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Session Initiation Protocol SIP

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SIP Architecture. A signaling protocol ... Accepts SIP REGISTER requests. Indicating the ... SIP Advantages. Attempt to keep the signaling as simple as possible ... – PowerPoint PPT presentation

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Title: Session Initiation Protocol SIP


1
Session Initiation Protocol (SIP)
  • Chapter 5

2
Introduction
  • A powerful alternative to H.323
  • More flexible, simpler
  • Easier to implement
  • Advanced features
  • Better suited to the support of intelligent user
    devices
  • A part of IETF multimedia data and control
    architecture
  • SDP, RTSP (Real-Time Streaming Protocol), SAP
    (Session Announcement Protocol)

3
The Popularity of SIP
  • Originally Developed in the MMUSIC
  • A separate SIP working group
  • RFC 2543
  • Many developers
  • SIP MGCP/MEGACO
  • The VoIP signaling in the future
  • back-off
  • Test products against each other
  • Will be hosted by ETSI

4
SIP Architecture
  • A signaling protocol
  • The setup, modification, and tear-down of
    multimedia sessions
  • SIP SDP
  • Describe the session characteristics
  • Separate signaling and media streams

5
SIP Network Entities
  • Clients
  • User agent clients
  • Application programs sending SIP requests
  • Servers
  • Responds to clients requests
  • Clients and servers may be in the same platform
  • Proxy
  • Acts as both clients and servers

6
  • Four types of servers
  • Proxy servers
  • Handle requests or forward requests to other
    servers
  • Can be used for call forwarding

7
  • Redirect servers
  • Map the destination address to zero or more new
    addresses
  • Do not initiate any SIP requests

8
  • A user agent server
  • Accept SIP requests and contacts the user
  • The user responds ? an SIP response
  • A SIP device
  • E.g., an SIP-enabled telephone
  • A registrar
  • Accepts SIP REGISTER requests
  • Indicating the user is at a particular address
  • Typically combined with a proxy or redirect
    server

9
SIP Call Establishment
  • It is simple
  • A number of interim responses

10
SIP Advantages
  • Attempt to keep the signaling as simple as
    possible
  • Various pieces of information can be included
    within the messages
  • Including non-standard information
  • Enable the users to make intelligent decisions
  • The user has control of call handling
  • No need to subscribe call features

11
  • Call Completion to Busy Subscriber service

12
Overview of SIP Messaging Syntax
  • Text-based
  • Similar to HTTP
  • SIP messages
  • message start-line
  • message-header CRLF
  • message-body
  • start-line request-line status-line
  • Request-line specifies the type of request
  • The response line
  • The success or failure of a given request

13
  • Message headers
  • Additional information of the request or response
  • E.g.,
  • The originator and recipient
  • Retry-after header
  • Subject header
  • Message body
  • Describe the type of session
  • The media format
  • SDP, Session Description Protocol
  • Could include an ISDN User Part message
  • Examined only at the two ends

14
SIP Requests
  • method SP request-URI SP SIP-version CRLF
  • request-URI
  • The address of the destination
  • Methods
  • INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER
  • INVITE
  • Initiate a session
  • Information of the calling and called parties
  • The type of media
  • IAM (initial address message) of ISUP
  • ACK only the final response

15
  • BYE
  • Terminate a session
  • Can be issued by either the calling or called
    party
  • Options
  • Query a server as to its capabilities
  • A particular type of media
  • The response if sent an INVITE
  • CANCEL
  • Terminate a pending request
  • E.g., an INVITE did not receive a final response
  • REGISTER
  • Log in and register the address with a SIP server
  • all SIP servers multicast address
    (224.0.1.1750)
  • Can register with multiple servers
  • Can have several registrations with one server

16
SIP Responses
  • SIP version SP status code SP reason-phrase CRLF
  • reason-phrase
  • A textual description of the outcome
  • Could be presented to the user
  • status code, RFC 2543
  • A three-digit number
  • 1XX Informational
  • 2XX Success (only code 200 is defined)
  • 3XX Redirection
  • 4XX Request Failure
  • 5XX Server Failure
  • 6XX Global Failure
  • All responses, except for 1XX, are considered
    final
  • Should be ACKed

