN P ublic S witched T elephone N 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 :-)

P ublic S witched T elephone N 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 :-)

PSTN vs. Internet Flow of information Channel vs. individual datagrams Data processing Inside the netwok vs the edges Standards setting organizations ITU vs. IETF Routing mechanisms Telephone numbers vs. IP addresses

Flow of information

Channel vs. individual datagrams

Data processing

Inside the netwok vs the edges

Standards setting organizations

ITU vs. IETF

Routing mechanisms

Telephone numbers vs. IP addresses

Signalling in traditional telephony “ signalling” and “data” are separated into different channels signalling: Is responsible for the establishment and status of the “call” Is used in coordination with the billing systems

“ signalling” and “data” are separated into different channels

signalling:

Is responsible for the establishment and status of the “call”

Is used in coordination with the billing systems

Signalling in PSTN PBXs are the PSTN “routers”. Two components according to the role FXO = Foreign Exchange Office FXS = Foreign Exchange Station

PBXs are the PSTN “routers”.

Two components according to the role

FXO = Foreign Exchange Office

FXS = Foreign Exchange Station

Foreign Exchange Office (FXO) Any device behaving as a “telephone” Accepts signalling on-hook/off-hook busy Starts and receives phone calls

Any device behaving as a “telephone”

Accepts signalling

on-hook/off-hook

busy

Starts and receives phone calls

Foreign Exchange Station (FXS) Generates dial and ring tones. In analogue lines: Generates calling pulses Provides DC voltage to telephone terminals

Generates dial and ring tones.

In analogue lines:

Generates calling pulses

Provides DC voltage to telephone terminals

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)

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)

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

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

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

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

FXO and FXS An Analogue Telephony Adapter, or ATA , acts as an FXS.

An Analogue Telephony Adapter, or ATA , acts as an FXS.

FXO and FXS A PBX can be fitted with either FXS or FXO interfaces

A PBX can be fitted with either FXS or FXO interfaces

Analogue signalling The signals transmitted between FXS and FXO are: Dial and busy tones Ring tone On-hook and off-hook

The signals transmitted between FXS and FXO are:

Dial and busy tones

Ring tone

On-hook and off-hook

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

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

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 and administrative signalling (SS7) These information circuits do not have to use the same physical channel

In the PSTN, voice and data are separated

One “circuit” is for the voice (the conversation)

A second “circuit” is for supervisory and administrative signalling (SS7)

These information circuits do not have to use the same physical channel

Signalling in IP telephony Signalling and conversations are separated (as in the PSTN) Each signalling mechanism represents a “cult”of followers

Signalling and conversations are separated (as in the PSTN)

Each signalling mechanism represents a “cult”of followers

Signalling in IP telephony Dozens of protocols and their cults: H.323 (Telco) SIP (Internet) Session Initiation Protocol IAX2 (Community) Inter- Asterisk eXchange

Dozens of protocols and their cults:

H.323 (Telco)

SIP (Internet) Session Initiation Protocol

IAX2 (Community) Inter- Asterisk eXchange

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

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

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 user was initially registered ) IP telephony allows to physically move the phone numbers

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 user was initially registered )

IP telephony allows to physically move the phone numbers

SIP proxy servers

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

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

RTP y NAT Pros NATs are easy to implement They connect machines without requiring more network resources Great acceptance and products

Pros

NATs are easy to implement

They connect machines without requiring more network resources

Great acceptance and products

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

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

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

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

Why is IAX2 better than SIP? It minimizes the bandwidth used per call 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

It minimizes the bandwidth used per call

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

VoIP Equipment The base PBX The terminals VoIP telephones Soft phones Analogue Telephone Adaptors (ATA) Connection to PSTN PSTN interface cards

The base

PBX

The terminals

VoIP telephones

Soft phones

Analogue Telephone Adaptors (ATA)

Connection to PSTN

PSTN interface cards

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)

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)

PBX

VoIP telephone Dedicated VoIP equipment When buying a VoIP phone 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: <$ 50 (2008)

Dedicated VoIP equipment

When buying a VoIP phone 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: <$ 50 (2008)

VoIP telephone Thompson Speedtouch2030 Four IP lines Web Interface Hands-free Price today: USD 125

Thompson Speedtouch2030

Four IP lines

Web Interface

Hands-free

Price today: USD 125

VoIP Telephones (WiFi) Zyxel Prestige 2000W WiFi VoIP One of the first models Price: USD 300 (2005)

Zyxel Prestige 2000W

WiFi VoIP

One of the first models

Price: USD 300 (2005)

Softphones PC installed software Requirements A sound card A firewall that does not block VoIP Price: 0 USD

