This document contains best current practice examples of Session Initiation Protocol (SIP) call flows showing interworking with the Public Switched Telephone Network (PSTN). Elements in these call flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways. Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP. PSTN telephony protocols are illustrated using ISDN (Integrated Services Digital Network), ISUP (ISDN User Part), and FGB (Feature Group B) circuit associated signaling. PSTN calls are illustrated using global telephone numbers from the PSTN and private extensions served on by a PBX (Private Branch Exchange). Call flow diagrams and message details are shown.
The call flows shown in this document were developed in the design of a SIP IP communications network. They represent an example of a minimum set of functionality.
It is the hope of the authors that this document will be useful for SIP implementers, designers, and protocol researchers alike and will help further the goal of a standard implementation of RFC 3261 . These flows represent carefully checked and working group reviewed scenarios of the most common SIP/PSTN interworking examples as a companion to the specifications.
These call flows are based on the current version 2.0 of SIP in RFC 3261  with SDP usage described in RFC 3264 . Other RFCs also comprise the SIP standard but are not used in this set of basic call flows. The SIP/ISUP mapping is based on RFC 3398 .
Various PSTN signaling protocols are illustrated in this document: ISDN (Integrated Services Digital Network), ISUP (ISDN User Part) and FGB (Feature Group B) circuit associated signaling. This document shows mainly ANSI ISUP due to its practical origins. However, as used in this document, the usage is virtually identical to the ITU-T International ISUP used as the reference in .
Basic SIP call flow examples are contained in a companion document, RFC 3665 .
A number of architecture, network, and protocol assumptions underlie the call flows in this document. Note that these assumptions are not requirements. They are outlined in this section so that they may be taken into consideration and to aid in the understanding of the call flow examples.
The authentication of SIP User Agents in these example call flows is performed using HTTP Digest as defined in  and .
Some Proxy Servers in these call flows insert Record-Route headers into requests to ensure that they are in the signaling path for future message exchanges.
These flows show TLS, TCP, and UDP for transport. SCTP could also be used. See the discussion in RFC 3261  for details on the transport issues for SIP.
The SIP Proxy Server has access to a Location Service and other databases. Information present in the Request-URI and the context (From header) is sufficient to determine to which proxy or gateway the message should be routed. In most cases, a primary and secondary route will be determined in case of a Proxy or Gateway failure downstream.
Gateways provide tones (ringing, busy, etc) and announcements to the PSTN side based on SIP response messages, or pass along audio in-band tones (ringing, busy tone, etc.) in an early media stream to the SIP side.
Telephone numbers are usually represented as SIP URIs. Note that an alternative is the use of the tel URI .
This document shows typical examples of SIP/ISUP interworking. Although in the spirit of the SIP-T framework , these examples do not represent a complete implementation of the framework. The examples here represent more of a minimal set of examples for very basic SIP to ISUP interworking, rather than the more complex goal of ISUP transparency. In particular, there are NO examples of encapsulated ISUP in this document. If present, these messages would show S/MIME encryption due to the sensitive nature of this information, as discussed in the SIP-T Framework security considerations section. (Note - RFC 3204  contains an example of an INVITE with encapsulated ISUP.) See the Security Considerations section for a more detailed discussion on the security of these call flows.
In ISUP, the Calling Party Number is abbreviated as CgPN and the Called Party Number is abbreviated as CdPN. Other abbreviations include Numbering Plan Indicator (NPI) and Nature of Address (NOA).
Dashed lines (---) represent signaling messages that are mandatory to the call scenario. These messages can be SIP or PSTN signaling. The arrow indicates the direction of message flow.
Messages with parentheses around their name represent optional messages.
This document does not prescribe the flows precisely as they are shown, but rather the flows illustrate the principles for best practice. They are best practices usages (orderings, syntax, selection of features for the purpose, handling of error) of SIP methods, headers and parameters. IMPORTANT: The exact flows here must not be copied as is by an implementer due to specific incorrect characteristics that were introduced into the document for convenience and are listed below. To sum up, the SIP/PSTN call flows represent well-reviewed examples of SIP usage, which are best common practice according to IETF consensus.
For simplicity in reading and editing the document, there are a number of differences between some of the examples and actual SIP messages. For example, the SIP Digest responses are not actual MD5 encodings. Call-IDs are often repeated, and CSeq counts often begin at 1. Header fields are usually shown in the same order. Usually only the minimum required header field set is shown, others that would normally be present, such as Accept, Supported, Allow, etc. are not shown.
Note that NGW 1 and NGW 2 also have device URIs (Contacts) of sip:ngw1@a.example.com and sip:ngw2@a.example.com which resolve to the Proxy Server sip:ss1.wcom.com using DNS SRV records.
