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Versions: 00 01 02 RFC 3666

   SIPPING Working Group                                    A. Johnston
   Internet Draft                                              WorldCom
   Document: draft-ietf-sipping-pstn-call-flows-02.txt       S. Donovan
   Expires: October 2003                                      R. Sparks
                                                          C. Cunningham
                                                            dynamicsoft
                                                             K. Summers
                                                                  Sonus
                                                             April 2003


                Session Initiation Protocol PSTN Call Flows


Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
        http://www.ietf.org/ietf/1id-abstracts.txt
   The list of Internet-Draft Shadow Directories can be accessed at
        http://www.ietf.org/shadow.html.


Abstract

   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.




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Conventions used in this document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC-2119 [1].

Table of Contents

   1. Overview.......................................................2
      1.1 General Assumptions........................................3
      1.2 Legend for Message Flows...................................4
      1.3 SIP Protocol Assumptions...................................5
   2. SIP to PSTN Dialing............................................6
      2.1 Successful SIP to ISUP PSTN call...........................7
      2.2 Successful SIP to ISDN PBX call...........................15
      2.3 Successful SIP to ISUP PSTN call with overflow............23
      2.4 Session established using ENUM Query......................32
      2.5 Unsuccessful SIP to PSTN call: Treatment from PSTN........38
      2.6 Unsuccessful SIP to PSTN: REL w/Cause from PSTN...........45
      2.7 Unsuccessful SIP to PSTN: ANM Timeout.....................50
   3. PSTN to SIP Dialing...........................................56
      3.1 Successful PSTN to SIP call...............................57
      3.2 Successful PSTN to SIP call, Fast Answer..................64
      3.3 Successful PBX to SIP call................................70
      3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL.....77
      3.5 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL......79
      3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones...83
      3.7 Unsuccessful PSTN->SIP, ACM timeout.......................87
      3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy......91
      3.9 Unsuccessful PSTN->SIP, Caller Abandonment................95
   4. PSTN to PSTN Dialing via SIP Network.........................101
      4.1 Successful ISUP PSTN to ISUP PSTN call...................102
      4.2 Successful FGB PBX to ISDN PBX call with overflow........110
   Security Considerations.........................................118
   Normative References............................................120
   Informative References..........................................120
   Acknowledgments.................................................121
   Author's Addresses..............................................121

1.   Overview

   The call flows shown in this document were developed in the design of
   a SIP IP communications network.  They represent an example
   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 [2].
   These flows represent carefully checked and working group reviewed


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                         SIP PSTN Call Flows               April 2003


   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 [2] with SDP usage described in RFC 3264 [3]. 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 zzzz [4].

   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 [4].

   Basic SIP call flow examples are contained in a companion document,
   RFC yyyy [11].

1.1     General Assumptions

   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 SIP Digest as defined in [3] and [5].

   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 [6] could
   also be used.  See the discussion in RFC 3261 [2] 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 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.



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   The interactions between the Proxy and Gateway can be summarized as
   follows:

     . The SIP Proxy Server performs digit analysis and lookup and
       locates the correct gateway.

     . The SIP Proxy Server performs gateway location based on primary
       and secondary routing.

   Telephone numbers are usually represented as SIP URIs.  Note that an
   alternative is the use of the tel URI [7].

   This document shows typical examples of SIP/ISUP interworking.
   Although in the spirit of the SIP-T framework [8], 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 [9] 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).

1.2     Legend for Message Flows

   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.

   Double dashed lines (===) represent media paths between network
   elements.

   Messages with parentheses around their name represent optional
   messages.

   Messages are identified in the Figures as F1, F2, etc.  This
   references the message details in the list that follows the Figure.
   Comments in the message details are shown in the following form:

    /* Comments. */




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1.3     SIP Protocol Assumptions

   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.

   Actors:

   Element       Display Name   URI                        IP Address
   -------       ------------   ---                        ----------

   User Agent    Alice          sip:alice@a.example.com    192.0.2.101
   User Agent    Bob            sip:bob@b.example.com      192.0.2.200
   Proxy Server                 sip:ss1.a.example.com      192.0.2.111
   User Agent (Gateway)         sip:gw1.a.example.com      192.0.2.201
   User Agent (Gateway)         sip:gw2.a.example.com      192.0.2.202
   User Agent (Gateway)         sip:gw3.a.example.com      192.0.2.203
   User Agent (Gateway)         sip:ngw1.a.example.com     192.0.2.103
   User Agent (Gateway)         sip:ngw2.a.example.com     192.0.2.102

   Note that NGW 1 and NGW 2 also have a device URIs (Contacts) of
   sip:ngw1@a.example.com and sip:ngw2@a.example.com which resolves to
   the Proxy Server sip:ss1.wcom.com using DNS SRV records.












