3-the session initiation protocol

7/31/2019 3-The Session Initiation Protocol

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The Session Initiation Protocol - SIP

Internation Institute of Telecommunications inc., 2000-2004

www.iitelecom.com


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H.323 Specific Protocols

TCPUDP

IP

Connection (IEEE 802.3)

RTP/RTCP

H.261

H.263

Audio Video

T.120H.225

RAS

H.225Q.931

Signaling

H.245

of controlsignaling

Control and managementof the calls

Data

G.711G.723.1

G.726G.728G.729


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H.323 environment and components

Network

IP

H.323 terminalremote access

Access Server

Gatekeeper

PBX

Gateway

PSTN

IP telephone

H.323 terminal

MCU


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Session Initiation Protocol (SIP)

Signaling protocol for multimedia applications

Independent of sub layer protocols (TCP, UDP)

Standard developed by the IETF (MMUSIC working group) - RFC 2543

SIP works in various phases of the call

Localization of the corresponding terminal

Analyze recipient profile and resources

Negotiation of the media type and of the communication parameters

Availability of the correspondent Call set-up and call follow-up

SIP uses several existing protocols

Message format (HTTP 1.1)

Media negotiation (SDP - Session Description Protocol),

Media (RTP) Name resolution and mobility (DNS and DHCP)

Applications encoding (MIME)


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SIP Specific Protocols

Physical

IP

TCP/UDP

RTP/RTCP

G.711G.729

G.723.1

H.261

H.263

Audio Video

SIP SDP

Signaling


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Registering an IP phone

Proxy Server

Registration Server

1. Register

2. 100 Trying

5. 200 OK

4. 200 OK

3. Register


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SIP Addressing

SIP Addresses are identified by URL, in the form

user@host user = name or telephone number

host = domain name or IP addresses

Examples

sip:[email protected] sip:[email protected]

sip:[email protected]; user=phone


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SIP Components

User Agent

An end user application initiating, receiving and terminating a call

Proxy Server

An application server conveying the requests on behalf of the enduser application

The request is processed and sent to the destination (called person)

or to another server

Redirect Server

An application server determining the destination address (To:)and returning it to the end user application


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SIP Components (cont'd)

Localization Server

Used by the Proxy Server and Redirect Server to obtain the location ofthe called user (one or more addresses)

Registration Server

Accept registration requests from the client applications

Generally, the service is offered by the Proxy Server or Redirect Server

DNS Server

Used to locate the Proxy Server or Redirect Server


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SIP components and services

SIP Servers and services

ProxySIP Server

Registrar RedirectLocationDatabase

Register

I am here

Redirect

Here is the address

LocateWhere this name is

or tel. number

INVITEI want to speak

with another agent.

Proxy INVITESI will call it

for you.

SIP UserAgents

GW SIP

SIP UserAgents


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INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 192.168.1.20:5060Call-ID: [email protected]: sip: [email protected]: sip:[email protected] 1 INVITESContent-type: application/sdpContent-Length: 98v = (protocol version)O = (owner/creator and session to identify)

C = (session information)T = (time the session is active)m = (media name and address transport)

Example: INVITE

SDP - Session Description Protocol

SDP defines the conversation parameters on the client application

(User Agent)

SDP transmits information required to establish a multimedia

session

SDP is similar to H.245 in H.323 functions

SDP contains the following parameters:

Medium to be used (codec, sampling rate)

Destination (IP address and port number)

Session name Session duration

Contact

etc


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SDP Messages

v=0

Indicates the start of the SDP content.

o=marie 3123 121231 IN IP4 192.190.132.20

o: session origin and owners name

c=IN IP4 192.190.132.20

c: connect information Specifies the IP address of a session.

m=audio 5004 RTP/AVP 0

Media name and transport address


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SIP session set-up

Contents

Signaling

INVITE

100 Trying

180 Ringing

200 OK

ACK

Logical opening of RTP channel

Bye

200 OKSignaling

Logical opening of RTCP channel

Logical opening of RTP channel

Logical opening of RTCP channel

Each end knows

the other one IP address

Media (UDP)


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SDP Messages in a SIP session

INVITE

100 Trying

180 Ringing

200 OK

ACK

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 192.168.1.20:5060Call-ID: [email protected]: sip: [email protected]: sip:[email protected] 1 INVITES

Content-type: application/sdpContent-Length: 98v=0o=marie 3123 121231 IN IP4 192.190.132.20c=IN IP4 192.190.132.20m=audio 5004 RTP/AVP 0

SIP/2.0 200 OKCall-ID: [email protected]: sip: [email protected]: sip:[email protected] 1 INVITESContent-type: application/sdpContent-Length: 98v=0o=pierre 5664 456456 IP IP4 192.190.132.31c=IN IP4 192.190.132.31m=audio5004 RTP/AVP 0

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 192.190.132.20:5060Call-ID: [email protected]: sip: mar [email protected]: sip:[email protected] 1 ACK

Marie192.168.1.20

Pierre192.168.1.31

Each end knows

the other one IP address


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SIP message types

SIP is modeled on HTTP

Use same syntax and semantics as HTTP Request

Method (INVITE, ACK, BYE, etc.) Header (Accept, Contact, etc.)

Answer Status code (200 OK, 180 Ringing, etc.)

Header (Content-type, Content-encoding, etc.)SIP Methods

INVITE Initiate a call by inviting a user to take part in a session.

ACK Confirm that the client received a final response

to a request INVITES.

