VoIP 4D. Building Voice Infrastructure in Developing Regions. IT +46 May 2008

Transcription

1 VoIP 4D Building Voice Infrastructure in Developing Regions IT +46 1

2 Agenda Part 1: VoIP in Developing Regions Part 2: Introduction to VoIP Challenges and Opportunities Building blocks and Hardware Part 3: VoIP Scenarios 2

3 Part 1: VoIP in Developing Regions Why VoIP and not conventional voice infrastructure? 3

4 Challenges for VoIP in developing regions Technical Private IP addresses only NATs blocks conversations Infrastructure VSAT: Delay (latency) and jitter Wireless: No frequency planning in urban areas Networks are not engineered for VoIP 4

5 Challenges for VoIP in developing regions Power No reliable source of power Regulations Ilegal service Licence needed 5

6 Solutions Technical: Use IAX2 signalling protocol Infrastructure: Wireless: Move to 5.4 Ghz (IEEE a) TDMA in urban areas (WiMAX, UMTS TDD) Add QoS Power: Low power, Solar powered systems Regulations: Lobby, Business models 6

7 Part 2: Introduction to VoIP What is so special about VoIP? 7

8 1. VoIP The magic potion Carries telephone conversations as IP packets 2. Open standards Allows everyone to implement interoperable communication systems. 3. Free and open source software Learn from existing experiences, integrate solutions and share our results with others 8

9 Our magic potion We have access to the software and hardware to exchange phone calls We have an open and public network to exchange the calls (the Internet) We are able to adapt and modify all these elements to meet our own needs 9

10 The typical question Why not just rely on Skype, or Google Talk? 10

11 The short answer Flexibility Ownership Opportunity Sustainability 11

12 The recipe of VoIP PBX (the core) PSTN (the gateway to the old POTS) Signalling in traditional telephony and/vs IP telephony VoIP hardware (the terminals) Quality of Service (the service) Codecs, Latency and Jitter 12

13 What is a PBX? Private (Automatic) Branch Exchange What? Allows to share telephone lines with multiple users Routes incoming and outgoing phone calls Who? The owner of the system manages the unit Decides how to share the external phone lines with its users 13

14 Other advantages of a PBX Value added services Transferring calls Three way calling Voice mail (or voice to e mail) Interactive Voice Response (IVR) services 14

15 What is Asterisk? Free and open source implementation of a telephone exchange Allows a number of attached phones to make calls to one another and to connect to the global telephony network Originally created by Mark Spencer (Digium) Best supported in GNU/Linux 15

16 PSTN Public Switched Telephone Network The network of the world's public circuit switched telephone networks Amalgamation of all circuit switched telephone networks in the world The toll to the legacy system 16

17 PSTN vs Internet Information flow Processing of data Inside the network vs end points Standardization bodies Channel vs. single data packet ITU vs IETF standard Addressing scheme for routing data Telephone numbers vs IP addresses 17

18 Signalling in Telephony Signalling and Data are separated channels Responsible of the establishment of the call Indicates to the parties the status of the call Closely related to billing practices The result of several legacy system old traditional telephony (RING!) ISDN (2A+B) GSM (SMS) 18

19 Signalling in PSTN Telephone exchanges are the routers of the PSTN. Two main components depending on their role FXO = Foreign Exchange Office FXS = Foreign Exchange Station 19

20 Foreign Exchange Office (FXO) Any device that acts like a regular telephone Accept signals ring on hook and off hook busy Send and receive voice calls 20

21 Foreign Exchange Station (FXS) Delivers dial tone and ring tone to the phones In analogue lines, an FXS supplies the ring tone the power to the phones to work 21

22 Two simple rules An FXS needs to be connected to an FXO Like a telephone line needs to be connected to a phone An FXS supplies power (active) to the phone FXO (passive) 22

23 FXO and FXS together with a PBX A PBX that integrates FXO and FXS interfaces can connect to the PSTN or host telephones Telephone lines coming from an operator need to be connected to an FXO interface Telephone(s) in your office must be connected to the FXS interface(s) of the PBX 23

24 FXO and FXS 24

25 SIP An Internet protocol developed by IETF Responsible for setting up calls and signalling Dealing with authentication Negotiating the quality of the phone call Handling the IP addresses and port numbers to use when sending the voice conversations 25

26 Number Mobility SIP Proxy servers facilitatates the establishment of a phone call acts as an intermediary (knows where a number is located registrated) Internet telephony allows physically move your phone number 26