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2.   SIP to PSTN Dialing


   In the following scenarios, Alice (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 User
   A 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, 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.












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2.1    Successful SIP to ISUP PSTN call

   Alice           Proxy 1           NGW 1          Switch B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     100  F2    |                |                |
     |<---------------|   INVITE F3    |                |
     |                |--------------->|                |
     |                |     100  F4    |                |
     |                |<---------------|     IAM F5     |
     |                |                |--------------->|
     |                |                |     ACM F6     |
     |                |     183 F7     |<---------------|
     |     183 F8     |<---------------|                |
     |<---------------|                |                |
     |        Both Way RTP Media       |  One Way Voice |
     |<===============================>|<===============|
     |                |                |      ANM F9    |
     |                |    200 F10     |<---------------|
     |     200 F11    |<---------------|                |
     |<---------------|                |                |
     |     ACK F12    |                |                |
     |--------------->|     ACK F13    |                |
     |                |--------------->|                |
     |        Both Way RTP Media       | Both Way Voice |
     |<===============================>|<==============>|
     |     BYE F14    |                |                |
     |--------------->|     BYE F15    |                |
     |                |--------------->|                |
     |                |     200 F16    |                |
     |     200 F17    |<---------------|     REL F18    |
     |<---------------|                |--------------->|
     |                |                |     RLC F19    |
     |                |                |<---------------|
     |                |                |                |



   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.

   Alice could use either their SIP address (sip:alice@a.example.com) or
   SIP telephone number (sip:+13145551111@ss1.a.example.com;user=phone)
   in the From header.  In this example, the telephone number is
   included, and it is shown as being passed as calling party


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   identification through the Network Gateway (NGW 1) to Bob (F5).  Note
   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 [4].

   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 [2]) the Contact
   URI for NGW 1 must resolve to Proxy 1.  This Contact URI is an AOR
   which 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.


   Message Details


   F1 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Proxy-Authorization: Digest username="alice", realm="a.example.com",
    nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
    uri="sip:+19725552222@ss1.a.example.com;user=phone",
    response="ccdca50cb091d587421457305d097458c"
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000



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   F2 100 Trying Proxy 1 -> Alice

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* Proxy 1 uses a Location Service function to determine the gateway
   for terminating this call.  The call is forwarded to NGW 1.  Client
   for A prepares to receive data on port 49172 from the
   network.*/

   F3 INVITE Proxy 1 -> NGW 1

   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying NGW 1 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1


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                         SIP PSTN Call Flows               April 2003


    ;received=192.0.2.111
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 IAM NGW 1 -> Bob

   IAM
   CdPN=972-555-2222,NPI=E.164,NOA=National
   CgPN=314-555-1111,NPI=E.164,NOA=National


   F6 ACM Bob -> NGW 1

   ACM


   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* NGW 1 sends PSTN audio (ringing) in the RTP path to A */


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   F8 183 Session Progress Proxy 1 -> Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 ANM Bob -> NGW 1

   ANM


   F10 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146



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                         SIP PSTN Call Flows               April 2003


   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 200 OK Proxy 1 -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F12 ACK Alice -> Proxy 1

   ACK sip:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0




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                         SIP PSTN Call Flows               April 2003


   F13 ACK Proxy 1 -> NGW 1

   ACK sip:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   /* Alice Hangs Up with Bob. */

   F14 BYE Alice -> Proxy 1

   BYE sip:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F15 BYE Proxy 1 -> NGW 1

   BYE sip:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0




Johnston et al          Expires - October 2002               [Page 13]


                         SIP PSTN Call Flows               April 2003


   F16 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F17 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F18 REL NGW 1 -> B

   REL
   CauseCode=16 Normal


   F19 RLC B -> NGW 1

   RLC












Johnston et al          Expires - October 2002               [Page 14]


