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SIP-SDP-2

Apr 05, 2018

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    Lecture No.9

    SIP - requests and responses,header fields and extensions

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    Methods

    INVITE :Establishes a session

    ACK :Confirms an INVITE request, 3-way hand-shaking

    BYE :Ends a established session

    CANCEL :Cancels establishing of a session

    REGISTER :Communicates user location (host name, IP)

    OPTIONS :Communicates information about thecapabilities of the calling and receiving SIP phones

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    Miroslav Voznak, lecture on SIP 3

    Methods extension

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    Responses 1xx informational responses, such as 180, which means

    ringing 200 - 699 final response

    2xx success responses, such as 200 OK

    3xx redirection responses

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    Responses

    4xx request failures 5xx server errors

    6xx global failures

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    TransactionTransaction

    a request and all responses

    old type hash To,From,RURI, CSeq

    new parameter branchinVia

    z9hG4bK is beginning of astring branch(new way oftransaction determination)

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    Dialog

    Dialog

    Call-ID, From tag, To tag

    CSeq identifies atransaction in dialog

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    Miroslav Voznak, lecture on SIP 8

    Registration

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    Miroslav Voznak, lecture on SIP 9

    Registration- bundling of user URI and device URI (From and Contact)

    - Expires is a time period of registration (Expires=0 , cancellation)

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    Miroslav Voznak, lecture on SIP 10

    Redirection

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    Miroslav Voznak, lecture on SIP 11

    SIP message

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    Miroslav Voznak, lecture on SIP 12

    Session Description Protocolv=0o=ja 987504994 987504994 IN IP4 1.2.3.4s=Hovor 1c=IN IP4 1.2.3.4t=3196493794 3196497394m=audio 10000 RTP/AVP 0 22a=rtpmap:0 aplication/g711a=rtpmap:22 aplication/g723.1

    SDP

    v = Version number (ignored by SIP)o = Session Origin used by SIP

    s = Subjectc = Connection Data (IN =internet, IP4 = IPv4, IP Address)t = Time (ignored by SIP)m = Media (type, port, RTP/AVP Profile)a = Attribute (profile, codec, sampling rate)

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    INVITEINVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP pc33.atlanta.com:5060;branch=z9hG4bK776asdhdsTo: Bob

    From: Alice ;tag=1928301774

    Call-ID: a84b4c76e66710

    CSeq: 314159 INVITE

    Contact:

    Content-Type: application/sdpContent-Length: 142

    (SDP part)

    the first line contains :

    method name: INVITE

    Request-URI (next hope is sip:[email protected])

    SIP version number: SIP/2.0

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    SIP viaINVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP proxy.stockholm.se:5060;branch=82.1

    Via: SIP/2.0/UDP pc33.atlanta.com:5060;branch=z9hG4bK776asdhds

    To: Bob

    From: Alice ;tag=1928301774

    Call-ID: a84b4c76e66710

    CSeq: 314159 INVITE

    Contact:

    Content-Type: application/sdp

    Content-Length: 142

    (SDP part)

    Via field:

    insterted records point to the used route, Proxy serversinsert next records

    response should use the same route as a request

    branch enables to detect the loops

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    Dialog (Call-leg)INVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP pc33.atlanta.com:5060;branch=z9hG4bK776asdhds

    To: Bob From: Alice ;tag=1928301774

    Call-ID: a84b4c76e66710

    CSeq: 314159 INVITE

    Contact:

    Content-Type: application/sdp

    Content-Length: 142

    (SDP part)

    Dialog the same information during the same call-leg are in therecords:

    To , From a Call-ID To a From specifies logical address of endpoints of connection

    Call-ID unique identifier during the dialog

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    CSeqINVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP pc33.atlanta.com:5060;branch=z9hG4bK776asdhds

    To: Bob

    From: Alice ;tag=1928301774

    Call-ID: a84b4c76e66710

    CSeq: 314159 INVITE

    Contact:

    Content-Type: application/sdp

    Content-Length: 142

    (SDP part)

    CSeq

    every new request increments CSeq repeated request has the sameCSeq

    all responses regarding the one request has the same CSeq

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    ContactINVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP pc33.atlanta.com:5060;branch=z9hG4bK776asdhds

