📄 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),Vemuri & Peterson Best Current Practice [Page 6] 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 flowsVemuri & Peterson Best Current Practice [Page 7] 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.Vemuri & Peterson Best Current Practice [Page 8] RFC 3372 SIP-T September 2002 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 terminationVemuri & Peterson Best Current Practice [Page 9] RFC 3372 SIP-T September 2002 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-------| | | | | |Vemuri & Peterson Best Current Practice [Page 10] RFC 3372 SIP-T September 20023.3 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.Vemuri & Peterson Best Current Practice [Page 11] 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:⌨️ 快捷键说明
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