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  1. 1. IP Telephony (VoIP)
  2. 2. Introduction (1) <ul><li>A recent application of Internet technology – Voice over IP (VoIP): Transmission of voice over Internet </li></ul><ul><li>How VoIP works </li></ul><ul><ul><li>Continuously sample audio </li></ul></ul><ul><ul><li>Convert each sample to digital form </li></ul></ul><ul><ul><li>Send digitized stream across Internet in packets </li></ul></ul><ul><ul><li>Convert the stream back to analog for playback </li></ul></ul><ul><li>Why VoIP </li></ul><ul><ul><li>IP telephony is economic; High costs for traditional telephone switching equipments. </li></ul></ul>
  3. 3. Introduction (2) <ul><li>Challenge </li></ul><ul><ul><li>Voice transmission delay </li></ul></ul><ul><ul><li>Call setup: call establishment, call termination, etc. </li></ul></ul><ul><ul><li>Backward compatibility with existing PSTN (Public Switched Telephone Network) </li></ul></ul><ul><li>IP Telephony Standards: </li></ul><ul><ul><li>ITU (International Telecommunication Union) controls telephony standards. </li></ul></ul><ul><ul><li>IETF (Internet Engineering Task Force) controls TCP/IP standards. </li></ul></ul>
  4. 4. Encoding, Transmission, & Playback (1) <ul><li>Both groups agree on the basics for encoding and transmission of audio: </li></ul><ul><ul><li>Audio is encoded using a well-known standard such as Pulse Code Modulation (PCM). </li></ul></ul><ul><ul><li>Audio is transferred using the Real-time Transport Protocol (RTP). </li></ul></ul><ul><ul><li>RTP message is encapsulated in a UDP datagram that is further encapsulated in an IP datagram for transmission. </li></ul></ul>
  5. 5. Encoding, Transmission, & Playback (2) <ul><li>UDP is used for transport because </li></ul><ul><ul><li>lower overhead: audio must be played as it arrives. </li></ul></ul><ul><ul><li>Playback cannot be stopped to wait for a retransmitted packet. </li></ul></ul><ul><li>Two independent RTP sessions exist, because an IP phone call involves transfer in two directions </li></ul><ul><ul><li>IP phone acts as sender for outgoing data, and </li></ul></ul><ul><ul><li>IP phone acts as receiver for incoming data. </li></ul></ul>
  6. 6. Signaling Systems & Protocols <ul><li>Main complexity of VoIP: Call setup and call management. </li></ul><ul><li>The process of establishing and terminating a call is called Signaling . </li></ul><ul><ul><li>In traditional telephone system, signaling protocol is SS7 (signaling System 7). </li></ul></ul><ul><ul><li>In VoIP, signaling protocols are: </li></ul></ul><ul><ul><ul><li>SIP (Session Initiation Protocol), by IETF </li></ul></ul></ul><ul><ul><ul><li>H.323, by ITU </li></ul></ul></ul><ul><ul><ul><li>Megaco & MGCP, jointly by IETF and IUT. </li></ul></ul></ul><ul><ul><li>VoIP signaling protocols should be able to interact with SS7. </li></ul></ul>
  7. 7. A Basic IP Telephone System <ul><li>The simplest IP telephone system uses two basic components: </li></ul><ul><ul><li>IP telephone : end device allowing humans to place and receive calls. </li></ul></ul><ul><ul><li>Media Gateway Controller : providing overall control and coordination between IP phones; allowing a caller to locate a callee (e.g. call forwarding) </li></ul></ul>
  8. 8. Interconnection with Others (1) <ul><li>IP telephone system needs to interoperate with PSTN or another IP telephone system. </li></ul><ul><li>Two additional components needed for such interconnection: </li></ul><ul><ul><li>Media Gateway </li></ul></ul><ul><ul><li>Signaling Gateway </li></ul></ul>
  9. 9. Interconnection with Others (2) <ul><li>Media gateway: translates audio between IP network and PSTN. </li></ul><ul><li>Signaling Gateway: translates signaling operations. </li></ul>
  10. 10. Signaling Protocols <ul><li>Two major protocols: H.323, SIP </li></ul><ul><li>H.323, invented by ITU, defines four elements that comprising a signaling system: </li></ul><ul><ul><li>Terminal: IP phone </li></ul></ul><ul><ul><li>Gatekeeper: provides location and signaling functions; coordinates operation of Gateway. </li></ul></ul><ul><ul><li>Gateway: used to interconnect IP telephone system with PSTN, handling both signaling and media translation. </li></ul></ul><ul><ul><li>Multipoint Control Unit: provides services such as multipoint conferencing. </li></ul></ul>
  11. 11. Signaling Protocols <ul><li>SIP: Session Initiation Protocol. Invented by IETF. </li></ul><ul><li>SIP defines three main elements that comprise a signaling system: </li></ul><ul><ul><li>User Agent: IP phone or applications </li></ul></ul><ul><ul><li>Location servers: stores information about user’s location or IP address </li></ul></ul><ul><ul><li>Support servers: </li></ul></ul><ul><ul><ul><li>Proxy Server : forwards requests from user agents to another location. </li></ul></ul></ul><ul><ul><ul><li>Redirect Server : provides an alternate called party’s location for the user agent to contact. </li></ul></ul></ul><ul><ul><ul><li>Registrar Server : receives user’s registration requests and updates the database that location server consults. </li></ul></ul></ul>
