RFC 3372 Session Initiation Protocol for Telephones SIPT

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RFC 3372 Session Initiation Protocol for Telephones (SIP-T): Context and Architectures Presented by Zhi-Hong

RFC 3372 Session Initiation Protocol for Telephones (SIP-T): Context and Architectures Presented by Zhi-Hong Guo Instructed by Assistant Professor Quincy Wu

Introduction l l l It is vital for a SIP telephony network to interwork

Introduction l l l It is vital for a SIP telephony network to interwork with the PSTN. An important characteristic of any SIP telephony network is feature transparency with respect to the PSTN. Another important characteristic of a SIP telephony network is routability of SIP requests.

Introduction (cont. ) l l The SIP-T effort provides a framework for the integration

Introduction (cont. ) l l The SIP-T effort provides a framework for the integration of legacy telephony signaling into SIP messages. SIP-T provides the above two characteristics through techniques known as 'encapsulation' and 'translation' respectively.

PSTN architecture l Three components: – Customer premises equipment (CPE) l – The transmission

PSTN architecture l Three components: – Customer premises equipment (CPE) l – The transmission facilities l – Telephone set, private branch exchanges (PBX) Trunks and subscriber lines The switching system l Central offices (CO), tandems

PSTN architecture (cont. )

PSTN architecture (cont. )

Call setup and release l l A call requires a communications circuit between two

Call setup and release l l A call requires a communications circuit between two subscribers. The setup and release of connection is triggered by signals.

Signaling System No. 7 (SS 7) SS 7 is a global standard for telecommunications

Signaling System No. 7 (SS 7) SS 7 is a global standard for telecommunications defined by ITU-T. l SS 7 follows ISO 7 layer architecture. l

SS 7 Protocol Stack l ISUP: For call control, it defines the protocol and

SS 7 Protocol Stack l ISUP: For call control, it defines the protocol and procedures used to set-up, manage and release trunk circuits. – Ex: Call setup or release

Basic call setup IAM: Initial Address Message ACM: Address Complete Message ANM: Answer Message

Basic call setup IAM: Initial Address Message ACM: Address Complete Message ANM: Answer Message

Basic call release REL: Release Message RLC: Release Complete Message

Basic call release REL: Release Message RLC: Release Complete Message

Encapsulation and translation l l Encapsulation: Some of SS 7 ISUP messages are encapsulated

Encapsulation and translation l l Encapsulation: Some of SS 7 ISUP messages are encapsulated into the SIP message body in order that information necessary for services is not discarded in the SIP request. Translation: Some critical SS 7 ISUP messages are translated into the corresponding SIP headers in order to determine how the SIP request will be routed.

SIP-T flows l l SIP Bridging (PSTN - IP - PSTN) PSTN origination -

SIP-T flows l l SIP Bridging (PSTN - IP - PSTN) PSTN origination - IP termination : The terminator has no use for the encapsulated ISUP and ignores it. IP origination - PSTN termination: The terminating gateway only performs translation. IP origination - IP termination: This is a case for pure SIP.

SIP Bridging (PSTN - IP - PSTN) PSTN SIP Proxy Vo. IP Network MGC

SIP Bridging (PSTN - IP - PSTN) PSTN SIP Proxy Vo. IP Network MGC SIP PSTN MGC SS 7 ISUP PSTN Phone SIP Proxy SS 7 ISUP PSTN Phone

Call-flow PSTN Phone IAM ACM ANM REL RLC MGC#1 SIP Proxy INVITE 100 TRYING

Call-flow PSTN Phone IAM ACM ANM REL RLC MGC#1 SIP Proxy INVITE 100 TRYING 18 X 200 OK ACK CONVERSATION BYE 200 OK MGC#2 PSTN Phone IAM ACM ANM REL RLC

PSTN origination - IP termination SIP Proxy Vo. IP Network SIP Proxy SIP PSTN

PSTN origination - IP termination SIP Proxy Vo. IP Network SIP Proxy SIP PSTN MGC SS 7 ISUP PSTN Phone SIP Proxy SIP Phone

Call-flow PSTN Phone IAM ACM ANM REL RLC MGC SIP Phone SIP Proxy INVITE

Call-flow PSTN Phone IAM ACM ANM REL RLC MGC SIP Phone SIP Proxy INVITE 100 TRYING 18 X 200 OK ACK CONVERSATION BYE 200 OK INVITE 18 X 200 OK ACK BYE 200 OK

