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Transcript of 1 IP Telephony (VoIP) CSI4118 Fall 2005. 2 Introduction (1) A recent application of Internet...
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IP Telephony (VoIP)
CSI4118
Fall 2005
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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.
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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.
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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.
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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.
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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.
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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)
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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
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Interconnection with Others (2)
• Media gateway: translates audio between IP network and PSTN.
• Signaling Gateway: translates signaling operations.
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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.
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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.
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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.
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H.323 Layering
• H.323 uses both UDP and TCP over IP.– Audio travels over UDP– Data travels over TCP
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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:[email protected]
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SIP Methods
• Six basic message types, known as methods:
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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.
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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:[email protected]
• 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
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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
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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)
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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.
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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
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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.