                         SIP PSTN Call Flows               April 2003


2.2    Successful SIP to ISDN PBX call

   Alice            Proxy 1           GW 1             PBX C
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     100  F2    |                |                |
     |<---------------|   INVITE F3    |                |
     |                |--------------->|                |
     |                |     100  F4    |                |
     |                |<---------------|    SETUP F5    |
     |                |                |--------------->|
     |                |                |  CALL PROC F6  |
     |                |                |<---------------|
     |                |                |   PROGress F7  |
     |                |    180 F8      |<---------------|
     |    180 F9      |<---------------|                |
     |<---------------|                |                |
     |                |                |  One Way Voice |
     |                |                |<===============|
     |                |                |   CONNect F10  |
     |                |                |<---------------|
     |                |                | CONNect ACK F11|
     |                |    200 F12     |--------------->|
     |     200 F13    |<---------------|                |
     |<---------------|                |                |
     |     ACK F14    |                |                |
     |--------------->|     ACK F15    |                |
     |                |--------------->|                |
     |        Both Way RTP Media       | Both Way Voice |
     |<===============================>|<==============>|
     |     BYE F16    |                |                |
     |--------------->|     BYE F17    |                |
     |                |--------------->|                |
     |                |     200 F18    |                |
     |     200 F19    |<---------------| DISConnect F20 |
     |<---------------|                |--------------->|
     |                |                |   RELease F21  |
     |                |                |<---------------|
     |                |                | RELease COM F22|
     |                |                |--------------->|
     |                |                |                |

   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


Johnston et al          Expires - October 2002               [Page 15]


                         SIP PSTN Call Flows               April 2003


   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.  Carolis identified by its extension
   (444-3333) in the Request-URI sent to GW 1.

   Note that the Contact URI for GW1 as used in messages F8, F9, F12,
   and F13 is sips:4443333@gw1.a.example.com which does resolve directly
   to the gateway.

   This flow shows the use of Secure SIP (sips) URIs.


   Message Details


   F1 INVITE Alice -> Proxy 1

   INVITE sips:+19185553333@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sips:alice@client.a.example.com>
   Proxy-Authorization: Digest username="alice",
    realm="a.example.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",
    opaque="", uri="sips:+19185553333@ss1.a.example.com;user=phone",
    response="6c792f5c9fa360358b93c7fb826bf550"
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 100 Trying Proxy 1 -> Alice

   SIP/2.0 100 Trying


Johnston et al          Expires - October 2002               [Page 16]


                         SIP PSTN Call Flows               April 2003


   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Content-Length: 0


   F3 INVITE Proxy 1 -> GW 1

   INVITE sips:4443333@gw1.a.example.com SIP/2.0
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sips:ss1.a.example.com;lr>
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sips:alice@client.a.example.com>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying GW -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Content-Length: 0




Johnston et al          Expires - October 2002               [Page 17]


                         SIP PSTN Call Flows               April 2003


   F5 SETUP GW 1 -> Carol

   Protocol discriminator=Q.931
   Message type=SETUP
   Bearer capability: Information transfer capability=0 (Speech) or 16
   (3.1 kHz audio)
   Channel identification=Preferred or exclusive B-channel
   Progress indicator=1 (Call is not end-to-end ISDN;further call
   progress information may be available inband)
   Called party number:
   Type of number unknown
   Digits=444-3333


   F6 CALL PROCeeding Carol-> GW 1

   Protocol discriminator=Q.931
   Message type=CALL PROC
   Channel identification=Exclusive B-channel


   F7 PROGress Carol-> GW 1

   Protocol discriminator=Q.931
   Message type=PROG
   Progress indicator=1 (Call is not end-to-end ISDN;further call
   progress information may be available inband)


   F8 180 Ringing GW 1 -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sips:ss1.a.example.com;lr>
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sips:4443333@gw1.a.example.com>
   Content-Length: 0


   F9 180 Ringing Proxy 1 -> Alice



Johnston et al          Expires - October 2002               [Page 18]


                         SIP PSTN Call Flows               April 2003


   SIP/2.0 180 Ringing
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sips:ss1.a.example.com;lr>
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sips:4443333@gw1.a.example.com>
   Content-Length: 0


   F10 CONNect Carol-> GW 1

   Protocol discriminator=Q.931
   Message type=CONN


   F11 CONNect ACK GW 1 -> Carol

   Protocol discriminator=Q.931
   Message type=CONN ACK


   F12 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sips:ss1.a.example.com;lr>
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sips:4443333@gw1.a.example.com>
   Content-Type: application/sdp
   Content-Length: 144

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0


Johnston et al          Expires - October 2002               [Page 19]