IP origination - PSTN termination PSTN SIP Proxy Vo. IP Network MGC SIP Proxy

IP origination - PSTN termination PSTN SIP Proxy Vo. IP Network MGC SIP Proxy SIP Phone SIP Proxy SS 7 ISUP PSTN Phone

Call-flow SIP Proxy SIP Phone INVITE 100 TRYING 18 X 200 OK ACK BYE

Call-flow SIP Proxy SIP Phone INVITE 100 TRYING 18 X 200 OK ACK BYE 200 OK MGC INVITE 100 TRYING 18 X 200 OK PSTN Phone IAM ACM ANM ACK CONVERSATION BYE 200 OK REL RLC

Components of the SIP-T Protocol l Core SIP: defined by RFC 3261 Encapsulation: SIP-T

Components of the SIP-T Protocol l Core SIP: defined by RFC 3261 Encapsulation: SIP-T uses multipart MIME bodies to enable SIP messages to contain multiple payloads (SDP, ISUP, etc. ). Translation: – ISUP SIP message mapping – ISUP parameter-SIP header mapping

ISUP-SIP message mapping l l l IAM<->INVITE ACM<->18 X REL<->BYE

ISUP-SIP message mapping l l l IAM<->INVITE ACM<->18 X REL<->BYE

ISUP parameter-SIP header mapping l l l Called Party Number <-> To、Request-URI The headers

ISUP parameter-SIP header mapping l l l Called Party Number <-> To、Request-URI The headers of a SIP request (especially an INVITE) may be transformed by intermediaries, and that consequently, the SIP headers and encapsulated ISUP bodies come to express conflicting values effectively. The SIP headers should take precedence over the ISUP as the contents of SIP headers may be updated in routing within the IP network.

SIP content negotiation l l Traditionally in SIP, if the terminating device does not

SIP content negotiation l l Traditionally in SIP, if the terminating device does not support a multipart payload and/or the ISUP MIME type , it would then reject the SIP request with a 415 Unsupported Media Type specifying the media types it supports. The originator would have to re-send the SIP request after stripping out the ISUP payload.

SIP content negotiation (cont. ) l l It is highly desirable to have a

SIP content negotiation (cont. ) l l It is highly desirable to have a mechanism by which the originator could signify which bodies are required and which are optional so that the terminator can silently discard optional bodies that it does not understand. This is upon the terminator having support for a Content-type of multipart/mixed and access to the Content-Disposition header to express criticality.

Support for ISUP is optional l UA 2 accepts the INVITE irrespective of whether

Support for ISUP is optional l UA 2 accepts the INVITE irrespective of whether it can process the ISUP. UA 1 UA 2 INVITE--> (Content-type: multipart/mixed; Content-type: application/sdp; Content-disposition: session; handling=required; Content-type: application/isup; Content-disposition: signal; handling=optional; ) <--18 x

Support for ISUP is preferred l UA 2 does not support the ISUP and

Support for ISUP is preferred l UA 2 does not support the ISUP and rejects the INVITE with a 415 Unsupported Media Type. UA 1 strips off the ISUP and re-sends the INVITE with SDP only. UA 1 UA 2 INVITE--> (Content-type: multipart/mixed; Content-type: application/sdp; Content-disposition: session; handling=required; Content-type: application/isup; Content-disposition: signal; handling=required; ) <--415 (Accept: application/sdp) ACK--> INVITE--> (Content-type: application/sdp) <--18 x

Support for ISUP is mandatory l UA 2 does not support the ISUP and

Support for ISUP is mandatory l UA 2 does not support the ISUP and rejects the INVITE with a 415 Unsupported Media type. UA 1 then directs its request to UA 3. UA 1 UA 2 INVITE--> (Content-type: multipart/mixed; Content-type: application/sdp; Content-disposition: session; handling=required; Content-type: application/isup; Content-disposition: signal; handling=required; ) <--415 (Accept: application/sdp) ACK-->

Support for ISUP is mandatory (cont. ) UA 1 UA 3 INVITE--> (Content-type: multipart/mixed;

Support for ISUP is mandatory (cont. ) UA 1 UA 3 INVITE--> (Content-type: multipart/mixed; Content-type: application/sdp; Content-disposition: session; handling=required; Content-type: application/isup; Content-disposition: signal; handling=required; )