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Voice Over IP

Voice Over IP. Group members. Muhammad Aatif Aneeq BSIT07-15 Shah Rukh BSIT07-22 Muhammad Wasif Laeeq BSIT07-01. BSIT07-15. Muhammad Aatif Aneeq. Circuit Switched Network:.

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Voice Over IP

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  1. Department of IT, Institute of Computing, BZU, Multan Voice Over IP

  2. Group members Department of IT, Institute of Computing, BZU, Multan Muhammad Aatif Aneeq BSIT07-15 Shah Rukh BSIT07-22 Muhammad Wasif Laeeq BSIT07-01

  3. BSIT07-15 Muhammad AatifAneeq Department of IT, Institute of Computing, BZU, Multan

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

  5. Packet Switched Network Department of IT, Institute of Computing, BZU, Multan 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.

  6. PSTN Department of IT, Institute of Computing, BZU, Multan 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

  7. PSTN Department of IT, Institute of Computing, BZU, Multan • PSTN lines come in two common standards • Analog • Single Dedicated line • Digital • Multiple lines in one line • e.g. T1 , E1

  8. VoIP Department of IT, Institute of Computing, BZU, Multan 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

  9. VOIP Department of IT, Institute of Computing, BZU, Multan

  10. 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. 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 Department of IT, Institute of Computing, BZU, Multan

  12. Origination Department of IT, Institute of Computing, BZU, Multan Two Types: • PC based origination • Phone based origination

  13. Phone Based Origination Department of IT, Institute of Computing, BZU, Multan 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

  14. Termination Department of IT, Institute of Computing, BZU, Multan 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…

  15. What actually those service Providers use? Digium Wildcard TE412P Department of IT, Institute of Computing, BZU, Multan

  16. BSIT07-22 Shah Rukh Department of IT, Institute of Computing, BZU, Multan

  17. The equipments (for client) Department of IT, Institute of Computing, BZU, Multan ATA Soft phone IP Phone Wi-Fi/WLAN phone

  18. ATA Department of IT, Institute of Computing, BZU, Multan • 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

  19. Linksys ATA Department of IT, Institute of Computing, BZU, Multan

  20. Soft phone Department of IT, Institute of Computing, BZU, Multan 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.

  21. X-lite : SIP based free softphone Department of IT, Institute of Computing, BZU, Multan

  22. IP phone Department of IT, Institute of Computing, BZU, Multan 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

  23. Linksys SPA941 SIP VOIP Phone  Department of IT, Institute of Computing, BZU, Multan

  24. Wi-Fi/WLAN phone Department of IT, Institute of Computing, BZU, Multan 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.

  25. Linksys WIP300 Wi-Fi IP Phone Department of IT, Institute of Computing, BZU, Multan

  26. VOIP connecting directly Department of IT, Institute of Computing, BZU, Multan 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

  27. Benefits of VoIP Department of IT, Institute of Computing, BZU, Multan • 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

  28. Benefits of VoIP (cont.) Department of IT, Institute of Computing, BZU, Multan 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).

  29. Benefits of VoIP (cont.) Department of IT, Institute of Computing, BZU, Multan 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.

  30. Protocols - the language of VOIP Department of IT, Institute of Computing, BZU, Multan • Many protocls… • Most commonly used • H.323 • SIP • IAX2 (Inter-asterisk exchange)

  31. Session Initiation Protocol Department of IT, Institute of Computing, BZU, Multan • 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

  32. SIP Architecture Department of IT, Institute of Computing, BZU, Multan • 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

  33. SIP Terminology Department of IT, Institute of Computing, BZU, Multan • 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

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

  35. Structure of a SIP message Department of IT, Institute of Computing, BZU, Multan • Request • Request URI sip:user@host • Headers To: …, From: …, etc. • Body SDP offer • Response • Status Line 180 Ringing • Headers To:…, From: …, etc. • Body SDP answer

  36. SIP Request Commands Department of IT, Institute of Computing, BZU, Multan • 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.

  37. BSIT07-01 Muhammad Wasif Laeeq Department of IT, Institute of Computing, BZU, Multan

  38. SIP Request Commands (cont.) Department of IT, Institute of Computing, BZU, Multan • ACK • Used to accept a session and confirm reliable message exchanges. • CANCEL • Used to cancel a pending request without terminating the session.

  39. SIP Request Commands (cont.) Department of IT, Institute of Computing, BZU, Multan • 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.

  40. Registration of UAC with Registrar Department of IT, Institute of Computing, BZU, Multan

  41. Request and Response Made through Proxy Server Department of IT, Institute of Computing, BZU, Multan

  42. SIP Responses Department of IT, Institute of Computing, BZU, Multan • 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).

  43. SIP Responses Department of IT, Institute of Computing, BZU, Multan • Client error (4xx) • The request contained errors, so the server can’t process the request • Server error (5xx) • Global failure (6xx)

  44. Private Branch eXchange Department of IT, Institute of Computing, BZU, Multan 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

  45. PBX Department of IT, Institute of Computing, BZU, Multan

  46. PBX Features Department of IT, Institute of Computing, BZU, Multan Welcome Message Voice Mail IVR Call Transfer Conference Call

  47. Department of IT, Institute of Computing, BZU, Multan

  48. What is Asterisk™? Department of IT, Institute of Computing, BZU, Multan 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.

  49. Asterisk™ Architecture Department of IT, Institute of Computing, BZU, Multan

  50. SIP Proxy SIPProxy #1 INVITE #3 INVITE #2 100 Attempt #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 Department of IT, Institute of Computing, BZU, Multan

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