BYE Indicate the end of the call.

CANCEL Cancel a request.

REGISTER To register the User Agent.

OPTIONS Used to know the capacities of the server.

SIPAnswers

1xx - Informational Messages.

2xx - Successful Responses.

3xx - Redirection Responses.

4xx - Request Failure Responses.

5xx - Server Failure Responses.

6xx - Global Failure Responses.


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SIP in Proxy mode

[email protected]

INVITE [email protected]

From: [email protected]

Pierre? [email protected]

Location

Server

[email protected]



200 OK200 OK

ACKACK

2 3

4

5

8

6

7

Established sessionProxy

Server


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SIP in Redirect mode

[email protected]



Pierre? [email protected]

Locationserver

[email protected]



302 Moved

Contact: [email protected]

ACK

2 3

6

200 OK7

ACK8

4

5

Established session

Redirect

Server


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SIP call exampleCall forward busy from B to C

INVITE

100 Trying 486 Busy

INVITE

ACK

INVITE

180 Ringing

200 OK

180 Ringing

200 OK

ACK

Established session

ACK

Proxy ServerUA A UA B UA C


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SIP call exampleCall transfer from A to C

Bye (also C)

100 Trying

INVITE (req A)

180 Ringing

200 OK

ACK

Established session

Proxy Server

Bye (also C)

200 OK

INVITE (req A)

200 OK

180 Ringing

200 OK

ACK

Established session

UA A UA B UA C


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Sending an INVITE through a SIP basedsystem

1. INVITE

2. 100 Trying

6. INVITE

7. 100 Trying

3. INVITE

5. ACK

10.ACK

9. 302 MovedTemporary

8. INVITE

11. INVITE

12. 100 Trying

Proxy Server A

4. 302 Movedtemporary

RedirectServer

Proxy Server B

SIP Phone ASIP Phone B


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Call set up through a SIP based phonesystem

15. 180

18. 200 OK 17. INVITE

20. ACK

13. 180 Ringing

16. 200 OK

Proxy Server A

Redirect Server

Proxy Server B


21. ACK

14. 180 Ringing

19. ACK

22. RTP


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Call teardown through a SIP-based system

27. BYE

28. 100 Trying 26. 100 Trying

30. ACK

23. BYE

24. 100 Trying

Proxy Server A

Redirect Server

Proxy Server B

SIP Phone A

SIP Phone B

31. 200 OK

25. BYE

29. 200 OK


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Forking: sending the invite to two phones

1. INVITE

2. 100 Trying

6. INVITE

7. 100 Trying

3. INVITE

5. ACK

10.ACK

9. 302 Moved

Temporary

8. INVITE

11. INVITE

14. 100 Trying

Proxy Server A

4. 302 Moved

temporary

RedirectServer

Proxy Server B


12. INVITE 13. 200 Trying

SIP Phone C


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Forking: the phones ring

17. 180 Ringing

20. 180 Ringing19. 180 Ringing

15. 180 Ringing

Proxy Server A

RedirectServer

Proxy Server B


18. 180 Ringing

SIP Phone C

16. 180 Ringing


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Forking: Phone B answers the phone

25. 200 OK23. 200 OK

21. 200OK

Proxy Server A

RedirectServer

Proxy Server B


22. CANCEL

SIP Phone C

24. 200 OK


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SIP and Security Considerations

SIP Security

Security Mechanisms

Attack and protection

Limitations


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SIP Security

Authentication: proof of identity

Confidentiality

Integrity


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Attacks and protections

Registration Hijacking

Authentication

Impersonating a Server (spoofing)

Server Authentication

Proxy Server can behave as Man in the Middle

End to end Authentication Confidentiality

Integrity


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Attacks and protections ( cont)

Tearing Down Sessions

Authentication of the BYE sender Confidentiality

Denial of Service

Authentication of Register request


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User Mobility

Terminal Mobility

Terminal Moving between networks

Session Mobility

User moving between terminals, in runtime

User Mobility

Users changing terminals

Service Mobility

Keep same services, while mobile


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SIP and Terminal Mobility

Terminal can move between sub networks

GSM and WLAN makes it possible

Mobile hosts use Register message to inform their server about their

new locations.


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SIP and terminal mobility


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SIP and Terminal Mobility


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SIP and Session mobility

Re-INVITE is used in session mobility


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SIP and Session Mobility


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SIP and User Mobility

User changes terminals

The users registers themselves to the appropriate server

Location server tracks the location of user

Redirect servers prioritize the possible locations of the user


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SIP and Service Mobility

Keep same services, while mobile

Services located at home server

Call is processed by home servers using RECORD-ROUTE


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3G and SIP

3G uses IP technology to deliver multimedia content

3G uses SIP for call control, multimedia and signaling function.

Users will be identified by SIP URLs and/or E.164 numbers, the

numbering system of the telephone system.

SIP handles the movement of the mobile user from one domain to

another


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Benefits of 3G

Mobility

Connectivity

Reachability

WLAN d SIP


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WLAN and SIP

Adds SIP based VoIP communications together with WiFi installations

Wireless access protocol: CSMA/ CA

Call control Protocol: SIP

SIP R f


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SIP References

Columbia university Web site

http://www.cs.columbia.edu/sip/

IETF SIP working group

RFC 3261

http://ietf.org/html.charters/sip-charter.html

SIP forum

http://www.sipforum.org

Ubiquity Information Center : SIP center

http://www.sipcenter.com


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3-the session initiation protocol

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