1. Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia University.

2. Johan Garcia Karlstads Universitet Datavetenskap 2 Datakommunikation II 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”

3. Johan Garcia Karlstads Universitet Datavetenskap 3 Datakommunikation II 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

4. Johan Garcia Karlstads Universitet Datavetenskap 4 Datakommunikation II 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

5. Johan Garcia Karlstads Universitet Datavetenskap 5 Datakommunikation II 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

6. Johan Garcia Karlstads Universitet Datavetenskap 6 Datakommunikation II Motivation for VoIP

7. Johan Garcia Karlstads Universitet Datavetenskap 7 Datakommunikation II Internet Telephony as PBX replacement

8. Johan Garcia Karlstads Universitet Datavetenskap 8 Datakommunikation II Switching Costs

9. Johan Garcia Karlstads Universitet Datavetenskap 9 Datakommunikation II 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

10. Johan Garcia Karlstads Universitet Datavetenskap 10 Datakommunikation II Internet Telephony Modes

11. Johan Garcia Karlstads Universitet Datavetenskap 11 Datakommunikation II Multimedia Protocol stack

12. Johan Garcia Karlstads Universitet Datavetenskap 12 Datakommunikation II 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

13. Johan Garcia Karlstads Universitet Datavetenskap 13 Datakommunikation II SIP meddelande

14. Johan Garcia Karlstads Universitet Datavetenskap 14 Datakommunikation II Metoder MESSAGE transport of an instant message body

15. Johan Garcia Karlstads Universitet Datavetenskap 15 Datakommunikation II Media negotiation

16. Johan Garcia Karlstads Universitet Datavetenskap 16 Datakommunikation II Resultatkoder Informational Server Failure Request FailureRedirectionSuccess Global Failure

17. Johan Garcia Karlstads Universitet Datavetenskap 17 Datakommunikation II SIP proxy mode

18. Johan Garcia Karlstads Universitet Datavetenskap 18 Datakommunikation II SIP redirect mode

19. Johan Garcia Karlstads Universitet Datavetenskap 19 Datakommunikation II DNS SRV

20. Johan Garcia Karlstads Universitet Datavetenskap 20 Datakommunikation II

21. Johan Garcia Karlstads Universitet Datavetenskap 21 Datakommunikation II SIP request forking

22. Johan Garcia Karlstads Universitet Datavetenskap 22 Datakommunikation II SIP sequential request forking

23. Johan Garcia Karlstads Universitet Datavetenskap 23 Datakommunikation II

24. Johan Garcia Karlstads Universitet Datavetenskap 24 Datakommunikation II 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.