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Kommunikatsiooniteenuste arendusIRT0080Loeng 4

Avo Otstelekommunikatsiooni õppetool,

TTÜ raadio- ja sidetehnika inst.avo.ots@ttu.ee

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SIP Overview(1) - History• SIP ( Session initiation protocol)

• Work began in 1995

• Developed in SIP Working Group in IETF

• Proposed standard RFC2543, February 1999

• New Version of SIP - RFC 3261, June 2002

• Split from MMUSIC (Multiparty Multimedia Session Control), Sep. 1999

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SIP Overview(2) - Main Functions

SIP is an application-layer control (signaling) protocol for :– initiating sessions

Find the user’s current location Carry session descriptions

– Modifying sessions– Terminating sessions

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SIP Overview(3) - What is a session?

• Refers to an active connection between two computers for the purpose of communicating and transferring information Internet telephone calls multimedia conferences Instant Messaging However it’s not limited to the above

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SIP Overview(4) • SIP is similar to HTTP :

both use a request/response model both are text-based much of SIP’s message and header field

syntax is identical to HTTP/1.1.

• Works independent of the underlying network

transmission protocol and indifferent to media

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SIP Overview(5) • Scalability

Functionality such as proxying, redirection, location, or registration can reside in

different physical servers. Distributed functionality allows new processes

to be added without affecting other components.

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SIP Overview(6) • Interoperability

An open standard Can implement to communicate with other

SIP based products

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SIP Overview(7) • Mobility

The user can be using a PC at work, PC at home ,wireless phone, IP phone, or regular phone.

Users must register/Update their current location.

Supports user mobility by proxying or redirecting requests to a user’s current location.

Proxy servers will forward calls to the user’s current location.

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Integration with IETF Protocols• SIP forms only part of an overall IP telephony system

• Other IETF protocol standards are used to build a fully functioning VoIP system.

• example: RSVP - to reserve network resources. RTP (Real Time Transport Protocol) - to transport real

time data RTSP (Real Time Streaming Protocol) - for controlling

delivery of streaming media. RTCP (Real Time Transport Control Protocol) – provides

information about the quality of the communication.

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Related Protocols

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SIP Capabilities(1/2)• Determine location of target points – Support address resolution, name mapping, call redirection• Determine media capabilities – SIP uses Session Description Protocol (SDP) for this• Determine availability – returns a message why the remote party cannot be contacted• Establish a session between end points – also support mid

call changes, changes of media characteristics or codec

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SIP Capabilities(2/2)• Permits interaction between devices via

signaling messages

• These messages can: Register a user with a system Invite a user to join an interactive session Establish a media stream between 2 or more

end

points

• Terminate a session

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SIP Network Architecture

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SIP Component – User Agent• User Agent Client (UAC)- Initiate SIP

Request

• User Agent Server (UAS)-Accepts or rejects call

• Phones – acts as UAC or UAS

• Implemented in Hardware or Software

Components

• Includes softphones, sip ip phones

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User Agents

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SIP Component – Proxy Server• Heart of SIP network that contains all

service logic.

• Receives SIP messages, forwards to next SIP UAS.

• Interprets, rewrites or translates a request message before forwarding it.

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Scenario with Proxy Server

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SIP Component – Redirect Server• Return routing information to the

originating endpoint.

• Does not accept or terminate calls

• Does not initiate its own SIP request

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Scenario with Redirect Server

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SIP Component – Registrar

• Accept registration requests from users

• Maintains user’s whereabouts at a

Location Server

• Typically co-located with a proxy server

or a redirect server and may offer

location services

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Scenario with Registrar

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Invitation for SIP Proxy Server

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Invitation for Redirect Server

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SIP Addressing• The SIP address is identified by a SIP

URL,in the format: user@host.

• Email-like identifier of the form

• Examples of SIP URLs:– sip:hostname@ntut.edu.tw– sip:hostname@192.168.10.1– sip:14083831088@vovida.org

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SIP Messages• The message syntax is identical to

HTTP/1.1(RFC2616).

• Two kinds of SIP messages

SIP-message = Request | Response

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SIP Messages - Requests• INVITE – Initiates a call by inviting user to

participate in session.

• ACK - Confirms that the client has received a final response to an INVITE request.

• BYE - Indicates termination of the call.

• CANCEL - Cancels a pending request.

• REGISTER – Registers the user agent.

• OPTIONS – Used to query the capabilities of a server.

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SIP Messages - Responses

• 1xx - Informational Responses– 180 ringing

• 2xx - Successful Responses– 200 OK

• 3xx - Redirection Responses– 302 Moved Temporarily

• 4xx - Request Failure Responses.– 404 Not Found

• 5xx - Server Failure Responses.– 503 Service Unavailable

• 6xx - Global Failures Responses.– 600 Busy Everywhere

• All responses, except for 1XX, are considered final responses

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Message Headers

• Provide further information about the message• E.g.,

– To : header in an INVITE• The called party

– From : header• The calling party

• Four main categories– General, Request, Response, and Entity headers

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Header Fields• General Header

– Be applied to both request and response messages

• Entity Header– Define information about the message body

• Request Header– Allow the client to pass additional information about

the request

• Response Header– Allow the server to pass additional information

about the response

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SIP Headers

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SIP Headers – From• A field required in all requests and response

messages

• Provides identity of request’s initiator

• Example: – From: Laura Brown <sip:Laura.Brown@university.com>

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SIP Headers – To• Provides identity of the intended

recipient of the request

• Example:– To: Bob Johnson <sip:Bob.Johnson@company.com>;

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SIP Headers – Call-ID(1/2)

• Provides a globally unique identifier to

distinguish specific invitations

• Typically uses a 32-bit cryptographically

random numbers

• Example:– Call-ID: ges456fcdw21lkfgte12ax@workstation1234.university.com

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SIP Headers – Call-ID(2/2)

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SIP Headers – Contact(1/2)• A Contact header provides a URL where

the user can be reached directly.

• This feature is important because it offloads SIP servers that do not need to be in the signalling path after routing the first INVITE.

• Example:– Contact: Bob Johnson<sip:Bob@140.124.181.227>

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SIP Headers – Contact(2/2)

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SIP Header - Via• History of message’s path through

network(s)

• Helps to prevent looping and ensures

replies route back to originator

• Indicates the used transport protocol, ip

address and port of sender• Via: SIP/2.0/UDP 192.168.6.21:5060

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SIP Header – Content-Type• Provides information about media type of

message body

• Content-Type: application/sdp

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SIP Header – CSeq(1/2)• CSeq or command sequence:• Needed in both request messages as well as response messages• Need to increment this when a user with the same Call-ID wants to send different SIP methods or content ( Except ACK and Cancel)• When sending responses to requests, CSeq should be the same• The numerical part of the Cseq is used to order different

requests within the same session. • CSeq: 1 INVITE

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SIP Header – CSeq(2/2)

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An example of SIP Request

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An example of SIP Response

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Reliable Transmission of Responses(1/2)

• Final responses are transmitted reliably between server and client, using retransmissions or a reliable transport protocol to ensure delivery.

• Provisional responses may either be received by the client or be lost in the network.

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Reliable Transmission of Responses(2/2)

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Reliable Transmission of Requests(1/2)

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Reliable Transmission of Requests(2/2)

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Summary

• SIP is gaining acceptance in the industry• Open Source projects are taking the lead in SIP implementations• New generation of services are already being offered• Staying with single a vender solution is currently

workable• Interoperability between vender products is still in early stages

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Reference• RFC 2543(SIP/1.0)

• RFC 3261(SIP/2.0)

• RFC 2327(SDP)