johan garcia karlstads universitet datavetenskap 1 datakommunikation ii signaling/voice over ip /...
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Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Signaling/Voice over IP / SIP
Based on material from Henning Schulzrinne, Columbia University.
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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What is signaling?
• ”Control of procedures”
• Network control systems
• Railway traffic systems
• Process control systems
• Telecom systems– ”the distribution of information and instructions from
one telphone node to one or several others to provide for calls, and for network management”
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Telecom signaling
• Two types: access and network signaling
• Signaling info is packet-based, i.e. transferred as messages
• Signaling protocol used today:– Signaling System No. 7 (SS7)
• SS7 constitutes separate network within telecom network
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Voice over IP - motivation
• Telephone switches not very cost effective– Between $150 and $500 for 64kb/s circuit– Ethernet switch $5 - $25 for 100Mb/s port
• Cheaper long-distance calls
• Cheaper to deploy in developing countries
• Cheaper ”advanced services”
• Less bandwidth needed– Higher compression, silence suppression
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Voice over IP – motivation (contd)
• In the future: increased functionality
• Tailored services
• Integration with other Internet services– E.g. web and email
• Integration– Single network for voice and data
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Motivation for VoIP
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Internet Telephony as PBX replacement
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Switching Costs
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Architecture
• Must be able to interwork with PSTN
Three classes:
• Trunk replacement– Caller and callee use circuit-switched phone
• Hop-on or hop-off– Call between PSTN phone to IP-based phone
• End-to-end– IP-based communication end-to-end
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Internet Telephony Modes
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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SIP –Session Initiation protocol• Designed for establishing, modifying and
terminating multimedia sessions• Does not describe audio and/or video components
– Relies on separate session description
• Location of called party, mapping of address types• User devices run SIP user agents
– Can act as both clients and servers
• Can be run over any transport protocol– UDP, TCP or SCTP
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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SIP meddelande
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Metoder
MESSAGE transport of an instant message body
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Media negotiation
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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ResultatkoderInformational
Server Failure
Request FailureRedirectionSuccess
Global Failure
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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SIP proxy mode
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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SIP redirect mode
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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DNS SRV
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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SIP request forking
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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SIP sequential request forking
Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap
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Comparison with H.323• H.323 is another signaling
protocol for real-time, interactive
• H.323 is a complete, vertically integrated suite of protocols for multimedia conferencing: signaling, registration, admission control, transport and codecs.
• SIP is a single component. Works with RTP, but does not mandate it. Can be combined with other protocols and services.
• H.323 comes from the ITU (telephony).
• SIP comes from IETF: Borrows much of its concepts from HTTP. SIP has a Web flavor, whereas H.323 has a telephony flavor.
• SIP uses the KISS principle: Keep it simple stupid.