1. The Session Initiation Protocol - SIP
© Internation Institute of Telecommunications inc.,
© , International Institute of Telecommunications Inc.
August 2003

2. H.323 Specific Protocols Control and management of the calls Audio
Video
Data
G.711
G.723.1
G.726
G.728
G.729
H.225
Q.931
Signaling
H.245
of control
signaling
H.261
H.263
H.225
RAS
T.120
RTP/RTCP
UDP
TCP IP
Connection (IEEE 802.3)

3. H.323 environment and components
IP telephone
Gatekeeper
Network IP
H.323 terminal
MCU
Gateway
PSTN
Access Server
VoIP samples the voice signal and encapsulates it in an IP packets, transports it via various equipment supporting VoIP. These voice packets are then recombined into a continuous voice signal which is transmitted to the recipient with an appropriate quality.
In our example, IP is on all the components except for the conventional telephones which have a PSTN connectivity to the H.323 network via a Gateway.
Connectivity with the IP network can be carried out by a modem, an ASDL access, a LAN, WLAN, etc. The users can communicate between them and the type of connection does not matter.
The remote access server (RAS) platform is the foundation on which Internet service providers (ISP) and other telecommunications carriers are bringing to market integrated, Internet-based versions of traditional services such as voice over IP (VoIP), fax over IP (FoIP), and data over IP.
H.323 terminal
remote access
PBX

4. 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)
The protocol responsible of initialing a SIP session is a signaling protocol belonging to the application layer of the OSI model. Its role is to open, modify and release the call session. The opening of these sessions makes it possible to carry out audio or videoconference, remote teaching, and multimedia diffusion primarily on IP. Let’s note that with SIP, the users who open a session can communicate in multicast mode or in point-to-point mode.
To open a session, the user sends an invitation transporting a session descriptor allowing the users wishing to communicate to agree on the compatibility of their media. Thus SIP can connect mobile stations by transmitting or redirecting the requests towards the current position of the station called.
Let’s also note that SIP has the advantage of not being attached to a particular medium and is being independent of the transport protocol of the lower layers. The goal of the RFC 2543 is to detail and specify these concepts.

5. SIP Specific Protocols
Audio
Video
Signaling
G.711
G.729
G.723.1 …
H.261
H.263
SIP
SDP
RTP/RTCP
TCP/UDP IP
Physical
Since SIP architecture is layered, it involves a group of protocol, similar to H.323.
RTP (Real-time Protocol) to transport information in real time;
RTCP (Real-time Control Protocol) to ensure the multimedia control of data flow;
SDP (Session Description Protocol) is a multimedia protocol for the description of the sessions.

6. Registering an IP phoneOK
Registration Server
1. Register
Trying OK
Proxy Server

7. SIP Addressing
SIP Addresses are identified by URL, in the form
user = name or telephone number
host = domain name or IP addresses
Examples
user=phone
In SIP, the addresses are identified by the URL (Uniform Resource Locator). These addresses take the form which is similar to an address.
SIP Addresses are written in the form
The user’s SIP address can also be a telephone number to allow the interconnection with the traditional telephone networks; example sip:514 To indicate that this one uses a telephone number, one adds to the URL the user=phone term. In this case, the call is conveyed to a gateway which makes interface with the traditional telephone network (PSTN).

8. SIP Components User Agent Proxy Server Redirect Server
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
An application server determining the destination address (To:) and returning it to the end user application
SIP Architecture includes several components:
The User agent (UA) resides at the terminal level (PC or a SIP phone). The user agent has a part known as the client (UAC) which sends the requests, and a part known as server (UAS) who answers the requests.
The Proxy server (PS) or relay agent is a terminal that conveys the received request of the client. Such a server can decode and modify the messages which it receives before retransmitting them.
The Redirect server (RS) provides the new destination address to the client. Let ‘s note that a Redirect Server is accessed by the UAC like a regular server and cannot send requests like the PS.
The Localization server (LS) provides the current users position.
The Registrar Server is a server which accepts REGISTER requests. Each PS or RS is generally connected to a Registrar.

9. 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
Used to locate the Proxy Server or Redirect Server

10. SIP components and services
SIP Servers and services
Locate
Where this name is
or tel. number…
Location Database
Proxy SIP Server
Registrar
Redirect
Redirect
Here is the address
Register
I am here
Proxy INVITES
I will call it
for you.
SIP User
Agents
INVITE
I want to speak
with another agent.
SIP User
Agents
GW SIP
To locate a SIP user, a terminal must record its position with the localization server by sending a REGISTER request. A terminal user can constantly move from one location to another. Several positions for the same terminal are then recorded.
When a SIP server accesses its localization server, it draws up a list of the possible user positions from the received results. This list contains one or more positions. In the case of a RS, this list is returned to the requesting client using the information in the header field.

