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Voice over IP
PREPARED BY:- SHIVANGI SINGH
DATA COMMUNICATION & NETWORKING (2171008)
E.C DEPT.
Voice over IP (Internet telephony)
 The idea is to use the Internet as a telephone network with some additional
capabilities.
 Instead of communicating over a circuit-switched network, this application allows
communication between two parties over the packet-switched Internet.
 Two protocols have been designed to handle this type of communication: SIP and
H.323.
SIP
 The Session Initiation Protocol (SIP) was designed by IETE It is an application
layer protocol that establishes, manages, and terminates a multimedia session
(call).
 It can be used to create two-party, multiparty, or multicast sessions.
 SIP is designed to be independent of the underlying transport layer; it can run on
UDP, TCP, or SCTP.
 SIP is a text-based protocol, as is HTTP. SIP, like HTTP, uses messages.
Messages
 Each message has a header and a body. The header consists of several lines that
describe the structure of the message, caller's capability, media type, and so on.
 We give a brief description of each message. Then we show their applications in a
simple session. The caller initializes a session with the INVITE message.
 After the callee answers the call, the caller sends an ACK message for
confirmation. The BYE message terminates a session.
 The OPTIONS message queries a machine about its capabilities. The CANCEL
message cancels an already started initialization process.
 The REGISTER message makes a connection when the callee is not available.
Addresses
 In a regular telephone communication, a telephone number identifies the sender,
and another telephone number identifies the receiver.
 SIP is very flexible. In SIP, an e-mail address, an IP address, a telephone number,
and other types of addresses can be used to identify the sender and receiver.
 However, the address needs to be in SIP format (also called scheme)
Simple Session
 A simple session using SIP consists ofthree modules: establishing, communicating,
and terminating.
Establishing a Session: Establishing a session in SIP requires a three-way
handshake.
 The caller sends an INVITE message, using UDP, TCP, or SCTP to begin the
communication.
 If the callee is willing to start the session, she sends a reply message. To confirm
that a reply code has been received, the caller sends an ACK message.
Communicating: After the session has been established, the caller and the caller can
communicate by using two temporary ports.
Terminating the Session: The session can be terminated with a BYE message sent
by either party.
Tracking the Callee
 When a caller needs to communicate with the callee, the caller can use the e-mail
address instead ofthe IP address in the INVITE message.
 The message goes to a proxy server. The proxy server sends a lookup message
(not part of SIP) to some registrar server that has registered the callee.
 When the proxy server receives a reply message from the registrar server, the
proxy server takes the caller's INVITE message and inserts the newly discovered
IP address of the callee.
 This message is then sent to the callee.
H.323
 H.323 is a standard designed by lTV to allow telephones on the public telephone
network to talk to computers (called terminals in H.323) connected to the Internet.
 A gateway connects the Internet to the telephone network.
 In general, a gateway is a five-layer device that can translate a message from one
protocol stack to another.
 The gateway here does exactly the same thing. It transforms a telephone network
message to an Internet message.
 The gatekeeper server on the local area network plays the role of the registrar
server, as we discussed in the SIP.
Protocols
 H.323 uses a number ofprotocols to establish and maintain voice (or video)
communication.
 H.323 uses G.71 or G.723.1 for compression.
 It uses a protocol named H.245 which allows the parties to negotiate the
compression method.
 Protocol Q.931 is used .for establishing and terminating connections.
 Another protocol called H.225, or RAS (Registration!Administration!Status), is
used for registration with the gatekeeper.
Voice over IP, Data Communication & Networking
Operation of a telephone communication
using H.323
The steps used by a terminal to communicate
with a telephone
 1. The terminal sends a broadcast message to the gatekeeper. The gatekeeper
responds with its IP address.
 2. The terminal and gatekeeper communicate, using H.225 to negotiate bandwidth.
 3. The terminal, gatekeeper, gateway, and telephone communicate by using Q.931
to set up a connection.
 4. The terminal, gatekeeper, gateway, and telephone communicate by using H.245
to negotiate the compression method.
 5. The terminal, gateway, and telephone exchange audio by using RTP under the
management of RTCP.
 6. The terminal, gatekeeper, gateway, and telephone communicate by using Q.931
to terminate the communication.