PC installed software

Requirements

A sound card

A firewall that does not block VoIP

Price: 0 USD

Softphones X-lite 0 USD Download: http://www.xten.com

X-lite

0 USD

Download: http://www.xten.com

Softphones iaxComm, kiax Supports IAX 0 USD Download: http://iaxclient.sourceforge.net

iaxComm, kiax

Supports IAX

0 USD

Download:

http://iaxclient.sourceforge.net

Analogue Telephone Adaptor (ATA) Connects an analogue telephone to a VoIP network It has an RJ-11 (phone jack) and an RJ-45 (Ethernet jack) An ATA is an FXS adaptor It “talks” analogue with the phone While “talking” digital with the VoIP network

Connects an analogue telephone to a VoIP network

It has an RJ-11 (phone jack) and an RJ-45 (Ethernet jack)

An ATA is an FXS adaptor

It “talks” analogue with the phone

While “talking” digital with the VoIP network

ATA Digium IAXy One of the first (the first) with IAX2 support Low power consumption Price: 95 USD (2005)

Digium IAXy

One of the first

(the first) with IAX2 support

Low power

consumption

Price: 95 USD (2005)

ATA Gateway IP- IAX IAD100 with one FXS It integrates an ATA with support for IAX2 and an NAT Price: 110 USD

Gateway IP- IAX IAD100 with one FXS

It integrates an ATA with support for IAX2 and an NAT

Price: 110 USD

ATA Sipura SPA-3000 A mini-PBX Price: 170 USD

Sipura SPA-3000

A mini-PBX

Price: 170 USD

PSTN interface cards Needed to connect our VoIP network to the PSTN It can incorporate FXO and FXS modules These cards traditionally used Digital Signal processors (DSP) Current trend is to move the intelligence to the CPU, (à la WinModem)

Needed to connect our VoIP network to the PSTN

It can incorporate FXO and FXS modules

These cards traditionally used Digital Signal processors (DSP)

Current trend is to move the intelligence to the CPU, (à la WinModem)

PSTN Interface TDM400P wildcard, 1FXO +1 FXS (Digium) Price: 190 USD

TDM400P

wildcard, 1FXO +1

FXS (Digium)

Price: 190 USD

Other aspects Quality of Service The ability of a network to provide better service to certain network traffic Optimize the available bandwidth ( codec ) Control jitter Minimize latency

The ability of a network to provide better service to certain network traffic

Optimize the available bandwidth ( codec )

Control jitter

Minimize latency

Codecs Coder/decoder It is employed to digitize voice into data and viceversa Higher compression leads to greater distortion One codec is better than another if it provides better voice quality using less bandwidth

Coder/decoder

It is employed to digitize voice into data and viceversa

Higher compression leads to greater distortion

One codec is better than another if it provides better voice quality using less bandwidth

Codecs PSTN normally uses PCM (Pulse Code Modulation),a codec that needs 64 kbps Two very common PCM standards are: µ-law (G711µ), USA/Canada/Japan A- law (G711a), rest of the world Since G711 does not have large processing requirements it is available in almost all equipments

PSTN normally uses PCM (Pulse Code Modulation),a codec that needs 64 kbps

Two very common PCM standards are:

µ-law (G711µ), USA/Canada/Japan

A- law (G711a), rest of the world

Since G711 does not have large processing requirements it is available in almost all equipments

Codecs G.711 codec is not appropriate in developing countries because it needs too much bandwidth We must use other codecs that use less bandwidth such as GSM or Speex G.729 is a good codec, but it has the disadvantage of requiring a license

G.711 codec is not appropriate in developing countries because it needs too much bandwidth

We must use other codecs that use less bandwidth such as GSM or Speex

G.729 is a good codec, but it has the disadvantage of requiring a license

Jitter Variation in the arriving time “between” packets Due to network congestion,route changes, or clock drift A jitter buffer can help alleviate this problem at the cost of additional latency

Variation in the arriving time “between” packets

Due to network congestion,route changes, or clock drift

A jitter buffer can help alleviate this problem at the cost of additional latency

Latency (delay) The time it takes for a packet of data (datagram) to get from one designated point to another In VoIP networks we must try to minimize the latency by giving priority to the voice traffic

The time it takes for a packet of data (datagram) to get from one designated point to another

In VoIP networks we must try to minimize the latency by giving priority to the voice traffic

Latency Latency cannot be reduced below the propagation time of the signal. In satellite links ~ 300 ms Always install your PBX where your network is less congested

Latency cannot be reduced below the propagation time of the signal. In satellite links ~ 300 ms

Always install your PBX where your network is less congested

VoIP implementation challenges Technical: Avoiding the negative impact of NAT Supporting infrastructure: Wireless network with great latency and jitter Networks that have not been designed to prioritize real time services

Technical: Avoiding the negative impact of NAT

Supporting infrastructure:

