IP Telephony (VoIP)

1. IP Telephony (VoIP) CSI4118 Fall 2005

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

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. 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 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.

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

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. sip:bob@somewhere.com

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

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. 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. sip:smith@uottawa.ca • 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. 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: e164.arpa • E.g. 1-800-555-1234  4.3.2.1.5.5.5.0.0.8.e164.arpa

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. 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. 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. 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 don’t.