1. www.bzupages.com Voice Over IP 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

2. www.bzupages.com Group members Muhammad Aatif Aneeq BSIT07-15 Shah Rukh BSIT07-22 Muhammad Wasif Laeeq BSIT07-01 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

3. www.bzupages.com MUHAMMAD AATIF ANEEQ BSIT07-15 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

4. www.bzupages.com Circuit Switched Network:  In circuit switched networks, a circuit is established when data is needed to be transferred & all the communication is done through that circuit. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

5. www.bzupages.com Packet Switched Network  It is a switching network, in which data is broken down in small chunks (Packets) and is transferred in form of packets. This data may reach to the destination from different paths.  Each packet finds its way using the information it carries, such as the source and destination IP addresses. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

6. www.bzupages.com PSTN  The public switched telephone network is the network of the world's public circuit-switched telephone networks.  Originally a network of fixed-line analog telephone systems  Example: PTCL Landline 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

7. www.bzupages.com PSTN  PSTN lines come in two common standards  Analog  Single Dedicated line  Digital  Multiple lines in one line e.g. T1, E1 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

8. www.bzupages.com VoIP  Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks.  IP telephony  Internet telephony  voice over broadband (VoBB)  broadband telephony  broadband phone 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

9. www.bzupages.com VOIP 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

10. www.bzupages.com codec  compressor-decompressor  coder-decoder  Voip will not be possible without compression/decompression.  Voice first encoded from Analog to digital IP Packets and then decoded back to analog at receiver end. Department of IT, Institute of Computing, BZU, Multan

11. www.bzupages.com Choice of codec Depends on requirements & equipment available… G.711 (PCM) : requires 64Kbps G.729A : requires 8Kbps (16kbps including overheads) Using G.729A. 16kbps * 30 = 480kbps 512kbits/second link is enough to carry 30 simultaneous voice channels on 11/18/2009Department of IT, Institute of Computing, BZU, Multan

12. www.bzupages.com Origination Two Types: »PC based origination »Phone based origination 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

13. www.bzupages.com Phone Based Origination  DID (Direct Inward Dialing)  Can setup your own DID’s or Purchase from other organizations…  SIP Origination: Call is transferred to your SIP address…  didx.net provides cheap wholesale DID’s 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

14. www.bzupages.com Termination  a gateway is used that takes calls off the Internet and delivers to PSTN lines.  Can also use termination service by other termination service providers…  almvoip.com provides cheapest white label termination for Pakistan… 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

15. www.bzupages.com What actually those service Providers use? Digium Wildcard TE412P 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

16. www.bzupages.com SHAH RUKH BSIT07-22 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

17. www.bzupages.com The equipments (for client)  ATA  Soft phone  IP Phone  Wi-Fi/WLAN phone 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

18. www.bzupages.com ATA  Analog Telephone Adaptor  converts analog signals to digital data  allows to connect a standard phone to your Internet connection for use with VoIP.  ATAs are sometimes referred to as VoIP gateways. Ordinary Phone  ATA  Ethernet  Router  Internet  Service Provider 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

19. www.bzupages.com Linksys ATA 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

20. www.bzupages.com Soft phone  A soft phone is actually a software application that you install on your computer to create a VoIP user interface. In order to use a soft phone, you’ll need a headset and/or microphone. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

21. www.bzupages.com X-lite : SIP based free softphone 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

22. www.bzupages.com IP phone  An IP phone, or hard phone, is a self-contained piece of equipment (that looks like a regular phone) that can communicate directly via your Internet connection.  IP Phone  Ethernet  Router  Internet  VOIP Service Provider 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

23. www.bzupages.com Linksys SPA941 SIP VOIP Phone 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

24. www.bzupages.com Wi-Fi/WLAN phone  Like IP phones, Wi-Fi/WLAN phones don’t require a computer or ATA to use VoIP. They link directly to your IP Internet connection. Unlike IP phones, they’re wireless and connect to the Internet via a wireless base station. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

25. www.bzupages.com Linksys WIP300 Wi-Fi IP Phone 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

26. www.bzupages.com VOIP connecting directly  It is also possible to bypass a VOIP Service Provider and directly connect to another VOIP user. However, if the VOIP devices are behind NAT routers, there may be problems with this approach. IP Phone  Ethernet  Router  Internet  Router  Ethernet  IP Phone 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

27. www.bzupages.com Benefits of VoIP  Operational cost  VoIP can be a benefit for reducing communication and infrastructure costs. Examples include:  Routing phone calls over existing data networks to avoid the need for separate voice and data networks.  Conference calling, IVR, call forwarding, automatic redial, and caller ID features that traditional telecommunication companies normally charge extra for are available free of charge from open source VoIP implementations such as Asterisk or FreeSWITCH 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

28. www.bzupages.com Benefits of VoIP (cont.)  Costs are lower, mainly because of the way Internet access is billed compared to regular telephone calls.  regular telephone calls are billed by the minute or second,  VoIP calls are billed per megabyte (MB). 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