11. 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…
Example: INVITE
INVITE SIP/2.0
Via: SIP/2.0/UDP :5060
Call-ID:
From: sip:
To:
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)

12. SDP Messages v=0 o=marie 3123 121231 IN IP4 192.190.132.20
Indicates the start of the SDP content.
o=marie IN IP
o: session origin and owner’s name
c=IN IP
c: connect information Specifies the IP address of a session.
m=audio 5004 RTP/AVP 0
Media name and transport address

13. SIP session set-up Each end knows the other one IP address INVITE
100 Trying
Signaling
180 Ringing
200 OK
ACK
Logical opening of RTP channel
Logical opening of RTCP channel
Contents
Media (UDP)
Logical opening of RTP channel
Logical opening of RTCP channel
The calling user (UAC) sends to the PS an INVITE request with the recipient SIP address :
The PS contacts the LS and provides it with all or part of the recipient SIP address : (e.g. pierre)
The PS may obtain a more precise address (e.g.
The PS sends an INVITE request to the recipient server whose address was obtain from the Localization Server localization service (e.g. lab)
The recipient UAS informs the terminal being called;
And returns to the caller PS the recipient agreement to communicate through a OK answer (code 200);
This PS returns to the calling client the recipient agreement;
The reception of the recipient agreement is acknowledged by the calling client with an ACK request;
This transaction is transmitted directly to the terminal being called.
Bye
Signaling
200 OK

14. SDP Messages in a SIP session
Each end knows
the other one IP address
Marie
Pierre
INVITE
INVITE SIP/2.0
Via: SIP/2.0/UDP :5060
Call-ID:
From: sip:
To:
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v=0
o=marie IN IP
c=IN IP
m=audio 5004 RTP/AVP 0
100 Trying
180 Ringing
SIP/ OK
Call-ID:
From: sip:
To:
Cseq 1 INVITES
Content-type: application/sdp
Content-Length: 98
v=0
o=pierre IP IP
c=IN IP
m=audio5004 RTP/AVP 0
200 OK
ACK SIP/2.0
Via: SIP/2.0/UDP :5060
Call-ID:
From: sip:
To:
Cseq 1 ACK
ACK

15. 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.
SIP Answers
1xx - Informational Messages.
2xx - Successful Responses.
3xx - Redirection Responses.
4xx - Request Failure Responses.
5xx - Server Failure Responses.
6xx - Global Failure Responses.

16. SIP in Proxy mode Location Server 2 Pierre? 3 Pierre@stanford.edu 4 1
INVITE From: 4
INVITE From: 1 6
200 OK 5
200 OK 7
ACK 8
ACK
Established session
The caller (UAC) sends an INVITE request with the recipient SIP address to the PS;
The PS contacts the LS and obtains the recipient SIP address
The PS obtains a more precise address
The PS sends an INVITE request to the recipient server whose address was provided to him by the LS localization service, The recipient PS informs the terminal being called;
The terminal being called returns to the calling PS the recipient agreement to communicate with an OK answer (code 200);
The PS returns to the caller the recipient agreement;
The reception of the recipient agreement is accepted by the caller with an ACK message;
This transaction is transmitted directly to the caller.
Proxy Server

17. SIP in Redirect mode Location server 2 Pierre? 3 Pierre@stanford.edu 1
INVITE From: 1 4
302 Moved
Contact:
Redirect Server 5
ACK 6
INVITE From:
The caller (UAC) sends an INVITE request to the Redirect server (RS) with the recipient address;
& 3. The RS contacts the Location Server, requesting and obtaining the recipient address
The RS returns to the caller the new address with a Moved answer (code 302) announcing that the recipient terminal changed PS;
The caller sends an ACK to the RS;
Then this caller sends an INVITE request to the recipient or the recipient server. This request has the same Call-ID as the first but its Cseq sequence number is higher. The recipient PS informs the caller UAS;
The recipient sends its agreement to communicate to the PS with an OK answer (code 200). The PS returns to the caller the recipient agreement;
The reception of the recipient agreement is approved by the caller with an ACK message. This transaction is transmitted directly to the recipient. 7
200 OK 8
ACK
Established session

