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  • NOTAS AL FORMADOR Esta unidad require que el formador lea en detalle la guia de VoIP4D. La presentación tiene tres objetivos fundamentales (1) entender los conceptos básicos de una infrastructura VoIP y el significado de covergencia entre la red telefónica tradicional y la IP (2) presentar las ventajas de FOSS como plataforma para el desarrollo de este tipo de infraestructura (3) introducir algunos retos concretos que aparecen en el desarrollo de este tipo de tecnología en regiones en desarrollo. Finalmente, la presentación incluye un ejemplo de implementación que debe ser necesario localizado al gusto del formador. Por su caracter técnico el formador debe tener experiencia real implementando VoIP. Esta presentación no cubre la segunda parte de la guía, la parte práctica de configuración de escenarios. Sólo se incluyen 4-5 diapositivas donde se introducen dos conceptos básicos en Asterisk (canal y plan de marcado). El formador debe introducir la lógica de Asterisk en la presentación y discutir las configuraciones y comandos durante una posible sesion práctica. Esta presentación necesita de 60 minutos, en caso de una audiencia poco avanzada se pueden suprimir las diapositivas 53-58. Es importante dedicarle tiempo a las diapositivas 59-60 porque resumen todos los conceptos aprendidos. Las diapositivas 3,4 y 5 introducen la guía y la motivación de los autores al crearla, el nuevo formador las pueden eliminar y sustituir por su motivación particular. Al final de esta presentación la audiencia debe recordar: porque existe convergencia, cual son las ventajas de FOSS e Internet respecto a la telefonía tradicional (mobilidad, flexibilidad, etc), cuales son las diferencias fundamentales entre SIP y IAX2, cual es el tipo de equipamiento que se va a necesitar, que es un ATA, VoIP Phone, PSTN/RTB interface, que existe algo que se llama Asterisk :-) y su logica de funcionamiento... Por último se debe aclarar que para empezar a aprender solo se necesita 3 PCs y que todo se puede hacer por software.

Wimax and VoIP Presentation Presentation Transcript

  • 1. 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
  • 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 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)
  • 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 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
  • 8.
    • 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
  • 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 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
  • 11. A typical question
    • Why not use Skype, 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 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
  • 15. PBX advantages
    • Value-added services
      • Call Transfer
      • Three way calling
      • Voice mail
      • Interactive Voice Response (IVR) services
  • 16. 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
  • 17. What is Asterisk?
    • Created by Mark Spencer (Digium)
      • Based on previous work of Jim Dixon (Zapata Telephony Project)
    • Runs better under GNU/Linux
  • 18. 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 :-)
  • 19. 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
  • 20. 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
  • 21. Signalling in PSTN
    • 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 dial and 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 connected to 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, or ATA , acts as an FXS.
  • 28. FXO and FXS
      • A PBX can 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 and administrative signalling (SS7)
    • These information circuits do not have to use the same physical channel
  • 32. Signalling in IP telephony
    • Signalling and conversations are separated (as in the PSTN)
    • Each signalling mechanism represents a “cult”of 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 user was 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 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
  • 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 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)
  • 46. VoIP telephone
    • Thompson Speedtouch2030
    • Four IP lines
    • Web Interface
    • Hands-free
    • Price today: USD 125
  • 47. VoIP Telephones (WiFi)
    • Zyxel Prestige 2000W
    • WiFi VoIP
    • One of the first models
    • Price: USD 300 (2005)
  • 48. Softphones
    • PC installed software
    • Requirements
        • A sound card
        • A firewall that does not block VoIP
    • Price: 0 USD
  • 49. Softphones
    • X-lite
    • 0 USD
    • Download: http://www.xten.com
  • 50. Softphones
    • iaxComm, kiax
    • Supports IAX
    • 0 USD
    • Download:
    • http://iaxclient.sourceforge.net
  • 51. 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
  • 52. ATA
    • Digium IAXy
    • One of the first
    • (the first) with IAX2 support
    • Low power
    • consumption
    • Price: 95 USD (2005)
  • 53. ATA
    • Gateway IP- IAX IAD100 with one FXS
    • It integrates an ATA with support for IAX2 and an NAT
    • Price: 110 USD
  • 54. ATA
    • Sipura SPA-3000
    • A mini-PBX
    • Price: 170 USD
  • 55. 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)
  • 56. PSTN Interface
    • TDM400P
    • wildcard, 1FXO +1
    • FXS (Digium)
    • Price: 190 USD
  • 57. 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
  • 58. 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
  • 59. 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
  • 60. 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
  • 61. 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
  • 62. 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
  • 63. 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
  • 64. 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
  • 65. VoIP implementation challenges
    • Energy: There are no reliable sources of energy
    • Regulatory Framework:
    • Illegal Service
    • Need for licenses
  • 66. 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
  • 67. Recommendations
    • Energy: Low power consumption equipment, solar energy
    • Regulatory Framework: Lobby, Business Models
  • 68. PART 2 – Hands on
  • 69. The components
    • 1 PC with Asterisk
      • Any distribution of the Linux operating system
    • 2 VoIP telephones
      • Alternative: 2 PCs with 2 soft phones
  • 70. You can build a portable PBX
    • Mini-ITX board with Digium TDM400P Card
  • 71. First steps: Installing Asterisk
    • Download the code
    • Compile
    • Learn the basic commands
  • 72. Configuration files
    • Step 1:
    • Define and configure communication channels
    • Step 2:
    • Define “rules” for the extensions (create a dial plan)
  • 73.
    • 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
  • 74. 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)
  • 75. Communication channels (3)
    • Each type of channel technology is configured in a specific file (sip.conf, iax.conf, zapata.conf, etc.)
  • 76. 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
  • 77. 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
  • 78. 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)
  • 79. PARTE 3 – Case study
  • 80.
      • Implementation example
      • Fantsuam Foundation Wireless ISP,
      • Kafanchan, Nigeria
      • VoIP as a business model
  • 81. Kafanchan, Nigeria
    • Schools
    • NGOs
    • Health sector
    • Religious sector
    • Private sector
    • Individuals
  • 82. Kafanchan, Nigeria
  • 83. Network Backbone
  • 84. Sectors in service
  • 85. Network topology
  • 86. Inclusion of VoIP in the NOC
  • 87. Wireless hub equipment
  • 88. Client equipment + VoIP ATA
  • 89. 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.
  • 90.
    • http://voip4d.it46.se