The Session Initiation Protocol - SIP

www.iitelecom.com
Internation Institute of Telecommunications inc., 2000-2004

H.323 Specific Protocols

Audio

Video

G.711
G.723.1
H.261
G.726
H.263
G.728
G.729
RTP/RTCP

Control and management

of the calls

Data

T.120

H.225
RAS

H.225
Q.931
Signaling

TCP

UDP
IP
Connection (IEEE 802.3)

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H.245
of control
signaling

H.323 environment and components

IP telephone

Gatekeeper

H.323 terminal

MCU

Network
IP

Gateway

PSTN

Access Server

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PBX

H.323 terminal
remote access

Session Initiation Protocol (SIP)

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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)

SIP Specific Protocols

Audio

Video

G.711
G.729
G.723.1

H.261
H.263

Signaling

SIP

RTP/RTCP

TCP/UDP
IP
Physical

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SDP

Registering an IP phone
3. Register
4. 200 OK

Registration Server

1. Register
2. 100 Trying
5. 200 OK

Proxy Server

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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

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sip:xyz@iitelecom.com
sip:xyz@192.168.10.1
sip:5141234567@iitelecom.com; user=phone

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 end

user application
The request is processed and sent to the destination (called person)
or to another server

Redirect Server

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An application server determining the destination address (To:) and

returning it to the end user application

SIP Components (cont'd)

Localization Server

Used by the Proxy Server and Redirect Server to obtain the location of
the 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

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Used to locate the Proxy Server or Redirect Server

SIP components and services

SIP Servers and services

Registrar

Redirect

Location
Database

Locate
Where this name is
or tel. number
Proxy
SIP Server

Register
I am here

SIP User
Agents

Redirect
Here is the address
Proxy INVITES
I will call it
for you.
INVITE
I want to speak
with another agent.

SIP User
Agents
GW SIP

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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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Example: INVITE
INVITE sip:pierre@192.168.1.31 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060
Call-ID: 141710@192.190.132.20
From: sip: marie@192.190.132.20
To: sip:pierre@192.190.132.31
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v = (protocol version)
O = (owner/creator and session to identify)
C = (session information)
T = (time the session is active)
m = (media name and address transport)

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

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Media name and transport address

SIP session set-up

Each end knows
the other one IP address
INVITE
Signaling

100 Trying
180
Ringing
200 OK
ACK
Logical opening of RTP channel
Logical opening of RTCP channel

Contents

Logical opening of RTP channel

Logical opening of RTCP channel
Bye

Signaling

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200 OK

Media (UDP)

SDP Messages in a SIP session

Marie
192.168.1.20

INVITE sip:pierre@192.168.1.31 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.20:5060
Call-ID: 141710@192.190.132.20
From: sip: marie@192.190.132.20
To: sip:pierre@192.190.132.31
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v=0
o=marie 3123 121231 IN IP4 192.190.132.20
c=IN IP4 192.190.132.20
m=audio 5004 RTP/AVP 0

ACK sip:pierre@192.190.132.31 SIP/2.0

Via: SIP/2.0/UDP 192.190.132.20:5060
Call-ID: 141710@192.190.132.20
From: sip: marie@192.190.132.20
To: sip:pierre@192.190.132.31
Cseq 1 ACK

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Each end knows

the other one IP address

Pierre
192.168.1.31

INVITE
100 Trying
180
Ringing
200 OK

SIP/2.0 200 OK
Call-ID: 141710@192.190.132.20
From: sip: marie@192.190.132.20
To: sip:pierre@192.190.132.31
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v=0
o=pierre 5664 456456 IP IP4 192.190.132.31
c=IN IP4 192.190.132.31
m=audio5004 RTP/AVP 0

ACK

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

SIP Answers
1xx - Informational Messages.

to a request INVITES.

2xx - Successful Responses.

3xx - Redirection Responses.

BYE

Indicate the end of the call.

4xx - Request Failure Responses.

CANCEL

Cancel a request.

5xx - Server Failure Responses.

6xx - Global Failure Responses.

REGISTER To register the User Agent.

OPTIONS Used to know the capacities of the server.

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SIP in Proxy mode

Location
Server

Pierre?

