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Vo Ip Rajibdeka

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A recent application of Internet technology – Voice over IP (VoIP): Transmission of voice over Internet.

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Vo Ip Rajibdeka

  1. 1. VoIP Basics Rajib Deka Sr. Programmer Siemens Ltd. Chennai-100. Rajib Deka 1
  2. 2. Agenda Internet Basics Protocol Layering Voice Over IP VoIP Architecture VoIP Network VoIP Protocols SIP Basics Conclusion Rajib Deka 2
  3. 3. Internet Basics The basic building block of networks is the IP datagram. Analogy to datagram - a postcard with  Destination address  Return address  A small amount of text A postcard might inform you of a friend’s holiday travels or remind you of a dentist’s appointment. The postal service doesn’t care which application (friend or dentist) sent the postcard—it just carries processed wood pulp with black marks. Rajib Deka 3
  4. 4. Protocol Layering Rajib Deka 4
  5. 5. Voice Over IP - Introduction A recent application of Internet technology – Voice over IP (VoIP): Transmission of voice over Internet. How VoIP works  Continuously sample audio  Convert each sample to digital form  Send digitized stream across Internet in packets  Convert the stream back to analog for playback Why VoIP  IP telephony is economic; High costs for traditional telephone switching equipments.  Call setup: call establishment, call termination, etc.  Backward compatibility with existing PSTN (Public Switched Telephone Network) Rajib Deka 5
  6. 6. Voice Over IP - Introduction IP Telephony Standards:  ITU (International Telecommunication Union) controls telephony standards.  IETF (Internet Engineering Task Force) controls TCP/IP standards.  Audio is encoded using a well-known standard such as Pulse Code Modulation (PCM). UDP is used for transport:  lower overhead: audio must be played as it arrives.  Playback cannot be stopped to wait for a retransmitted packet. Rajib Deka 6
  7. 7. VoIP Architecture VoIP Server:  Media Gateway.  Media Gateway Controller.  Signaling Gateway.  IP PBX and Proxy. VoIP Client:  Soft Phones.  IP Phones. Rajib Deka 7
  8. 8. VoIP Network Rajib Deka 8
  9. 9. VoIP Network Rajib Deka 9
  10. 10. VoIP Protocols Main complexity of VoIP: Call setup and call management. The process of establishing and terminating a call is called Signaling. In traditional telephone system, signaling protocol is SS7. In VoIP, signaling protocols are:  SIP (Session Initiation Protocol), by IETF  H.323, by ITU  Megaco & MGCP, jointly by IETF and IUT. Audio Signaling:  RTP: Real-time Transport Protocol.  RTCP: Real Time Control Protocol. VoIP signaling protocols should be able to interact with SS7. Rajib Deka 10
  11. 11. SIP Basics SIP: Session Initiation Protocol. Invented by IETF. SIP defines three main elements that comprise a signaling system:  User Agent: IP phone or applications  Location servers: stores information about user’s location or IP address  Support servers:  Proxy Server: forwards requests from user agents to another location.  Redirect Server: provides an alternate called party’s location for the user agent to contact.  Registrar Server: receives user’s registration requests and updates the database that location server consults. Rajib Deka 11
  12. 12. SIP Characteristics Operates at the application layer. Encompasses all aspects of signaling, e.g. location of called party, ringing a phone, accepting a call, and terminating a call. Provides services such as call forwarding. Relies on multicast for conference calls. Allows two sides to negotiate capabilities and choose the media and parameters to be used. SIP URI is similar to email address. (with prefix “sip:”) E.g. sip:rajib.deka@siemens.com Rajib Deka 12
  13. 13. SIP Methods Six basic message types, known as methods: Rajib Deka 13
  14. 14. SIP Session  User agent A contacts DNS server to map domain name in SIP request to IP address.  User agent A sends a INVITE message to proxy server that uses location server to find the location of user agent B.  Call is established between A and B. Then media session begins.  Finally, B terminates the call by sending a BYE request. Rajib Deka 14
  15. 15. RTP and RTCP RTP used to send real-time streams of data across a network is simply called the Real Time Protocol (RTP for short). RTP has been originally defined by IETF. RTCP accompanies RTP and is used to transmit control information about the RTP session. RTCP packets are send only from time to time since there is a recommendation that the RTCP traffic should consume less than 5 percent of the session bandwidth. The most important content types carried in RTCP packets include:  Information about call participants (for example, name and e-mail address)  Statistics about the quality of the transmission (for example inter- arrival jitter and the number of lost packets). RTCP to monitor Quality of Service (QoS). Rajib Deka 15
  16. 16. VoIP and QoS QoS (Quality of Service) is a major issue in VOIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interference from other lower priority traffic. Things to consider are  Latency: Delay for packet delivery  Jitter: Variations in delay of packet delivery  Packet loss: Too much traffic in the network causes the network to drop packets  Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in bursts. Rajib Deka 16
  17. 17. VoIP Developer’s choice Language:  C, C++, Java for Core protocol stack development.  Java, C# for middle tier or application development. Open Source VoIP Servers (Linux Based)  Asterisk PBX (Multi protocol support)  OpenSIPS (for SIP only)  sipXecs (for SIP only) VoIP Clients  X-lite.  VoIP Communicator. Open Source SIP stack  JainSIP  sipXecs  PJSIP Rajib Deka 17
  18. 18. Conclusion IP telephony or VoIP refers to the transmission of voice telephone calls over IP networks. Hot area both in research and market because of low cost Challenge in backward compatibility with PSTN The complexity of IP telephony is on signaling. Both ITU and IETF propose signaling standards.  H.323, by IUT  SIP, by IETF, offering similar functions to H.323, but simpler than H.323.  Both are competing to be recognized as #1 signaling protocol. Rajib Deka 18
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