                         SIP PSTN Call Flows               April 2003


   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F13 200 OK Proxy 1 -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sips:ss1.a.example.com;lr>
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sips:4443333@gw1.a.example.com>
   Content-Type: application/sdp
   Content-Length: 144

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 ACK Alice -> Proxy 1

   ACK sips:4443333@gw1.a.example.com SIP/2.0
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sips:ss1.a.example.com;lr>
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 ACK
   Content-Length: 0


   F15 ACK Proxy 1 -> GW 1

   ACK sips:4443333@gw1.a.example.com SIP/2.0
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9


Johnston et al          Expires - October 2002               [Page 20]


                         SIP PSTN Call Flows               April 2003


    ;received=192.0.2.101
   Max-Forwards: 69
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 ACK
   Content-Length: 0


   /* Alice Hangs Up with Bob. */

   F16 BYE Alice -> Proxy 1

   BYE sips:4443333@gw1.a.example.com SIP/2.0
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sips:ss1.a.example.com;lr>
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 3 BYE
   Content-Length: 0


   F17 BYE Proxy 1 -> GW 1

   BYE sips:4443333@gw1.a.example.com SIP/2.0
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 3 BYE
   Content-Length: 0


   F18 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111


Johnston et al          Expires - October 2002               [Page 21]


                         SIP PSTN Call Flows               April 2003


   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 3 BYE
   Content-Length: 0


   F19 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 3 BYE
   Content-Length: 0


   F20 DISConnect GW 1 -> Carol

   Protocol discriminator=Q.931
   Message type=DISC
   Cause=16 (Normal clearing)


   F21 RELease Carol-> GW 1

   Protocol discriminator=Q.931
   Message type=REL


   F22 RELease COMplete GW 1 -> Carol

   Protocol discriminator=Q.931
   Message type=REL COM









Johnston et al          Expires - October 2002               [Page 22]


                         SIP PSTN Call Flows               April 2003


2.3    Successful SIP to ISUP PSTN call with overflow

   Alice          Proxy 1         NGW 1          NGW 2        Switch B
    |              |              |              |              |
    |  INVITE F1   |              |              |              |
    |------------->|              |              |              |
    |              |  INVITE F2   |              |              |
    |    100  F3   |------------->|              |              |
    |<-------------|    503 F4    |              |              |
    |              |<-------------|              |              |
    |              |    ACK F5    |              |              |
    |              |------------->|              |              |
    |              |   INVITE F6                 |              |
    |              |---------------------------->|     IAM F7   |
    |              |                             |------------->|
    |              |                             |     ACM F8   |
    |              |            183 F9           |<-------------|
    |   183 F10    |<----------------------------|              |
    |<-------------|                             |              |
    |               Two Way RTP Media            | One Way Voice|
    |<==========================================>|<=============|
    |              |                             |    ANM F11   |
    |              |           200 F12           |<-------------|
    |    200 F13   |<----------------------------|              |
    |<-------------|                             |              |
    |    ACK F14   |                             |              |
    |------------->|            ACK F15          |              |
    |              |---------------------------->|              |
    |             Both Way RTP Media             |Both Way Voice|
    |<==========================================>|<============>|
    |    BYE F16   |                             |              |
    |------------->|           BYE F17           |              |
    |              |---------------------------->|              |
    |              |           200 F18           |              |
    |    200 F19   |<----------------------------|    REL F20   |
    |<-------------|                             |------------->|
    |              |                             |    RLC F21   |
    |              |                             |<-------------|
    |              |                             |              |

   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.


Johnston et al          Expires - October 2002               [Page 23]


                         SIP PSTN Call Flows               April 2003



   This flow shows UDP transport.


   Message Details


   F1 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com>
   Proxy-Authorization: Digest username="alice",
    realm="a.example.com", nonce="b59311c3ba05b401cf80b2a2c5ac51b0",
    opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
    response="ba6ab44923fa2614b28e3e3957789ab0"
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine where B is
   located.  Proxy 1 receives a primary route NGW 1 and a secondary
   route NGW 2.  NGW 1 is tried first */

   F2 INVITE Proxy 1 -> NGW 1

   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>


Johnston et al          Expires - October 2002               [Page 24]


                         SIP PSTN Call Flows               April 2003


   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 100 Trying Proxy 1 -> Alice

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F4 503 Service Unavailable NGW 1 -> Proxy 1

