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1 IP Telephony (VoIP) CSI4118 Fall 2005. 2 Introduction (1) A recent application of Internet technology – Voice over IP (VoIP): Transmission of voice.

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Presentation on theme: "1 IP Telephony (VoIP) CSI4118 Fall 2005. 2 Introduction (1) A recent application of Internet technology – Voice over IP (VoIP): Transmission of voice."— Presentation transcript:

1 1 IP Telephony (VoIP) CSI4118 Fall 2005

2 2 Introduction (1) 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.

3 3 Introduction (2) Challenge –Voice transmission delay –Call setup: call establishment, call termination, etc. –Backward compatibility with existing PSTN (Public Switched Telephone Network) IP Telephony Standards: –ITU (International Telecommunication Union) controls telephony standards. –IETF (Internet Engineering Task Force) controls TCP/IP standards.

4 4 Encoding, Transmission, & Playback (1) Both groups agree on the basics for encoding and transmission of audio: –Audio is encoded using a well-known standard such as Pulse Code Modulation (PCM). –Audio is transferred using the Real-time Transport Protocol (RTP). –RTP message is encapsulated in a UDP datagram that is further encapsulated in an IP datagram for transmission.

5 5 Encoding, Transmission, & Playback (2) UDP is used for transport because –lower overhead: audio must be played as it arrives. –Playback cannot be stopped to wait for a retransmitted packet. Two independent RTP sessions exist, because an IP phone call involves transfer in two directions –IP phone acts as sender for outgoing data, and –IP phone acts as receiver for incoming data.

6 6 Signaling Systems & 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 (signaling System 7). –In VoIP, signaling protocols are: SIP (Session Initiation Protocol), by IETF H.323, by ITU Megaco & MGCP, jointly by IETF and IUT. –VoIP signaling protocols should be able to interact with SS7.

7 7 A Basic IP Telephone System The simplest IP telephone system uses two basic components: - IP telephone: end device allowing humans to place and receive calls. - Media Gateway Controller: providing overall control and coordination between IP phones; allowing a caller to locate a callee (e.g. call forwarding)

8 8 Interconnection with Others (1) IP telephone system needs to interoperate with PSTN or another IP telephone system. Two additional components needed for such interconnection: –Media Gateway –Signaling Gateway

9 9 Interconnection with Others (2) Media gateway: translates audio between IP network and PSTN. Signaling Gateway: translates signaling operations.

10 10 Signaling Protocols Two major protocols: H.323, SIP H.323, invented by ITU, defines four elements that comprising a signaling system: –Terminal: IP phone –Gatekeeper: provides location and signaling functions; coordinates operation of Gateway. –Gateway: used to interconnect IP telephone system with PSTN, handling both signaling and media translation. –Multipoint Control Unit: provides services such as multipoint conferencing.

11 11 Signaling Protocols 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 users location or IP address –Support servers: Proxy Server: forwards requests from user agents to another location. Redirect Server: provides an alternate called partys location for the user agent to contact. Registrar Server: receives users registration requests and updates the database that location server consults.

12 12 H.323 Characteristics H.323 consists of a set of protocols that work together to handle all aspects of communication, including: –Transmission of a digital audio phone call –Signaling to set up and manage phone call –Allows transmission of video and data while a phone call is in progress –Sends binary message –Incorporates protocols for security –Uses a special hardware Multipoint Control Unit for conferencing calls –Defines servers for address resolution, authentication, accounting, features, etc.

13 13 H.323 Layering H.323 uses both UDP and TCP over IP. –Audio travels over UDP –Data travels over TCP

14 14 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.

15 15 SIP Methods Six basic message types, known as methods:

16 16 An Example 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.

17 17 Telephone Number Mapping & Routing (1) How should users be named? –PSTN follows ITU standard E.164 for phone numbers. E.g. 1- 613-123-4567 –SIP uses IP addresses. E.g. In an integrated network (PSTN + IP), two problems defined: –Locate a user –Find a efficient route to the user IETF proposed two protocols: –ENUM: E.164 NUMbers –TRIP: Telephone Routing over IP

18 18 Telephone Number Mapping & Routing (2) ENUM –Converting E.164 phone number into a Uniform Resource Identifier (URI) –Using Domain Name System to store mapping –A phone number is converted into a special domain name: E.g. 1-800-555-1234

19 19 Telephone Number Mapping & Routing (3) TRIP –Finding a user in an integrated network –Used by location server or other NEs to advertise routes –Independent of signaling protocols –Dividing the world into a set of IP Telephone Administrative Domains (ITADs)

20 20 IP Telephones and Electrical Power Analog telephone system continues to work when electrical power are unavailable –The wires that connect a telephone to the central office supply the power Currently, IP telephones have to depend on an external source of power –IP phones must have both network connection and power connection. –Several mechanism proposed to integrate power with network connections.

21 21 Summary (1) 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

22 22 Summary (2) H.323 uses a set of protocols for call setup and management SIP uses a set of servers to handle various aspects of signaling ENUM maps an E.164 telephone number into a URI (usually SIP URI) TRIP provides routing among IP telephone administrative domains IP telephones depends on external power, while analog phones dont.

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