In the following scenarios, Alice (sip:alice@a.example.com) is a SIP phone or other SIP-enabled device. Bob is reachable via the PSTN at global telephone number +19725552222. Alice places a call to Bob through a Proxy Server, Proxy 1, and a Network Gateway. In other scenarios, Alice places calls to Carol, who is served via a PBX (Private Branch Exchange) and is identified by a private extension 444-3333, or global number +1-918-555-3333. Note that Alice uses his/her global telephone number +1-314-555-1111 in the From header in the INVITE messages. This then gives the Gateway the option of using this header to populate the calling party identification field in subsequent signaling. Left open is the issue of how the Gateway can determine the accuracy of the telephone number which is necessary before passing it as a valid calling party number in the PSTN.
In these scenarios, Alice is a SIP phone or other SIP-enabled device. Alice places a call to Bob in the PSTN or Carol on a PBX through a Proxy Server and a Gateway.
In the failure scenarios, the call does not complete. In some cases however, a media stream is still setup. This is due to the fact that some failures in dialing to the PSTN result in in-band tones (busy, reorder tones or announcements - "The number you have dialed has changed. The new number is..."). The 183 Session Progress response containing SDP media information is used to setup this early media path so that the caller Alice knows the final disposition of the call.
The media stream is either terminated by the caller after the tone or announcement has been heard and understood, or by the Gateway after a timer expires.
In other failure scenarios, a SS7 Release with Cause Code is mapped to a SIP response. In these scenarios, the early media path is not used, but the actual failure code is conveyed to the caller by the SIP User Agent Client.
Alice dials the globalized E.164 number +19725552222 to reach Bob. Note that A might have only dialed the last 7 digits, or some other dialing plan. It is assumed that the SIP User Agent Client converts the digits into a global number and puts them into a SIP URI. Note that tel URIs could be used instead of SIP URIs.
that for this number to be passed into the SS7 network, it would have to be somehow verified for accuracy.
In this scenario, Bob answers the call, then Alice disconnects the call. Signaling between NGW 1 and Bob's telephone switch is ANSI ISUP. For the details of SIP to ISUP mapping, refer to .
In this flow, notice that the Contact returned by NGW 1 in messages F7-11 is sip:ngw1@a.example.com. This is because NGW 1 only accepts SIP messages that come through Proxy 1 - any direct signaling will be ignored. Since this Contact URI may be used outside of this dialog and must be routable (Section 8.1.1.8 in RFC 3261 ) the Contact URI for NGW 1 must resolve to Proxy 1. This Contact URI resolves via DNS to Proxy 1 (sip:ss1.a.example.com) which then resolves it to sip:ngw1.a.example.com which is the address of NGW 1.
This flow shows TCP transport.
Alice is a SIP device while Carol is connected via a Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk group. Alice dials Carol's telephone number (918-555-3333) which is globalized and put into a SIP URI.
The host portion of the Request-URI in the INVITE F3 is used to identify the context (customer, trunk group, or line) in which the private number 444-3333 is valid. Otherwise, this INVITE message could get forwarded by GW 1 and the context of the digits could become lost and the call unroutable.
Proxy 1 looks up the telephone number and locates the gateway that serves Carol. Carol is identified by its extension (444-3333) in the Request-URI sent to GW 1.
Note that the Contact URI for GW 1, as used in messages F8, F9, F12, and F13, is sips:4443333@gw1.a.example.com, which resolves directly to the gateway.
This flow shows the use of Secure SIP (sips) URIs.
Alice calls Bob through Proxy 1. Proxy 1 tries to route to a Network Gateway NGW 1. NGW 1 is not available and responds with a 503 Service Unavailable (F4). The call is then routed to Network Gateway NGW 2. Bob answers the call. The call is terminated when Alice disconnects the call. NGW 2 and Bob's telephone switch use ANSI ISUP signaling.
NGW 2 also only accepts SIP messages that come through Proxy 1, so the Contact URI sip:ngw2@a.example.com is used in this flow.
This flow shows UDP transport.
(sip:+19725552222@b.example.com). As a result, Alice's UA sends an INVITE and the call completes over IP bypassing the PSTN.
The call is terminated when Bob sends a BYE message.
IN NAPTR 100 10 "u" "sip+E2U"
Alice calls PSTN Bob through a Proxy Server Proxy 1 and a Network Gateway NGW 1. The call is rejected by the PSTN with a ANSI ISUP Release message REL containing a specific Cause code. This cause value (1) is mapped by the Gateway to a SIP 404 Address Incomplete response which is proxied back to Alice. For more details of ISUP cause value to SIP response mapping, refer to .
located.  Based upon location analysis the call is forwarded to NGW1.
Client for A prepares to receive data on port 49172 from the network.
