📄 rfc3372.txt
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By encapsulating ISUP information in the SIP signaling, a SIP network
can ensure that no SS7 information that is critical to the
instantiation of features is lost when SIP bridges calls between two
segments of the PSTN.
That much said, if only the exchange of ISUP between gateways were
relevant here, any protocol for the transport of signaling
information may be used to achieve this, obviating the need for SIP
and consequently that of SIP-T. SIP-T is employed in order to
leverage the intrinsic benefits of utilizing SIP: request routing and
call control leveraging proxy servers (including the use of forking),
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RFC 3372 SIP-T September 2002
ease of SIP service creation, SIP's capability negotiation systems,
and so on. Translation of information from the received ISUP message
parameters to SIP header fields enables SIP intermediaries to
consider this information as they handle requests. SIP-T thus
facilitates call establishment and the enabling of new telephony
services over the IP network while simultaneously providing a method
of feature-rich interconnection with the PSTN.
Finally, the scenario in Figure 2 is just one of several flows in
which SIP-T can be used - voice calls do not always both originate
and terminate in the PSTN (via gateways); SIP phones can also be
endpoints in a SIP-T session. In subsequent sections, the following
possible flows will be further detailed:
1. PSTN origination - PSTN termination: The originating gateway
receives ISUP from the PSTN and it preserves this information
(via encapsulation and translation) in the SIP messages that it
transmits towards the terminating gateway. The terminator
extracts the ISUP content from the SIP message that it receives
and it reuses this information in signaling sent to the PSTN.
2. PSTN origination - IP termination: The originating gateway
receives ISUP from the PSTN and it preserves this ISUP
information in the SIP messages (via encapsulation and
translation) that it directs towards the terminating SIP user
agent. The terminator has no use for the encapsulated ISUP and
ignores it.
3. IP origination - PSTN termination: A SIP phone originates a VoIP
call that is routed by one or more proxy servers to the
appropriate terminating gateway. The terminating gateway
converts to ISUP signaling and directs the call to an appropriate
PSTN interface, based on information that is present in the
received SIP header.
4. IP origination - IP termination: This is a case for pure SIP.
SIP-T (either encapsulation or translation of ISUP) does not come
into play as there is no PSTN interworking.
3. SIP-T Flows
The follow sections explore the essential SIP-T flows in detail.
Note that because proxy servers are usually responsible for routing
SIP requests (based on the Request-URI) the eventual endpoints at
which a SIP request will terminate is generally not known to the
originator. So the originator does not select from the flows
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RFC 3372 SIP-T September 2002
described in this section, as a matter of static configuration or on
a per-call basis - rather, each call is routed by the SIP network
independently, and it may instantiate any of the flows below as the
routing logic of the network dictates.
3.1 SIP Bridging (PSTN - IP - PSTN)
********************
*** ***
* *
* ------- *
* |proxy| *
* ------- *
|---| |---|
/|MGC| VoIP Network |MGC|\
/ --- --- \
/ * * \
/ * ------- * \
/ * |proxy| * \
-------- * ------- * ---------
| PSTN | *** *** | PSTN |
-------- ********************* ---------
Figure 2: PSTN origination - PSTN termination (SIP Bridging)
A scenario in which a SIP network connects two segments of the PSTN
is referred to as 'SIP bridging'. When a call destined for the SIP
network originates in the PSTN, an SS7 ISUP message will eventually
be received by the gateway that is the point of interconnection with
the PSTN network. This gateway is from the perspective of the SIP
protocol the user agent client for this call setup request.
Traditional SIP routing is used in the IP network to determine the
appropriate point of termination (in this instance a gateway) and to
establish a SIP dialog and begin negotiation of a media session
between the origination and termination endpoints. The egress
gateway then signals ISUP to the PSTN, reusing any encapsulated ISUP
present in the SIP request it receives as appropriate.
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A very elementary call-flow for SIP bridging is shown below.
PSTN MGC#1 Proxy MGC#2 PSTN
|-------IAM------>| | | |
| |-----INVITE---->| |
| | | |-----IAM----->|
| |<--100 TRYING---| |
| | | |<----ACM------|
| |<-----18x-------| |
|<------ACM-------| | | |
| | | |<----ANM------|
| |<----200 OK-----| |
|<------ANM-------| | | |
| |------ACK------>| |
|====================Conversation=================|
|-------REL------>| | | |
|<------RLC-------|------BYE------>| |
| | | |-----REL----->|
| |<----200 OK-----| |
| | | |<----RLC------|
| | | | |
3.2 PSTN origination - IP termination
********************
*** ***
* *
* *
* *
* *
|----| |-----|
/|MGC | VoIP Network |proxy|\
/ ---- ----- \
/ * * \
/ * * \
/ * * \
-------- * * -------------
| PSTN | ** ** | SIP phone |
-------- ********************* -------------
Figure 3: PSTN origination - IP termination
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A call originates from the PSTN and terminates at a SIP phone. Note
that in Figure 5, the proxy server acts as the registrar for the SIP
phone in question.
A simple call-flow depicting the ISUP and SIP signaling for a PSTN-
originated call terminating at a SIP endpoint follows:
PSTN MGC Proxy SIP phone
|----IAM----->| | |
| |--------INVITE------>| |
| | |-------INVITE------->|
| |<------100 TRYING----| |
| | |<-------18x----------|
| |<---------18x--------| |
|<----ACM-----| | |
| | |<-------200 OK-------|
| |<-------200 OK-------| |
|<----ANM-----| | |
| |---------ACK-------->| |
| | |---------ACK-------->|
|=====================Conversation========================|
|-----REL---->| | |
| |----------BYE------->| |
|<----RLC-----| |---------BYE-------->|
| | |<-------200 OK-------|
| |<-------200 OK-------| |
| | | |
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RFC 3372 SIP-T September 2002
3.3 IP origination - PSTN termination
********************
*** ***
* *
* *
* *
* *
|-----| |----|
/|proxy| VoIP Network |MGC |\
/ ----- ---- \
/ * * \
/ * * \
/ * * \
------------ * * ---------
|SIP phone | ** ** | PSTN |
------------ ********************* ---------
Figure 4: IP origination - PSTN termination
A call originates from a SIP phone and terminates in the PSTN.
Unlike the previous two flows, there is therefore no ISUP
encapsulation in the request - the terminating gateway therefore only
performs translation on the SIP headers to derive values for ISUP
parameters.
A simple call-flow illustrating the different legs in the call is as
shown below.
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RFC 3372 SIP-T September 2002
SIP phone Proxy MGC PSTN
|-----INVITE----->| | |
| |--------INVITE-------->| |
|<---100 TRYING---| |-----IAM---->|
| |<------100 TRYING------| |
| | |<----ACM-----|
| |<---------18x----------| |
|<------18x-------| | |
| | |<----ANM-----|
| |<--------200 OK--------| |
|<-----200 OK-----| | |
|-------ACK------>| | |
| |----------ACK--------->| |
|========================Conversation===================|
|-------BYE------>| | |
| |----------BYE--------->| |
| | |-----REL---->|
| |<--------200 OK--------| |
|<-----200 OK-----| |<----RLC-----|
4. SIP-T Roles and Behavior
There are three distinct sorts of elements (from a functional point
of view) in a SIP VoIP network that interconnects with the PSTN:
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