Wireless network with great latency and jitter

Networks that have not been designed to prioritize real time services

VoIP implementation challenges Energy: There are no reliable sources of energy Regulatory Framework: Illegal Service Need for licenses

Energy: There are no reliable sources of energy

Regulatory Framework:

Illegal Service

Need for licenses

Recommendations Technical: Use IAX2 Supporting infrastructure: Wireless networks: Use the 5 GHz band (IEEE 802.11a). Use protocols with TDMA in urban areas (WiMAX) Incorporate QoS

Technical: Use IAX2

Supporting infrastructure:

Wireless networks: Use the 5 GHz band (IEEE 802.11a). Use protocols with TDMA in urban areas (WiMAX)

Incorporate QoS

Recommendations Energy: Low power consumption equipment, solar energy Regulatory Framework: Lobby, Business Models

Energy: Low power consumption equipment, solar energy

Regulatory Framework: Lobby, Business Models

PART 2 – Hands on

The components 1 PC with Asterisk Any distribution of the Linux operating system 2 VoIP telephones Alternative: 2 PCs with 2 soft phones

1 PC with Asterisk

Any distribution of the Linux operating system

2 VoIP telephones

Alternative: 2 PCs with 2 soft phones

You can build a portable PBX Mini-ITX board with Digium TDM400P Card

Mini-ITX board with Digium TDM400P Card

First steps: Installing Asterisk Download the code Compile Learn the basic commands

Download the code

Compile

Learn the basic commands

Configuration files Step 1: Define and configure communication channels Step 2: Define “rules” for the extensions (create a dial plan)

Step 1:

Define and configure communication channels

Step 2:

Define “rules” for the extensions (create a dial plan)

A communication channel in Asterisk is like a virtual phone wire The channels are your virtual PBX wires On the Internet you can have more than one conversation on the same physical channel In this section, you need to define the type of channels (SIP, IAX , connection to the PSTN, etc.) Communication channels

A communication channel in Asterisk is like a virtual phone wire

The channels are your virtual PBX wires

On the Internet you can have more than one conversation on the same physical channel

In this section, you need to define the type of channels (SIP, IAX , connection to the PSTN, etc.)

Communication channels (2) Asterisk lets you interconnect devices that use different protocols You can connect devices with IP support (VoIP Phones, ATA, Softphones) with other digital and analogue devices (PSTN, ISDN PRI / BRI)

Asterisk lets you interconnect devices that use different protocols

You can connect devices with IP support (VoIP Phones, ATA, Softphones) with other digital and analogue devices (PSTN, ISDN PRI / BRI)

Communication channels (3) Each type of channel technology is configured in a specific file (sip.conf, iax.conf, zapata.conf, etc.)

Each type of channel technology is configured in a specific file (sip.conf, iax.conf, zapata.conf, etc.)

Define extensions rules (create your dial plan) All incoming and outgoing calls use the channels that you have previously defined The dial rules indicate how channels interact

All incoming and outgoing calls use the channels that you have previously defined

The dial rules indicate how channels interact

Define extension rules (create your dial plan) This &quot;intelligent&quot; aspect of routing between calls is specified in the extensions file (extensions.conf) The extensions file is known as the dial plan

This &quot;intelligent&quot; aspect of routing between calls is specified in the extensions file (extensions.conf)

The extensions file is known as the dial plan

Asterisk follows the old telephony operator's logic To make a call (via two communication channels) We contact the operator (PBX) We indicate the person we want to contact The operator makes the connection depending on the type of line (extensions.conf file)

To make a call (via two communication channels)

We contact the operator (PBX)

We indicate the person we want to contact

The operator makes the connection depending on the type of line (extensions.conf file)

PARTE 3 – Case study

Implementation example Fantsuam Foundation Wireless ISP, Kafanchan, Nigeria VoIP as a business model

Implementation example

Fantsuam Foundation Wireless ISP,

Kafanchan, Nigeria

VoIP as a business model

Kafanchan, Nigeria Schools NGOs Health sector Religious sector Private sector Individuals

Schools

NGOs

Health sector

Religious sector

Private sector

Individuals

Kafanchan, Nigeria

Network Backbone

Sectors in service

Network topology

Inclusion of VoIP in the NOC

Wireless hub equipment

Client equipment + VoIP ATA

Conclusions The convergence of telephony is unstoppable. The future is moving toward an integrated IP network Open standards, the Internet and FOSS (Asterisk), allow us to implement VoIP networks The developing regions have to deal with very specific local challenges: access to bandwidth, energy and an apropriate network infrastructure.

The convergence of telephony is unstoppable. The future is moving toward an integrated IP network

Open standards, the Internet and FOSS (Asterisk), allow us to implement VoIP networks

The developing regions have to deal with very specific local challenges: access to bandwidth, energy and an apropriate network infrastructure.

http://voip4d.it46.se

http://voip4d.it46.se