Alice calls Bob in the PSTN through a proxy server Proxy 1 and Network Gateway NGW 1. The call is released by the Gateway after a timer expires due to no ANswer Message (ANM) being received. The Gateway sends an ISUP Release REL message to the PSTN and a 480 Temporarily Unavailable response to Alice in the SIP network.
In these scenarios, Alice is placing calls from the PSTN to Bob in a SIP network. Alice's telephone switch signals to a Network Gateway (NGW 1) using ANSI ISUP.
recordings are generated by NGW 1 and played in this speech path. When the call completes successfully, NGW 1 bridges the PSTN speech path with the IP media path.
To reduce the number of messages, only a single proxy server is shown in these flows, which means that the a.example.com proxy server has access to the b.example.com location service.
In this scenario, Alice from the PSTN calls Bob through a Network Gateway NGW1 and Proxy Server Proxy 1. When Bob answers the call, the media path is setup end-to-end. The call terminates when Alice hangs up the call, with Alice's telephone switch sending an ISUP RELease message that is mapped to a BYE by NGW 1.
This "fast answer" scenario is similar to 3.1., except that Bob immediately accepts the call, sending a 200 OK (F5) without sending a 180 Ringing response. The Gateway then sends an Answer Message (ANM) without sending an Address Complete Message (ACM). Note that for ETSI and some other ISUP variants, a CONnect message (CON) would be sent instead of the ANM.
In this scenario, Alice dials from PBX A to Bob through GW 1 and Proxy 1. This is an example of a call that appears destined for the PSTN but is instead routed to a SIP Client.
Signaling between PBX A and GW 1 is Feature Group B (FGB) circuit associated signaling, in-band Mult-Frequency (MF) outpulsing. After the receipt of the 180 Ringing from Bob, GW 1 generates a ringing tone for Alice.
Bob answers the call by sending a 200 OK. The call terminates when Alice hangs up, causing GW1 to send a BYE.
The Gateway can only identify the trunk group that the call came in on; it cannot identify the individual line on PBX A that is placing the call. The SIP URI used to identify the caller is shown in these flows as sip:551313@gw1.a.example.com.
Alice attempts to place a call through Gateway GW 1 and Proxy 1, which is unable to find any routing for the number. The call is rejected by Proxy 1 with a REL message containing a specific Cause value mapped by the gateway based on the SIP error.
In this scenario, Alice calls Bob through Network Gateway NGW 1 and Proxy 1. The call is routed to Bob by Proxy 1. The call is rejected by Bob who sends a 600 Busy Everywhere response. The Gateway sends a REL message containing a specific Cause value mapped by the gateway based on the SIP error.
Since no interworking is indicated in the IAM (F1), the busy tone is generated locally by Alice's telephone switch. In some scenarios, the busy signal is generated by the Gateway since interworking is indicated. For more discussion on interworking, refer to .
NGW 1 plays the busy tone since the IAM (F1) indicates the interworking is present. In scenario 5.2.2., with no interworking, the busy indication is carried in the REL Cause value and is generated locally instead.
Again, note that for ETSI or ITU ISUP, a CONnect message would be sent instead of the Answer Message.
Alice calls Bob through NGW 1 and Proxy 1. Proxy 1 re-sends the INVITE after the expiration of SIP timer T1 without receiving any response from Bob. Bob never responds with 180 Ringing or any other response (it is reachable but unresponsive). After the expiration of a timer, Alice's network disconnects the call by sending a Release message REL. The Gateway maps this to a CANCEL.
In this scenario, Alice calls Bob through NGW 1 and Proxy 1. Since Proxy 1 is stateless (it does not send a 100 Trying response), NGW 1 re-sends the INVITE message after the expiration of SIP timer T1. Bob does not respond with 180 Ringing. Alice's network disconnects the call with a release REL (CauseCode=102 Timeout).
CANCEL. Note that if Bob had sent a 200 OK response after the REL, NGW 1 would have sent an ACK and then a BYE to properly terminate the call.
In these scenarios, both the caller and the called party are in the telephone network, either normal PSTN subscribers or PBX extensions. The calls route through two Gateways and at least one SIP Proxy Server. The Proxy Server performs the authentication and location of the Gateways.
Again it is noted that the intent of this call flows document is not to provide a detailed parameter level mapping of SIP to PSTN protocols. For information on SIP to ISUP mapping, the reader is referred to other references .
In these scenarios, the call is successfully completed between the two Gateways, allowing the PSTN or PBX users to communicate. The 183 Session Progress response is used to indicate that in-band alerting may flow from the called party telephone switch to the caller.
is served by GW 2. The Proxy in F3 uses the host portion of the Request-URI to identify what private dialing plan is being referenced. The INVITE is then forwarded to GW 2 for call completion. An early media path is established end-to-end so that Alice can hear the ringing tone generated by PBX C.