29. www.bzupages.com Benefits of VoIP (cont.) Increased Functionality Incoming phone calls are automatically routed to your VOIP phone where ever you plug it into the network. Take your VOIP phone with you on a trip, and anywhere you connect it to the Internet, you can receive your incoming calls. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

30. www.bzupages.com Protocols - the language of VOIP  Many protocls…  Most commonly used  H.323  SIP  IAX2 (Inter-asterisk exchange) 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

31. www.bzupages.com Session Initiation Protocol  IETF-based  Developed from work on multi-party conferences  The protocol chosen for next generation mobile and fixed networks (3GPP and IMS)  Huge amount of work extending the protocol 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

32. www.bzupages.com SIP Architecture  SIP is used for  Registration and Call Routing  Call Admission Control (performed by proxy)  Call Establishment  SDP (attached to SIP messages) is used to negotiate the media for the call  RTP/RTCP carries the media directly between the endpoints 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

33. www.bzupages.com SIP Terminology  Endpoints are SIP User Agents (UA)  User Agent Clients (UAC) send requests  User Agent Servers (UAS) process requests and send responses  Most endpoints are both UAC and UAS  Proxies forward requests and responses  They cannot generate new requests  Registrars are UAS that record the location of clients  A Registrar is normally colocated with a proxy 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

34. www.bzupages.com SIP URI  sip:user:password@host:port;uri- parameters?headers  Password can be passed in URI but should not be passed in URI for security. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

35. www.bzupages.com Structure of a SIP message  Request  Request URIsip:user@host  HeadersTo: …, From: …, etc.  BodySDP offer  Response  Status Line180 Ringing  HeadersTo:…, From: …, etc.  BodySDP answer 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

36. www.bzupages.com SIP Request Commands  REGISTER  Used when a user agent first goes online and registers their SIP address and IP address with a Registrar server.  INVITE  Used to invite another User agent to communicate, and then establish a SIP session between them. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

37. www.bzupages.com MUHAMMAD WASIF LAEEQ BSIT07-01 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

38. www.bzupages.com SIP Request Commands (cont.)  ACK  Used to accept a session and confirm reliable message exchanges.  CANCEL  Used to cancel a pending request without terminating the session. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

39. www.bzupages.com SIP Request Commands (cont.)  BYE  Used to terminate the session.  Either the user agent who initiated the session, or the one being called can use the BYE command at any time to terminate the session. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

40. www.bzupages.com Registration of UAC with Registrar 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

41. www.bzupages.com Request and Response Made through Proxy Server 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

42. www.bzupages.com SIP Responses  Informational (1xx)  The request has been received and is being processed.  Success (2xx)  The request was acknowledged and accepted.  Redirection (3xx)  The request can’t be completed and additional steps are required (such as redirecting the user agent to another IP address). 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

43. www.bzupages.com SIP Responses  Client error (4xx)  The request contained errors, so the server can’t process the request  Server error (5xx)  Global failure (6xx) 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

44. www.bzupages.com Private Branch eXchange  A telephone system within an enterprise that switches calls between enterprise users on local lines  Allowing all users to share a certain number of external phone lines.  The main purpose of a PBX is to save the cost of requiring a line for each user to the telephone company's central office 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

45. www.bzupages.com PBX 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

46. www.bzupages.com PBX Features  Welcome Message  Voice Mail  IVR  Call Transfer  Conference Call 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

47. www.bzupages.com 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

48. www.bzupages.com What is Asterisk™?  Asterisk™ is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.  Development of Asterisk™ is governed by Digium. 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

49. www.bzupages.com Asterisk™ Architecture 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

50. www.bzupages.com SIP Proxy #1 INVITE #2 100 Attempt #3 INVITE #4 180 Ringing #5 180 Ringing #6 200 OK #7 200 OK #8 SIP ACK #9 Bi-directional RTP channel #10 SIP BYE #11 SIP 200 OK 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

51. www.bzupages.com Another Implementation of VOIP  Using Jingle  An extension of XMPP (eXtensible Messaging & Presence Protocol)  XMPP Sponsors: 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

52. www.bzupages.com XMPP  XML based, making it very easy to use and extend <message to=‘info@bzupages.com' from=‘wasiflaeeq@voovi.org' type='chat'> thread1 How's that presentation going? 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

53. www.bzupages.com Jingle  Google launched their XMPP network with voice support, then joined the standards effort to define Jingle.  File transfer  Screen sharing  Video  Whiteboard  Anything else that uses a lot of bandwidth or that does streaming 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

54. www.bzupages.com Voovi using XMPP and Jingle 11/18/2009 Department of IT, Institute of Computing, BZU, Multan Client can be downloaded from http://voovi.org/client.rar

55. www.bzupages.com Voovi using XMPP and Jingle 11/18/2009 Department of IT, Institute of Computing, BZU, Multan

56. www.bzupages.com  Thanks 11/18/2009 Department of IT, Institute of Computing, BZU, Multan