1. Session Initiation Protocol

2. Lecture plan:
1. SIP Basics
2. SIP network architecture
3. Message structure
4. Commands (requests)
5. Replies
6. SIP in the NGN

3. Literature VoIP books:
Daniel Collins. Carrier Grade Voice Over IP (second edition) - McGraw-Hill Professional, 2002
Theodore Wallingford. Switching to VoIP - O'Reilly Media, 2005
Jim Van Meggelen. Asterisk: The Future of Telephony - O'Reilly Media, 2005
SIP books:
Gonzalo Camarillo. SIP Demystified - McGraw-Hill Professional, 2001
Alan B. Johnston. SIP: Understanding the Session Initiation Protocol, Second Edition - Artech House, 2003
3. Rfc 3261 and so on…
RTP books:
1. Colin Perkins. RTP: Audio and Video for the Internet - Addison-Wesley Professional, 2003

4. SIP Basics
Session Initiation Protocol (SIP) – is an application layer protocol and is intended for the organization, modification and completion of communication session: multimedia conferences, telephone calls and distribution of multimedia information.
This protocol is designed by IETF committee (Internet Engineering Task Force); protocol specifications are in RFC 2543.

5. SIP is based on the following principles:
• Personal mobility for users. A unique identifier is assigned to user, and the network provides communications services to him, regardless of where it is;
• scalability of the network (characterized primarily by the possibility of increasing the number of network elements in its expansion);
• extensibility of the protocol is characterized by the ability to supplement the protocol with new features with the introduction of new services and adapt it to different applications
Another important principle of SIP is its independence of transport technologies. As transport protocols can be used UDP or TCP.

6. Place in IP-model

7. SIP network architecture
Main functional elements:
Redirect Server
Registrar/ Location Server
PSTN GW
Proxy Server
User Agent

8. Network elements necessary for SIP
Server – an application that allows the system to accept requests, execute them and send replies.
Types of servers:
SIP Proxy Server
transmits signaling – works as a client and a server
uses the principle of transaction
do not store any data about the connection
performs routing (routing) – defines who (UA / proxy / redirect) want to send messages
programmability provides routing
provides separation (Forking) posts – may require multiple destinations simultaneously or sequentially
SIP Redirect Server
redirects calls to other servers or directly to the called user
SIP Registrar
accepts registration requests from users
stores information about a visitor
acts as a gatekeeper (Gateway) in the direction of the PSTN

9. Network elements necessary for SIP
User Agent (UA) - an application that consists of two parts:
1. User agent client, UAC - an application that initiates the SIP-request (request);
2. User agent server, UAS - application communicate with the user after the SIP-request, return response (response) at the request of the user.
User Agent
User Agent
method (request)
UAC
UAS
response
UAS
UAC

10. SIP message structure
A SIP message is either a request from a client to a server, or a response from a server to a client.
Both Request and Response messages use the generic-message format.
Both types of messages consist of a start-line, one or more header fields (also known as "headers"), an empty line (i.e., a line with nothing preceding the carriage-return line-feed (CRLF)) indicating the end of the header fields, and an optional message-body.
Start Line
Headers
Empty Line
Message body

11. Examples request INVITE sip:nekdo@iskratel.si SIP/2.0
To: Nekdo
From:
Call-Id: a84b4c76e66710
Content-Type: application/sdp
Content-Length: 142
response
SIP/ Trying
From: Jaz

12. Methods:
"INVITE"
"ACK"
"OPTIONS"
"BYE"
"CANCEL"
"REGISTER"

13. The INVITE method indicates that the user or service is being invited to participate in a session. The message body contains a description of the session to which the callee is being invited. For two-party calls, the caller indicates the type of media it is able to receive and possibly the media it is willing to send as well as their parameters such as network destination. A success response MUST indicate in its message body which media the callee wishes to receive and MAY indicate the media the callee is going to send.
A server MAY automatically respond to an invitation for a conference the user is already participating in, identified either by the SIP Call-ID or a globally unique identifier within the session description, with a 200 (OK) response.
The ACK request confirms that the client has received a final response to an INVITE request. (ACK is used only with INVITE requests.) 2xx responses are acknowledged by client user agents, all other final responses by the first proxy or client user agent to receive the response.
OPTIONS The server is being queried as to its capabilities. A server that believes it can contact the user, such as a user agent where the user is logged in and has been recently active, MAY respond to this request with a capability set.
The user agent client uses BYE to indicate to the server that it wishes to release the call. A BYE request is forwarded like an INVITE request and MAY be issued by either caller or callee. A party to a call SHOULD issue a BYE request before releasing a call ("hanging up"). A party receiving a BYE request MUST cease transmitting media streams specifically directed at the party issuing the BYE request.
The CANCEL request cancels a pending request with the same Call-ID, To, From and CSeq (sequence number only) header field values, but does not affect a completed request. (A request is considered completed if the server has returned a final status response.)
A user agent client or proxy client MAY issue a CANCEL request at any time.
A client uses the REGISTER method to register the address listed in the To header field with a SIP server.

14. Replies:
SIP/2.0 allows 6 values for the first digit:
1xx: Informational -- request received, continuing to process the
request;
2xx: Success -- the action was successfully received, understood, and
accepted;
3xx: Redirection -- further action needs to be taken in order to
complete the request;
4xx: Client Error -- the request contains bad syntax or cannot be
fulfilled at this server;
5xx: Server Error -- the server failed to fulfill an apparently valid
6xx: Global Failure -- the request cannot be fulfilled at any server.

15. Registration Main scenarios Location service Registrar User REGISTER
store
200 (OK)

16. SIP – Call (redirect mode)
Main scenarios
Location
service
Пользователь A
Proxy
Пользователь B
Registrar
INVITE
query
resp
301 (Moved)
ACK

17. SIP – Call (redirect mode)
Main scenarios
Location
service
Пользователь A
Proxy
Пользователь B
Registrar
INVITE
180 (Ringing)
200 (OK)
ACK

18. Main scenarios Пользователь A Пользователь B Proxy 1 Proxy 2 INVITE
100 (Trying)
INVITE
100 (Trying)
180 (Ringing)
180 (Ringing)
180 (Ringing)
200 (OK)
200 (OK)
200 (OK)
ACK

19. Перенос данных (media session)
Main scenarios
Пользователь A
Пользователь B
Proxy 1
Proxy 2
ACK
Перенос данных (media session)
BYE
200 (OK)

20. Main scenarios
INVITE
100
INVITE
183
PRACK
200
180
180
PRACK
200
200
ACK
200
ACK
BAY
200
BAY
200

21. A next-generation network (NGN) is a packet-based network which can provide services including Telecommunication Services and able to make use of multiple broadband, quality of Service-enabled transport technologies and in which service-related functions are independent from underlying transport-related technologies. It offers unrestricted access by users to different service providers. It supports generalized mobility which will allow consistent and ubiquitous provision of services to users.
For voice applications one of the most important devices in NGN is a Softswitch – a programmable device that controls Voice over IP (VoIP) calls. It enables correct integration of different protocols within NGN. The most important function of the Softswitch is creating the interface to the existing telephone network, PSTN, through Signalling Gateways and Media Gateways.
A Media gateway is a translation device or service that converts digital media streams between disparate telecommunications networks such as PSTN, SS7, Next Generation Networks.