  12. 12. H.323 Characteristics <ul><li>H.323 consists of a set of protocols that work together to handle all aspects of communication, including: </li></ul><ul><ul><li>Transmission of a digital audio phone call </li></ul></ul><ul><ul><li>Signaling to set up and manage phone call </li></ul></ul><ul><ul><li>Allows transmission of video and data while a phone call is in progress </li></ul></ul><ul><ul><li>Sends binary message </li></ul></ul><ul><ul><li>Incorporates protocols for security </li></ul></ul><ul><ul><li>Uses a special hardware Multipoint Control Unit for conferencing calls </li></ul></ul><ul><ul><li>Defines servers for address resolution, authentication, accounting, features, etc. </li></ul></ul>
  13. 13. H.323 Layering <ul><li>H.323 uses both UDP and TCP over IP. </li></ul><ul><ul><li>Audio travels over UDP </li></ul></ul><ul><ul><li>Data travels over TCP </li></ul></ul>
  14. 14. SIP Characteristics <ul><li>Operates at the application layer. </li></ul><ul><li>Encompasses all aspects of signaling, e.g. location of called party, ringing a phone, accepting a call, and terminating a call. </li></ul><ul><li>Provides services such as call forwarding. </li></ul><ul><li>Relies on multicast for conference calls. </li></ul><ul><li>Allows two sides to negotiate capabilities and choose the media and parameters to be used. </li></ul><ul><li>SIP URI is similar to email address. (with prefix “sip:”) E.g. </li></ul>
  15. 15. SIP Methods <ul><li>Six basic message types, known as methods : </li></ul>
  16. 16. An Example SIP Session <ul><li>User agent A contacts DNS server to map domain name in SIP request to IP address. </li></ul><ul><li>User agent A sends a INVITE message to proxy server that uses location server to find the location of user agent B. </li></ul><ul><li>Call is established between A and B. Then media session begins. </li></ul><ul><li>Finally, B terminates the call by sending a BYE request. </li></ul>
  17. 17. Telephone Number Mapping & Routing (1) <ul><li>How should users be named? </li></ul><ul><ul><li>PSTN follows ITU standard E.164 for phone numbers. E.g. 1-613-123-4567 </li></ul></ul><ul><ul><li>SIP uses IP addresses. E.g. </li></ul></ul><ul><li>In an integrated network (PSTN + IP), two problems defined: </li></ul><ul><ul><li>Locate a user </li></ul></ul><ul><ul><li>Find a efficient route to the user </li></ul></ul><ul><li>IETF proposed two protocols: </li></ul><ul><ul><li>ENUM: E.164 NUMbers </li></ul></ul><ul><ul><li>TRIP: Telephone Routing over IP </li></ul></ul>
  18. 18. Telephone Number Mapping & Routing (2) <ul><li>ENUM </li></ul><ul><ul><li>Converting E.164 phone number into a Uniform Resource Identifier (URI) </li></ul></ul><ul><ul><li>Using Domain Name System to store mapping </li></ul></ul><ul><ul><li>A phone number is converted into a special domain name: </li></ul></ul><ul><ul><ul><li>E.g. 1-800-555-1234  </li></ul></ul></ul>
  19. 19. Telephone Number Mapping & Routing (3) <ul><li>TRIP </li></ul><ul><ul><li>Finding a user in an integrated network </li></ul></ul><ul><ul><li>Used by location server or other NEs to advertise routes </li></ul></ul><ul><ul><li>Independent of signaling protocols </li></ul></ul><ul><ul><li>Dividing the world into a set of IP Telephone Administrative Domains (ITADs) </li></ul></ul>
  20. 20. IP Telephones and Electrical Power <ul><li>Analog telephone system continues to work when electrical power are unavailable </li></ul><ul><ul><li>The wires that connect a telephone to the central office supply the power </li></ul></ul><ul><li>Currently, IP telephones have to depend on an external source of power </li></ul><ul><ul><li>IP phones must have both network connection and power connection. </li></ul></ul><ul><ul><li>Several mechanism proposed to integrate power with network connections. </li></ul></ul>
  21. 21. Summary (1) <ul><li>IP telephony or VoIP refers to the transmission of voice telephone calls over IP networks. </li></ul><ul><li>Hot area both in research and market because of low cost </li></ul><ul><li>Challenge in backward compatibility with PSTN </li></ul><ul><li>The complexity of IP telephony is on signaling. Both ITU and IETF propose signaling standards. </li></ul><ul><ul><li>H.323, by IUT </li></ul></ul><ul><ul><li>SIP, by IETF, offering similar functions to H.323, but simpler than H.323. </li></ul></ul><ul><ul><li>Both are competing to be recognized as #1 signaling protocol </li></ul></ul>
  22. 22. Summary (2) <ul><li>H.323 uses a set of protocols for call setup and management </li></ul><ul><li>SIP uses a set of servers to handle various aspects of signaling </li></ul><ul><li>ENUM maps an E.164 telephone number into a URI (usually SIP URI) </li></ul><ul><li>TRIP provides routing among IP telephone administrative domains </li></ul><ul><li>IP telephones depends on external power, while analog phones don’t. </li></ul>