   SIP/2.0 503 Service Unavailable
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 ACK Proxy 1 -> NGW 1



Johnston et al          Expires - October 2002               [Page 25]


                         SIP PSTN Call Flows               April 2003


   ACK sip:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com>;user=phone>
    ;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   /* Proxy 1 now tries secondary route to NGW 2 */

   F6 INVITE Proxy 1 -> NGW 2

   INVITE sip:+19725552222@ngw2.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 IAM NGW 2 -> Bob

   IAM
   CdPN=972-555-2222,NPI=E.164,NOA=National
   CgPN=314-555-1111,NPI=E.164,NOA=National


   F8 ACM Bob -> NGW 2



Johnston et al          Expires - October 2002               [Page 26]


                         SIP PSTN Call Flows               April 2003


   ACM


   F9 183 Session Progress NGW 2 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw2@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com
   s=-
   c=IN IP4 ngw2.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* RTP packets are sent by GW to A for audio (e.g. ring tone) */

   F10 183 Session Progress Proxy 1 -> Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw2@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0


Johnston et al          Expires - October 2002               [Page 27]


                         SIP PSTN Call Flows               April 2003


   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com
   s=-
   c=IN IP4 ngw2.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 ANM Bob -> NGW 2

   ANM


   F12 200 OK NGW 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw2@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com
   s=-
   c=IN IP4 ngw2.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F13 200 OK Proxy 1 -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>


Johnston et al          Expires - October 2002               [Page 28]


                         SIP PSTN Call Flows               April 2003


    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw2@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com
   s=-
   c=IN IP4 ngw2.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 ACK Alice -> Proxy 1

   ACK sip:ngw2@a.example.com SIP/2.0
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F15 ACK Proxy 1 -> NGW 2

   ACK sip:ngw2@a.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between A and B(via the GW) */



Johnston et al          Expires - October 2002               [Page 29]


                         SIP PSTN Call Flows               April 2003


   /* Alice Hangs Up with Bob. */

   F16 BYE Alice -> Proxy 1

   BYE sip:ngw2@a.example.com SIP/2.0
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F17 BYE Proxy 1 -> NGW 2

   BYE sip:ngw2@a.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F18 200 OK NGW 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0




Johnston et al          Expires - October 2002               [Page 30]


                         SIP PSTN Call Flows               April 2003


   F19 200 OK Proxy 1 -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F20 REL NGW 2 -> B

   REL
   CauseCode=16 Normal

   F21 RLC B -> NGW 2

   RLC





























Johnston et al          Expires - October 2002               [Page 31]


                         SIP PSTN Call Flows               April 2003


2.4    Successful SIP to SIP using ENUM Query

   Alice         DNS Server         Proxy 3            Bob
     |                |                |                |
     |  ENUM Query F1 |                |                |
     |--------------->|                |                |
     |   Response F2  |                |                |
     |<---------------|                |                |
     |            INVITE F3            |                |
     |-------------------------------->|    INVITE F4   |
     |             100 F5              |--------------->|
     |<--------------------------------|      180 F6    |
     |             180 F7              |<---------------|
     |<--------------------------------|                |
     |                                 |     200 F8     |
     |             200 F9              |<---------------|
     |<--------------------------------|                |
     |             ACK F10             |                |
     |-------------------------------->|     ACK F11    |
     |                                 |--------------->|
     |                Both Way RTP Media                |
     |<================================================>|
     |                                 |     BYE F12    |
     |             BYE F13             |<---------------|
     |<--------------------------------|                |
     |             200 F14             |                |
     |-------------------------------->|     200 F15    |
     |                                 |--------------->|
     |                                 |                |

   In this scenario, Alice places a call to Bob by dialing Bob's
   telephone number (9725552222).  Alice's UA converts the phone number
   to an E.164 number (+19725552222) performs an ENUM query [10] on the
   E.164 number (2.2.2.2.5.5.5.2.7.9.1.e164.arpa) which returns a NAPTR
   record containing a SIP AOR URI for Bob
   (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.


   Message Details


   F1 ENUM Query Alice -> DNS Server

   2.2.2.2.5.5.5.2.7.9.1.e164.arpa




Johnston et al          Expires - October 2002               [Page 32]


                         SIP PSTN Call Flows               April 2003


   F2 ENUM NAPTR Set DNS Server -> Alice

   $ORIGIN 2.2.2.2.5.5.5.2.7.9.1.e164.arpa.
         IN NAPTR 100 10 "u" "sip+E2U"
                "!^.*$!sip:+19725552222@b.example.com!".


