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.