INVITE pierre@uqam.ca

Pierre@stanford.edu
4 INVITE pierre@stanford.edu

1 From: marie@iitelecom.com

From: marie@iitelecom.com

6 200 OK

5 200 OK

7 ACK
marie@iitelecom.com

8 ACK
Established session
Proxy
Server

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pierre@stanford.edu

SIP in Redirect mode

Location
server

Pierre?

Pierre@stanford.edu

INVITE pierre@uqam.ca

1 From: marie@iitelecom.com
4

302 Moved
Contact: pierre@stanford.edu

Redirect
Server

5 ACK
marie@iitelecom.com

6 INVITE pierre@stanford.edu
From: marie@iitelecom.com

7 200 OK
8 ACK
pierre@stanford.edu

Established session

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SIP call example

Call forward busy from B to C
Proxy Server

UA A

INVITE
100 Trying

UA B

INVITE
486 Busy
ACK
INVITE
180
Ringing
200 OK

180
Ringing
200 OK
ACK

ACK
Established session

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UA C

SIP call example

Call transfer from A to C
Proxy Server

UA B

UA A

UA C

Established session
Bye (also C)
200 OK

Bye (also C)
200 OK
INVITE (req A)
INVITE (req A)
100 Trying

180
Ringing

180
Ringing
200 OK

200 OK
ACK
ACK
Established session
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Sending an INVITE through a SIP based

system
3. INVITE

10.ACK
9. 302 Moved
Temporary

4. 302 Moved
temporary

8. INVITE

5. ACK
Redirect
Server
1. INVITE
2. 100 Trying

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12. 100 Trying

7. 100 Trying

Proxy Server A

SIP Phone A

11. INVITE

6. INVITE

Proxy Server B

SIP Phone B

Call set up through a SIP based phone

system

Redirect Server

15. 180

14. 180 Ringing

18. 200 OK

17. INVITE

19. ACK

13. 180 Ringing

16. 200 OK
21. ACK

20. ACK

Proxy Server B

Proxy Server A
22. RTP

SIP Phone A

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SIP Phone B

Call teardown through a SIP-based system

Redirect Server

27. BYE

25. BYE

28. 100 Trying

26. 100 Trying

29. 200 OK

23. BYE
24. 100 Trying
31. 200 OK

30. ACK

Proxy Server A

Proxy Server B

SIP Phone B
SIP Phone A

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Forking: sending the invite to two phones

10.ACK

3. INVITE

9. 302 Moved
Temporary

4. 302 Moved
temporary

8. INVITE

5. ACK
Redirect
Server

11. INVITE

1. INVITE
6. INVITE

14. 100 Trying

2. 100 Trying

7. 100 Trying

Proxy Server A

Proxy Server B

12. INVITE

13. 200 Trying

SIP Phone B

SIP Phone A

SIP Phone C

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Forking: the phones ring

Redirect
Server
17. 180 Ringing
16. 180 Ringing
20. 180 Ringing

Proxy Server A

15. 180 Ringing

19. 180 Ringing
Proxy Server B

18. 180 Ringing

SIP Phone B

SIP Phone A

SIP Phone C

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Forking: Phone B answers the phone

Redirect
Server

21. 200OK

25. 200 OK

23. 200 OK

Proxy Server A

Proxy Server B

24. 200 OK

22. CANCEL
SIP Phone B

SIP Phone A

SIP Phone C

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SIP and Security Considerations

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SIP Security

Security Mechanisms

Attack and protection

Limitations

SIP Security

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Authentication: proof of identity

Confidentiality

Integrity

Attacks and protections

Registration Hijacking

Impersonating a Server (spoofing)

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Authentication
Server Authentication

Proxy Server can behave as Man in the Middle

End to end Authentication

Confidentiality

Integrity

Attacks and protections ( cont)

Tearing Down Sessions

Authentication of the BYE sender

Confidentiality

Denial of Service

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Authentication of Register request

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

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SIP and Service Mobility

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Keep same services, while mobile

Services located at home server

Call is processed by home servers using RECORD-ROUTE

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.

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

Benefits of 3G

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Mobility

Connectivity

Reachability

WLAN and SIP

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Adds SIP based VoIP communications together with WiFi installations

Wireless access protocol: CSMA/ CA

Call control Protocol: SIP

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

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http://www.sipcenter.com

Questions?

?
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