johan garcia karlstads universitet datavetenskap 1 datakommunikation ii signaling/voice over ip /...

22
Johan Garcia Datakommunikation II Karlstads Universitet Datavetenskap 1 Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia University.

Upload: dario-corley

Post on 31-Mar-2015

218 views

Category:

Documents


1 download

TRANSCRIPT

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

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

1

Signaling/Voice over IP / SIP

Based on material from Henning Schulzrinne, Columbia University.

Page 2: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

2

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”

Page 3: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

3

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

Page 4: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

4

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

Page 5: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

5

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

Page 6: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

6

Motivation for VoIP

Page 7: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

7

Internet Telephony as PBX replacement

Page 8: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

8

Switching Costs

Page 9: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

9

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

Page 10: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

10

Internet Telephony Modes

Page 11: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

12

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

Page 12: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

13

SIP meddelande

Page 13: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

14

Metoder

MESSAGE transport of an instant message body

Page 14: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

15

Media negotiation

Page 15: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

16

ResultatkoderInformational

Server Failure

Request FailureRedirectionSuccess

Global Failure

Page 16: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

17

SIP proxy mode

Page 17: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

18

SIP redirect mode

Page 18: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

19

DNS SRV

Page 19: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

21

SIP request forking

Page 20: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

22

SIP sequential request forking

Page 21: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

23

Page 22: Johan Garcia Karlstads Universitet Datavetenskap 1 Datakommunikation II Signaling/Voice over IP / SIP Based on material from Henning Schulzrinne, Columbia

Johan Garcia Datakommunikation IIKarlstads UniversitetDatavetenskap

24

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.