Carol answers the call and the media path is cut through in both directions. Bob hangs up terminating the call.
PBX Alice calls PBX Carol via Gateway GW 1 and Proxy 1. During the attempt to reach Carol via GW 2, an error is encountered - Proxy 1 receives a 503 Service Unavailable (F4) response to the forwarded INVITE. This could be due to all circuits being busy, or some other outage at GW 2. Proxy 1 recognizes the error and uses an alternative route via GW 3 to terminate the call. From there, the call proceeds normally with Carol answering the call. The call is terminated when Carol hangs up.
This document provides examples of mapping from SIP to ISUP and ISUP to SIP. The gateways in these examples are compliant with the Security Considerations Section of RFC 3398  which is summarized here.
There are few security concerns relating to the mapping of ISUP to SIP besides privacy considerations in the calling party number passing. Some concerns relating to the mapping from tel URI parameters to ISUP include the user creation of parameters and codes relating to called number and local number portability (LNP). An operator of a gateway should use policies similar to those present in PSTN switches to avoid security problems.
The mapping from a SIP response code to an ISUP Cause Code presents a theoretical risk, so a gateway operator may implement policies controlling this mapping. Gateways should also not rely on the contents of the From header field for identity information, as it may be arbitrarily populated by a user. Instead, some sort of cryptographic authentication and authorization should be used for identity determination. These flows show both HTTP Digest for authentication of users, although for brevity, the challenge is not always shown.
The early media cut-through shown in some flows is another potential security risk, but it is also required for proper interaction with the PSTN. Again, a gateway operator should use proper policies relating to early media to prevent fraud and misuse. Finally, a user agent (even a properly authenticated one) can launch multiple simultaneous requests through a gateway, constituting a denial of service attack. The adoption of policies to limit the number of simultaneous requests from a single entity may be used to prevent this attack.
As discussed in the SIP-T framework , SIP/ISUP interworking can be employed as an interdomain signaling mechanism that may be subject to pre-existing trust relationships between administrative domains. Any administrative domain implementing SIP-T or SIP/ISUP interworking should have an adequate security apparatus (including elements that manage any appropriate policies to manage fraud and billing in an interdomain environment) in place to ensure that the translation of ISUP information does not result in any security violations.
Although no examples of this are shown in this document, transporting ISUP in SIP bodies may provide opportunities for abuse, fraud, and privacy concerns, especially when SIP-T requests can be generated, inspected or modified by arbitrary SIP endpoints. ISUP MIME bodies should be secured (preferably with S/MIME as detailed in RFC 3261 ) to alleviate this concern. Authentication properties provided by S/MIME would allow the recipient of a SIP-T message to ensure that the ISUP MIME body was generated by an authorized entity. Encryption would ensure that only carriers possessing a particular decryption key are capable of inspecting encapsulated ISUP MIME bodies in a SIP request.
 Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M. E. and Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002.
 Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with the Session Description Protocol (SDP)", RFC 3264, June 2002.
 Camarillo, G., Roach, A. B., Peterson, J. and L. Ong, "Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping", RFC 3398, December 2002.
 Vaha-Sipila, A., "URLs for Telephone Calls", RFC 2806, April 2000.
 Vemuri, A. and J. Peterson, "Session Initiation Protocol for Telephones (SIP-T): Context and Architectures", BCP 63, RFC 3372, September 2002.
 Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F., Watson, M. and M. Zonoun, "MIME media types for ISUP and QSIG Objects", RFC 3204, December 2001.
 Johnston, A., Donovan, S., Sparks, R., Cunningham, C. and K. Summers, "Session Initiation Protocol (SIP) Basic Call Flow Examples", RFC 3665, December 2003.
Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings, and Tom Taylor for their detailed comments during the final review. Thanks to Dean Willis for his early contributions to the development of this document. Thanks to Jon Peterson for his help on the security section.
The authors wish to thank Kundan Singh for performing parser validation of messages.
The authors wish to thank the following individuals for their participation in a detailed review of this call flows document: Aseem Agarwal, Rafi Assadi, Ben Campbell, Sunitha Kumar, Jon Peterson, Marc Petit-Huguenin, Vidhi Rastogi, and Bodgey Yin Shaohua.
The authors also wish to thank the following individuals for their assistance: Jean-Francois Mule, Hemant Agrawal, Henry Sinnreich, David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole MCI WorldCom IPOP Design team, Scott Orton, Greg Osterhout, Pat Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise Ingram, Denise Caballero, Tom Redman, Ilya Slain, Pat Sollee, John Truetken, and others from MCI WorldCom, 3Com, Cisco, Lucent and Nortel.
All listed authors actively contributed large amounts of text to this document.