    To: Bob

    From: Alice ;tag=1928301774

    Call-ID: a84b4c76e66710

    CSeq: 314159 INVITE

    Contact:

    Content-Type: application/sdp

    Content-Length: 142

    Contact

    contains SIP URI for direct communication, if SIP Proxy

    doesnt insertRecord-Route field than nexttransctions can be routed directly

    field Contact is also stated in response 200 OK

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    Content type, LengthINVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP pc33.atlanta.com:5060;branch=z9hG4bK776asdhdsTo: Bob

    From: Alice ;tag=1928301774

    Call-ID: a84b4c76e66710

    CSeq: 314159 INVITE

    Contact:

    Content-Type: application/sdpContent-Length: 142

    Content-Type describes the type of body (SDP)

    Content-Length includes the size of content (in bytes), thevalue 0means without content

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    Max ForwardsINVITE sip:[email protected] SIP/2.0

    Via: SIP/2.0/UDP pc33.atlanta.com:5060;branch=z9hG4bK776asdhds

    Max-Forwards: 30To: Bob

    From: Alice ;tag=1928301774

    Call-ID: a84b4c76e66710

    CSeq: 314159 INVITE

    Contact:

    Content-Type: application/sdpContent-Length: 142

    Max-Forwards

    decreases with every forwarded request at the Proxy

    if its value is 0 than is sent 483 Too Many Hops

    stateless Proxy detects a loop by field Max-Forwards

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    INVITE and response 200 OK

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    Tagging

    in the case of forking there would be difficult to resolve whoanswered

    UAS adds a tag into the response

    To: sip:[email protected];tag=a48s

    next request contains tag inserted into field To

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    Miroslav Voznak, lecture on SIP 22

    Response routing the reguests are routed according to Request URI (RURI)

    and the responses according to field Via

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    Forking

    the first 200 OK

    confirmed by ACK

    the rest obtains487 CancelledINVITE

    R d R ti

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    Miroslav Voznak, lecture on SIP 24

    Record Routing- Record route field is inserted by SIP Proxy not only intothe header of method (request) but into the final responseas well- a next transaction is routed via the elements stated in therecord route

    SIP t id

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    SIP trapezoid

    - Outbound Proxy has to find a SIP Proxy which is able toserve the request

    - SIP Proxy serves one or more domains (multidomain Proxy)- destination SIP Proxy can be find via DNS (SRV record in

    DNS resolves serving SIP Proxy) or by static mapping

    ENUM

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    ENUM

    - SIP addressing is based on SIP URI

    - the number (E.164) can be resolved via DNS whereinNAPTR record contains a regular expression which is appliedto ENUM query and the result of this operation is SIP URI (ora list of URIs)

    - SIP URI contains a host (users domain), via SRV recordis resolved serving SIP Proxy

    - IP address of SIP proxy is resolved via A record in DNS

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    Call Setup

    Termination of established session

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    Termination of established session- Bye

    d bli hi i

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    and establishing session Cancel

    - two requests INVITE and CANCEL open two transactions

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    SIP and PSTN Interworking

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    Presence

    UA gets status and presence

    information which correspondsto users of server

    Users are subscribed to users

    presence

    NOTIFY is sent on an event (achange of the state)

    SUBCRIBE creates a dialogwherein the NOTIFY is sent ona change of the state

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    Subscribe a Notify

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    Instant Messaging

    Method MESSAGE isused for IM

    text is in body ofmessage

    request Messagedoesnt create a dialog

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    Miroslav Voznak, lecture on SIP 34

    NAT Issue: clients behind NAT insert private IP

    into signaling in SIP fields Via, From,Contact and SDP c-line (contact)

    NAT

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    Miroslav Voznak, lecture on SIP 35

    NAT Solution: an intervention on application layer

    ALG (Application Layer Gateway), ALG workstogether with NAT and makes rewriting inapplication layer

    NAT s pport

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    NAT support STUN, TURN, ICE help on the Client Side

    (these protocols help to find out public IP)

    Session Border Controller (SBC) help on

    the Provider Side, clients behind NATcommunicate through a border device

    SIP Proxy with s Media Relay support (RTPProxy) in order to ensure signaling and mediarouting on the same IP

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    Thank you for your attention

    [email protected]

    mailto:[email protected]:[email protected]:[email protected]:[email protected]