17
One number service
18
SIP Addressing
  • SIP URLs (Uniform Resource Locators)
  • user_at_host
  • E.g.,
  • sipcollins_at_home.net
  • sip3344556789_at_telco.net
  • Supplement the URL
  • sip3344556789_at_telco.netuserphone

19
Message Headers
  • Provide further information about the message
  • information elements
  • E.g.,
  • Toheader in an INVITE
  • The called party
  • Fromheader
  • The caling party
  • Four main categories
  • General, request, response, and entity headers
  • A list in Table 5-2
  • Mapping in Table 5-3

20
General Headers
  • Used in both requests and responses
  • Basic information
  • E.g., To, From, Call-ID, …
  • Contact
  • A URL for future communication
  • May be different from the From header
  • Requests passed through proxies

21
  • Request Headers
  • Apply only to SIP requests
  • Addition information about the request or the
    client
  • E.g.,
  • Subject
  • Priority, urgency of the request
  • Authorization, authentication of the request
    originator
  • Response Headers
  • Further information about the response
  • E.g.,
  • Unsupported, features
  • Retry-After

22
  • Entity Header
  • Session information presented to the user
  • Session description, SDP
  • The RTP payload type, an address and port
  • Content-Length, the length of the message body
  • Content-Type, the media type of the message
  • Content-Encoding, for message compression
  • Content Disposition,
  • Content-Language,
  • Allow, used in a Request to indicate the set of
    methods supported
  • Expires, the date and time

23
Example of SIP Message Sequences
  • Registration
  • Via
  • Call-ID
  • host-specific
  • Content-Length
  • Zero, no msg body
  • Cseg
  • Avoid ambiguity
  • Expires
  • TTL
  • 0, unreg
  • Contact

24
Invitation
  • A two-party call
  • Subject
  • optional
  • Content-Type
  • application/sdp

25
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26
Termination of a Call
  • Cseq
  • Has changed

27
Redirect Servers
  • An alternative address
  • 302, Moved temporarily
  • Another INVITE
  • Same Call-ID
  • Cseq

28
Proxy Servers
  • Entity headers are omitted
  • Changes the Req-URI
  • Via
  • The path
  • Loop detected, 482
  • For a response
  • The 1st Via header
  • Checked
  • removed

29
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30
Proxy state
  • Can be either stateless or stateful
  • Record-Route
  • The messages and responses may not pass through
    the same proxy
  • Use Contact
  • A Proxy might require that it remains in the
    signaling path
  • In particular, for a stateful proxy
  • Insert its address into the Record-Route header
  • The response includes the Record-Route header
  • The Record-Route header is used in the
    subsequent requests
  • The Route header the Record-Route header in
    reverse order, excluding the first proxy
  • Each proxy remove the next from the Route header

31
Forking Proxy
  • fork requests
  • A user is registered at several locations
  • branchxxx

32
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33
The Session Description Protocol
  • The message body
  • SDP, RFC 2327
  • The Structure of SDP
  • Session Level Info
  • Name
  • The originator
  • The time
  • Media Level Info
  • Media type
  • Port number
  • Transport protocol
  • Media format

34
SDP Syntax
  • A number of lines of text
  • In each line
  • fieldvalue
  • Session-level fields first
  • Media-level fields
  • Begin with media description field (m)

35
Mandatory Fields
  • v(protocol version)
  • o(session origin or creator and session id)
  • s(session name), a text string
  • t(time of the session), the start time and stop
    time
  • m(media)
  • Media type
  • The transport port
  • The transport protocol
  • The media format, an RTP payload format

36
Optional Fileds
  • i(session information)
  • A text description
  • At both session and media levels
  • u(URI of description)
  • Where further session information can be obtained
  • Only at session level
  • e(e-mail address)
  • Who is responsible for the session
  • Only at the session level
  • p(phone number)
  • Only at the session level