27 SIP proxy servers 27

28 Voice calls and NATs SIP voice calls are carried by RTP (Real time Transport Protocol Network Address Translators (NATs) are the big enemies of RTP A NAT network consists of several computers that share one public IP address with the external world. 28

29 RTP and NAT Pros NATs are useful to connect many computers to the Internet They are simple to setup Thousands of vendors 29

30 Cons RTP and NAT Limitations in routability Several problems related to NAT and VoIP one way audio Public addresses are rare specially in developing regions 30

31 IAX2 Created as part of the open source PBX Asterisk Uses one single data stream to communicate between the end points (SIP uses two streams) A friendly protocol for scenarios with NATs Allows multiple calls to be merged in a single set of IP packets (bandwidth saving) Trunking 31

32 Why IAX2 is better than SIP? 1) Minimizes the use of bandwidth per single call 2) Provides native support for NATs Easier to use behind firewalls 3) Minimizes the use of bandwidth for several calls, by trunking 32

33 VoIP Hardware 33

34 PBX Do it yourself (2005 Open Hardware (2007) 34

35 Do It Yourself [2005] Motherboard: VIA Mini-ITX Epia M10000 Chassi: Morex Mini-ITX Chassi Cubid 2688 Harddrive: 40 GB IDE UDMA133 Memory: 512 MB DDR PC MHz 35

36 PSTN interface card What: TDM400P wildcard, 1FXO +1 FXS (Digium) ~ 190 USD Many

37 What we have seen since 2005? Specific VoIP distributions with Graphical Interfaces The use of virtualization to test and evaluate new technology The first asterisk-based appliances Open hardware Many

38 IP04 38

39 VoIP phones Dedicated hardware for VoIP To consider when buying a VoIP phone Support for bandwidth) high compression Administration interface Audio interface codecs (low Price today: USD 39

40 VoIP phones What: Thompson Speedtouch 2030 Four IP lines Web interface Hands free Price today: 125 USD 40

41 VoIP phones What: Zyxel Prestige 2000W WiFi VoIP One of the first models How much: 300 USD (2005) 41

42 Softphones Software installed in a PC Requirements: A working sound card No firewall blocking the application Price range: 0 USD 42

43 Softphones What: X lite How much: 0 USD Download: 43

44 Softphones What: iaxcomm How much: 0 USD Download: 44

45 Analogue Telephone Adaptor (ATA) Connects a standard analogue telephone to a VoIP network Equipped with an RJ 11 (phone jack) and an RJ 45 (Ethernet) jack An ATA acts like an FXS adapter Talking analogue with the phone Talking digital with the VoIP network 45

46 ATA What: Digium IAXy Proof of concept (first IAX2 ATA) Low power How much: 95 USD (2005) 46

47 PSTN interface card Needed to route calls from a VoIP terminal to the PSTN Receives analog/digital calls from PSTN and route them into the IP network FXO and FXS slots Traditionally with expensive DSP chips Moving the intelligence to the PC CPU 47

48 PSTN interface card What: TDM400P wildcard, 1FXO +1 FXS (Digium) How much: 190 USD 48

49 Quality of Service The ability of a network to provide improved (better) service to selected network traffic Optimized the available bandwidth (codec) Control jitter Minimize latency 49

50 Codecs Compressor/de compressor algorithm Used to digitize voice into data or convert digitized voice back to an analogue signal Higher compression > more distortion One codec is better than another if it provides better voice quality using similar amount of bandwidth 50

51 Codecs PSTN normally uses PCM (Pulse Code Modulation) PCM: a high quality codec that requires 64 kbps Two common PCM compression standards: Micro law (G711u), North America A law (G711a), Europe G711 does not require much processing power and is available in most VoIP appliances. 51

52 Codecs In developing countries, G.711 requires too much bandwidth Good and free codecs that use low bandwidth are GSM and Speex G.729 is another good codec but it requires a licence for commercial usage 52

53 Jitter The variation in time between packets arriving Caused by network congestion, timing drift, or route changes. Jitter buffer can handle jitter and decrease its negative effects Deliberately delays the playback of the conversation, thus ensuring that the lazy packets have arrived 53

54 Latency (delay) How much time it takes for a packet of data to get from one designated point to another Minimize the delay by giving priority to the voice traffic Satellite communication: latency of ~300 ms Place your PBX in the least congested or saturated part of the network 54

55 Part 3: VoIP Scenarios When, where and how? 55

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59 References [1] Vonage, About Vonage, [2] Jaanus, 5 million online Skypers, in News, Events, Milestones, Skype, skypers.html 59