18. SIP call example Call forward busy from B to C
Proxy Server
UA A
UA B
UA C
INVITE
INVITE
100 Trying
486 Busy
ACK
INVITE
180 Ringing
180 Ringing
200 OK
200 OK
ACK
ACK
Established session

19. SIP call example Call transfer from A to C
Proxy Server
UA A
UA B
UA C
Established session
Bye (also C)
Bye (also C)
200 OK
200 OK
INVITE (req A)
INVITE (req A)
100 Trying
180 Ringing
180 Ringing
200 OK
200 OK
ACK
ACK
Established session

20. Sending an INVITE through a SIP based system
10.ACK
Moved temporary
Moved Temporary
8. INVITE
5. ACK
Redirect Server
11. INVITE
1. INVITE
6. INVITE
Trying
Trying
Trying
Proxy Server B
Proxy Server A .
SIP Phone B
SIP Phone A

21. Call set up through a SIP based phone system
Redirect Server
Ringing
Ringing OK
17. INVITE OK
21. ACK
19. ACK
20. ACK
Proxy Server B
Proxy Server A
22. RTP .
SIP Phone B
SIP Phone A

22. Call teardown through a SIP-based system
Redirect Server
23. BYE
27. BYE
25. BYE
Trying
Trying
Trying OK OK
30. ACK
Proxy Server B
Proxy Server A .
SIP Phone B
SIP Phone A

23. Forking: sending the invite to two phones
10.ACK
Moved temporary
Moved Temporary
8. INVITE
5. ACK
Redirect Server
11. INVITE
1. INVITE
6. INVITE
Trying
Trying
Trying
Proxy Server B
Proxy Server A
12. INVITE
Trying .
SIP Phone B
SIP Phone A
SIP Phone C

24. Forking: the phones ring
Redirect Server
Ringing
Ringing
Ringing
Ringing
Ringing
Proxy Server B
Proxy Server A
Ringing .
SIP Phone B
SIP Phone A
SIP Phone C

25. Forking: Phone B answers the phone
Redirect Server OK OK OK
Proxy Server B
Proxy Server A OK
22. CANCEL .
SIP Phone B
SIP Phone A
SIP Phone C

26. SIP and Security Considerations
SIP Security
Security Mechanisms
Attack and protection
Limitations

27. SIP Security Authentication: proof of identity Confidentiality
Integrity

28. 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
Registration Hijacking is the simplest type of attack. The attacker registers himself with someone else’s identifier. In this case, the attacker receives the call instead of the real owner. Solution: authentication
Impersonating a Server:UAs sends its request to its server in order to be connected to another user agent. The domain of the server is specified in the request URI. An attacker may impersonate the remote server, and that the UA's request could be intercepted by some other party. Solution: Server Authentication
User agent trusts its own proxy server; however, the contents of its body messages should be protected from the proxy server since the proxy server can behave as Man in the middle. To solve such a problem, confidentiality, integrity, and end to end authentication should be used.

29. Attacks and protections ( cont…)
Tearing Down Sessions
Authentication of the BYE sender
Confidentiality
Denial of Service
Authentication of Register request
Tearing Down Sessions: An attacker captures some initial messages in a dialog between two called parties then inserts a BYE request into the session. When BYE is received by the called party, the session will be torn down prematurely.
Solutions:
_Authentication of the BYE sender.
_Confidentiality: The attacker won’t be able to learn the parameters of the session
Denial of Service : An attacker sends an excessive amount of traffic to the interface, which causes a particular network element unavailable.

30. 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

31. 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.
Re-INVITE is used in mid call mobility (session mobility)

32. SIP and terminal mobility
HP = home premises
FP = foreign premises

33. SIP and Terminal Mobility

34. SIP and Session mobility
Re-INVITE is used in session mobility
Session Mobility
– User moving between terminals, in “runtime”

35. SIP and Session Mobility

36. 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

37. SIP and Service Mobility
Keep same services, while mobile
Services located at home server
Call is processed by home servers using RECORD-ROUTE
Service Mobility
– Keep same services, while mobile

38. 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

39. Benefits of 3G
Mobility
Connectivity
Reachability

40. WLAN and SIP
Adds SIP based VoIP communications together with WiFi installations
Wireless access protocol: CSMA/ CA
Call control Protocol: SIP
WIFI is based on IEEE standard.

41. SIP References Columbia university Web site IETF SIP working group
IETF SIP working group
RFC 3261
SIP forum
Ubiquity Information Center : SIP center

42. Questions? ?