37
  • c(connection information)
  • Connection type, network type, and connection
    address
  • At session or media level
  • b(bandwidth information)
  • In kilobits per second
  • At session or media level
  • r(repeat times)
  • For regularly scheduled session
  • How often and how many times
  • z(timezone adjustments)
  • For regularly scheduled seesion
  • Standard time and Daylight Savings Time

38
  • k(encryption key)
  • An encryption key or a mechanism to obtain it
  • At session or media level
  • a(attributes)
  • Describe additional attributes

39
Ordering of Fields
  • Session Level
  • Protocol version (v)
  • Origin (o)
  • Session name (s)
  • Session information (i)
  • URI (u)
  • E-mail address (e)
  • Phone number (p)
  • Connection info (c)
  • Bandwidth info (b)
  • Time description (t)
  • Repeat info (r)
  • Time zone adjustments (z)
  • Encryption key (k)
  • Attributes (a)
  • Media level
  • Media description (m)
  • Media info (i)
  • Connection info (c)
  • Optional if specified at the session level
  • Bandwidth info (b)
  • Encryption key (k)
  • Attributes (a)

40
Subfields
  • Field ltvalue of subfield1gt ltvalue of subfield2gt
    ltvalue of subfield3gt.
  • Origin (o)
  • Username, the originators login id or -
  • session ID
  • A unique ID
  • Make use of NTP timestamp
  • version, a version number for this particular
    session
  • network type
  • A text string
  • IN refers to Internet
  • address type
  • IP4, IP6
  • Address, a fully-qualified domain name or the IP
    address

41
  • Connection Data
  • The network and address at which media data are
    to be received
  • Network type
  • Address type
  • Connection address
  • Media Information
  • Media type
  • Audio, video, application, data, or control
  • Port, 1024-65535
  • Format
  • List the various types of media
  • RTP/AVP payload types
  • m audio 45678 RTP/AVP 15 3 0
  • G.728, GSM, G.711

42
  • Attributes
  • Property attribute
  • asendonly
  • arecvonly
  • value attribute
  • aorientlandscape
  • rtpmap attribute
  • The use of dynamic payload type
  • artpmapltpayload typegt ltencoding namegt/ltclock
    rategt /ltencoding parametersgt.
  • mvideo 54678 RTP/AVP 98
  • artpmap 98 L16/16000/2

43
Usage of SDP with SIP
  • SIP for the establishment of multimedia sessions
  • SDP a structured language for describing the
    sessions
  • The entity header

44
Negotiation of Media
  • Fig 5-15
  • G.728 is selected
  • If a mismatch
  • 488 or 606
  • Not Acceptable
  • A Warning header
  • INVITE with multiple media streams
  • Unsupported should also be returned
  • With a port number of zero

45
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46
  • Options Method
  • Determine the capabilities of a potential called
    party

47
Usage of SIP for Features/Services
  • Personal mobility by registration
  • Can carry MIME (Multi-Purpose Internet Mail
    Extension) content
  • Text, HTML documents, an image, etc.
  • SIP address is a URL
  • Click-to-call applications
  • Supplementary Custom Local Area Signaling Service
    (CLASS) services
  • Call waiting, call forwarding, multi-party
    calling, call screening
  • Proxy-controlled QoS, IN SCP, INAP

48
Call Forwarding
  • On busy
  • 486, busy here

49
Consultation Hold
  • C0
  • An address

50
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51
SIP Extensions and Enhancements
  • RFC 2543, March 1999
  • Proposed standard, March 2002
  • SIP-T
  • Include various extensions
  • Will be enhanced considerably before it becomes
    an Internet standard

52
183 Session-Progress Message
  • The addition of a new response
  • Status code 183
  • To open a one-way media path
  • From the called party to calling party
  • Enable in-band call progress information to be
    transmitted
  • Tones or announcements
  • ACM (address complete message) of SS7
  • For SIP PSTN SIP connections
  • When a temporary media stream is needed
  • Note that alerting signal can be
  • Status code 180 (ringing)
  • The temporary media stream will be terminated
  • As soon as the called user answers