THANK YOU

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Voice over IP, Data Communication & Networking

  • 1. Voice over IP PREPARED BY:- SHIVANGI SINGH DATA COMMUNICATION & NETWORKING (2171008) E.C DEPT.
  • 2. Voice over IP (Internet telephony)  The idea is to use the Internet as a telephone network with some additional capabilities.  Instead of communicating over a circuit-switched network, this application allows communication between two parties over the packet-switched Internet.  Two protocols have been designed to handle this type of communication: SIP and H.323.
  • 3. SIP  The Session Initiation Protocol (SIP) was designed by IETE It is an application layer protocol that establishes, manages, and terminates a multimedia session (call).  It can be used to create two-party, multiparty, or multicast sessions.  SIP is designed to be independent of the underlying transport layer; it can run on UDP, TCP, or SCTP.  SIP is a text-based protocol, as is HTTP. SIP, like HTTP, uses messages.
  • 5.  Each message has a header and a body. The header consists of several lines that describe the structure of the message, caller's capability, media type, and so on.  We give a brief description of each message. Then we show their applications in a simple session. The caller initializes a session with the INVITE message.  After the callee answers the call, the caller sends an ACK message for confirmation. The BYE message terminates a session.  The OPTIONS message queries a machine about its capabilities. The CANCEL message cancels an already started initialization process.  The REGISTER message makes a connection when the callee is not available.
  • 6. Addresses  In a regular telephone communication, a telephone number identifies the sender, and another telephone number identifies the receiver.  SIP is very flexible. In SIP, an e-mail address, an IP address, a telephone number, and other types of addresses can be used to identify the sender and receiver.  However, the address needs to be in SIP format (also called scheme)
  • 7. Simple Session  A simple session using SIP consists ofthree modules: establishing, communicating, and terminating.
  • 8. Establishing a Session: Establishing a session in SIP requires a three-way handshake.  The caller sends an INVITE message, using UDP, TCP, or SCTP to begin the communication.  If the callee is willing to start the session, she sends a reply message. To confirm that a reply code has been received, the caller sends an ACK message. Communicating: After the session has been established, the caller and the caller can communicate by using two temporary ports. Terminating the Session: The session can be terminated with a BYE message sent by either party.
  • 10.  When a caller needs to communicate with the callee, the caller can use the e-mail address instead ofthe IP address in the INVITE message.  The message goes to a proxy server. The proxy server sends a lookup message (not part of SIP) to some registrar server that has registered the callee.  When the proxy server receives a reply message from the registrar server, the proxy server takes the caller's INVITE message and inserts the newly discovered IP address of the callee.  This message is then sent to the callee.
  • 11. H.323  H.323 is a standard designed by lTV to allow telephones on the public telephone network to talk to computers (called terminals in H.323) connected to the Internet.
  • 12.  A gateway connects the Internet to the telephone network.  In general, a gateway is a five-layer device that can translate a message from one protocol stack to another.  The gateway here does exactly the same thing. It transforms a telephone network message to an Internet message.  The gatekeeper server on the local area network plays the role of the registrar server, as we discussed in the SIP.
  • 13. Protocols  H.323 uses a number ofprotocols to establish and maintain voice (or video) communication.  H.323 uses G.71 or G.723.1 for compression.  It uses a protocol named H.245 which allows the parties to negotiate the compression method.  Protocol Q.931 is used .for establishing and terminating connections.  Another protocol called H.225, or RAS (Registration!Administration!Status), is used for registration with the gatekeeper.
  • 15. Operation of a telephone communication using H.323
  • 16. The steps used by a terminal to communicate with a telephone  1. The terminal sends a broadcast message to the gatekeeper. The gatekeeper responds with its IP address.  2. The terminal and gatekeeper communicate, using H.225 to negotiate bandwidth.  3. The terminal, gatekeeper, gateway, and telephone communicate by using Q.931 to set up a connection.  4. The terminal, gatekeeper, gateway, and telephone communicate by using H.245 to negotiate the compression method.  5. The terminal, gateway, and telephone exchange audio by using RTP under the management of RTCP.  6. The terminal, gatekeeper, gateway, and telephone communicate by using Q.931 to terminate the communication.