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the ppt describes about voip,its advantages and also describes the protocols and the packet formats involved in a voip session .

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  1. 1. What is VOIP ? <ul><li>VOIP - Voice over internet protocol </li></ul><ul><li>VOIP is the protocol used to transfer voice through packet switched networks </li></ul><ul><li>Voice-over-IP systems carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data-packet stream over IP. </li></ul>
  2. 2. VOIP modes of operation <ul><li>pc to pc </li></ul><ul><li>pc to telephone </li></ul><ul><li>telephone to pc </li></ul><ul><li>telephone to telephone </li></ul>
  3. 3. WHY VOIP ? <ul><li>Cheaper calls </li></ul><ul><li>For a pc-to-pc call , the call is absolutely free </li></ul><ul><li>Merging of data/voice infrastructures </li></ul><ul><li>No worrying about cell phone coverage, roaming, or long distance charges. </li></ul><ul><li>Services like call forwarding, call waiting, voicemail, caller ID and more are available through your ip phone, usually at no extra charge. </li></ul>
  4. 4. PROTOCOLS INVOLVED IN VOIP <ul><li>SESSION ESTABLISHMENT : </li></ul><ul><li>* SIP [ session initiation protocol ] </li></ul><ul><li>* H.323 </li></ul><ul><li>Protocols involved in transfer of data : </li></ul><ul><li>* RTP [ real time transport protocol ] </li></ul><ul><li>* RTCP [ real time transport control protocol ] </li></ul><ul><li>* UDP </li></ul><ul><li>* IP </li></ul>
  5. 5. RTP <ul><li>REAL TIME TRANSPORT PROTOCOL </li></ul><ul><li>Application layer protocol for transmitting realtime data (audio, video, ...) </li></ul><ul><li>Includes payload type identification, sequence numbering, timestamping, delivery monitoring </li></ul><ul><li>Mostly over UDP </li></ul><ul><li>Supports multicast & unicast </li></ul>
  6. 6. Control Protocol - RTCP <ul><li>RTP Control Protocol </li></ul><ul><li>Periodic transmission of control packets to all participants in the session </li></ul><ul><li>Main functions: </li></ul><ul><li>provide feedback on quality of data distribution </li></ul><ul><li>carries a persistent transport-level identifier for an RTP source (CNAME) </li></ul><ul><li>each participant sends control packets to all others which independently observe the number of participants </li></ul>
  7. 7. SIP – Session Initiation Protocol <ul><li>Developed by IETF since 1999 </li></ul><ul><li>SIP is a text-based protocol similar to HTTP and </li></ul><ul><li>SMTP, for initiating interactive communication </li></ul><ul><li>sessions between users </li></ul><ul><li>SIP is an application-layer control (signalling) </li></ul><ul><li>protocol for creating, modifying and terminating </li></ul><ul><li>sessions with one or more participants </li></ul><ul><li>Sessions include Internet Multimedia </li></ul><ul><li>conferences, Internet Telephone calls and </li></ul><ul><li>Multimedia distribution </li></ul>
  8. 8. ENCODING USED FOR VOIP <ul><li>Bandwidth </li></ul><ul><li>Generally modest (64 kbps or less) </li></ul><ul><li>G.711 [ 64 Kbps ] </li></ul><ul><li>G.722 [ 48 – 64 Kbps ] </li></ul><ul><li>G.729 [ 8 Kbps ] </li></ul>
  10. 10. <ul><li>VOIP PACKET FORMAT : </li></ul>
  11. 11. <ul><li>RTP header format : </li></ul>
  12. 12. SIP REQUEST PACKET <ul><li>FIELDS IN SIP REQUEST PACKET : </li></ul><ul><li>1 . Method - method to be performed on the resource. Possible methods are Invite, Ack, Options, Bye, Cancel, Register </li></ul><ul><li>2. request URI - A SIP URL or a general Uniform Resource Identifier, this is the user or service to which this request is being addressed. </li></ul><ul><li>3.SIP version - The SIP version being used; this should be version 2.0 </li></ul>
  13. 13. METHODS IN SIP <ul><li>INVITE [ for initiation of a session ] </li></ul><ul><li>ACK [ confirm final response ] </li></ul><ul><li>BYE [ terminates the call ] </li></ul><ul><li>CANCEL [ cancel searches and ringing ] </li></ul><ul><li>REGISTER [ register with location service] </li></ul>
  14. 14. SIP packet format <ul><li>Fields in a “SIP REQUEST” packet : </li></ul><ul><li>1. SIP VERSION </li></ul><ul><li>2.STATUS CODE - a 3-digit integer result code of the attempt to understand and satisfy the request. </li></ul><ul><li>3.REASON PHRASE </li></ul>
  15. 15. Status codes in SIP <ul><li>1xx Searching, ringing, queuing </li></ul><ul><li>2xx Success </li></ul><ul><li>3xx Forwarding </li></ul><ul><li>4xx Client mistakes </li></ul><ul><li>5xx Server failures </li></ul><ul><li>6xx Busy, refuse, not available anywhere </li></ul>