53
SIP INFO Method
  • A new SIP method
  • The transfer of information in the middle of a
    call
  • DTMF digits, account-balance information,
    mid-call signaling information (from PSTN)
  • A powerful, flexible tool to support new services

54
The SIP Supported Header
  • The Require header
  • a client indicates to a server that the server
    must support certain features
  • In responses
  • 421, extension required
  • The Supported header
  • For server to know a clients capabilities
  • Included in both requests and responses
  • BYE, CANCEL, INVITE, OPTIONS and REGISTER
  • Should not be included in the ACK

55
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56
Reliability of Provisional Responses
  • Provisional Responses
  • 100 (trying), 180 (ringing), 183 (session in
    progress)
  • Are not answered with an ACK
  • If the messages is sent over UDP
  • Unreliable
  • Lost provisional response may cause problems when
    interoperating with other network
  • 180, 183 ? Q931 alerting or ISUP ACM
  • To drive a state machine
  • E.g., a call to an unassigned number
  • ACM to create a one-way path

57
  • RSeq
  • Response seq
  • 1, when retxm
  • Rack
  • Response ACK
  • PRACK
  • Prov Resp ACK
  • Should not
  • Apply to 100

58
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59
Integration of SIP an Resource Mang
  • The signaling might take a different path from
    the media
  • Assume resource-reservation mechanisms available
    (Chapter 8)
  • A new SIP header in the INVITE
  • Resources reservation is needed
  • The user should not yet be alerted
  • But unrecognized header is ignored

60
  • Integration of Resource Management and SIP for IP
    Telephony
  • A new method, PRECONDITION-MET
  • The far-end phone will not ring until
  • Also specifies extensions to SDP
  • Can define any number of preconditions in SDP
    without revise SIP every time
  • Being sent end-to-end
  • aqos strength-tag SP direction-tag SP
    confirmation-tag
  • asecure strength-tag SP direction-tag SP
    confirmation-tag
  • If failed, could select a lower-bandwidth codec

61
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62
Interworking
  • PSTN Interworking
  • A SIP URL
  • A network gateway
  • Fig. 5-23
  • SIP to PSTN call
  • Fig. 5-24
  • PSTN to SIP call
  • PSTN SIP PSTN
  • MIME media types
  • For ISUP and QSIG

63
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64
Interworking with H.323
  • An Internet draft
  • SIP-H.323 interworking gateway

65
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66
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67
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68
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69
Summary
  • The future for signaling in VoIP networks
  • Simple, yet flexible
  • Easier to implement
  • Fit well with the media gateway control protocols

70
Reference Architecture of R00
71
Support of Roaming Subscribers
  • via the service platform in the Home Network
  • via an external service platform

72
UE Accessing IM Subsystem Services
  • In the visited network

73
Public and Private Identities
  • One private user identity
  • IMSI within a NAI
  • One or more public user identities

74
Roles of CSCF
  • Call State Control Function
  • SIP proxy servers
  • Proxy CSCF
  • The first contact point within the IM subsystem
  • Forward the SIP register requests from the UE to
    an I-CSCF
  • Forward the SIP messages from the UE to the
    S-CSCF
  • CDR
  • FFS
  • Authorization, QoS management, security

75
  • Interrogating CSCF
  • The contact point within an operators network
  • Assign an S-CSCF to a user performing SIP Reg
  • Obtain from HSS the address of the S-CSCF
  • Forward SIP messages to the S-CSCF
  • Forward SIP messages to the MGCF
  • CDR
  • FFS
  • Inter-operator security

76
  • Serving CSCF
  • Perform the session control services
  • A registrar
  • Interaction with Services Platforms
  • On behalf of an originating endpoint
  • On behalf of an destination endpoint
  • CDR
  • FFS
  • Security issues

77
Registration Information Flow
78
Registration
79
Call flow
80
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81
Different Kinds of CSCFs
82
Interwork with PSTN
83
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