   F3 INVITE Alice -> Proxy 3

   INVITE sip:+19725552222@b.example.com SIP/2.0
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sip:+13145551111@client.a.example.com>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 INVITE Proxy 3 -> Bob

   INVITE sip:+19725552222@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss3.b.example.com;lr>
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sip:+13145551111@client.a.example.com>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com


Johnston et al          Expires - October 2002               [Page 33]


                         SIP PSTN Call Flows               April 2003


   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 100 Trying Proxy 3 -> Alice

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Content-Length: 0


   F6 180 Ringing B -> Proxy 3

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
    ;received=192.0.2.233
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss3.b.example.com;lr>
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sip:+19725552222@client.b.example.com>
   Content-Length: 0


   F7 180 Ringing Proxy 3 -> Alice

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss3.b.example.com;lr>
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sip:+19725552222@client.b.example.com>
   Content-Length: 0


   F8 200 OK Bob -> Proxy 3



Johnston et al          Expires - October 2002               [Page 34]


                         SIP PSTN Call Flows               April 2003


   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
    ;received=192.0.2.233
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss3.b.example.com;lr>
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sip:+19725552222@client.b.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 151

   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 client.b.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 200 OK Proxy -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss3.b.example.com;lr>
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 INVITE
   Contact: <sip:+19725552222@client.b.example.com>
   Content-Type: application/sdp
   Content-Length: 151

   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 192.0.2.100
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F10 ACK Alice -> Proxy 3

   ACK sip:+19725552222@client.b.example.com SIP/2.0


Johnston et al          Expires - October 2002               [Page 35]


                         SIP PSTN Call Flows               April 2003


   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
   Max-Forwards: 70
   Route: <sip:ss3.b.example.com;lr>
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 ACK
   Content-Length: 0


   F11 ACK Proxy 3 -> Bob

   ACK sip:+19725552222@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
    ;received=192.0.2.101
   Max-Forwards: 69
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   To: <sip:+19725552222@b.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 2 ACK
   Content-Type: application/sdp
   Content-Length: 0


   /* RTP streams are established between A and B*/

   /* User B Hangs Up with User A. */

   F12 BYE Bob -> Proxy 3

   BYE sip:+13145551111@client.a.example.com SIP/2.0
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
   Max-Forwards: 70
   Route: <sip:ss3.b.example.com;lr>
   From: <sip:+19725552222@b.example.com>;tag=314159
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 BYE
   Content-Length: 0


   F13 BYE Proxy 3 -> Alice

   BYE sip:+13145551111@client.a.example.com SIP/2.0
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
    ;received=192.0.2.100
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
   Max-Forwards: 69


Johnston et al          Expires - October 2002               [Page 36]


                         SIP PSTN Call Flows               April 2003


   From: <sip:+19725552222@b.example.com>;tag=314159
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 BYE
   Content-Length: 0


   F14 200 OK Alice -> Proxy 3

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
    ;received=192.0.2.233
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
    ;received=192.0.2.100
   From: <sip:+19725552222@b.example.com>;tag=314159
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 BYE
   Content-Length: 0


   F15 200 OK Proxy 3 -> Bob

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
    ;received=192.0.2.100
   From: <sip:+19725552222@b.example.com>;tag=314159
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 BYE
   Content-Length: 0




















Johnston et al          Expires - October 2002               [Page 37]


                         SIP PSTN Call Flows               April 2003


2.5    Unsuccessful SIP to PSTN call: Treatment from PSTN

   Alice            Proxy 1           NGW 1            Bob
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     100  F2    |                |                |
     |<---------------|   INVITE F3    |                |
     |                |--------------->|                |
     |                |     100  F4    |                |
     |                |<---------------|     IAM F5     |
     |                |                |--------------->|
     |                |                |     ACM F6     |
     |                |     183 F7     |<---------------|
     |     183 F8     |<---------------|                |
     |<---------------|                |                |
     |         Two Way RTP Media       |  One Way Voice |
     |<===============================>|<===============|
     |                 Treatment Applied                |
     |<=================================================|
     |   CANCEL F9    |                |                |
     |--------------->|                |                |
     |     200 F10    |                |                |
     |<---------------|   CANCEL F11   |                |
     |                |--------------->|                |
     |                |     200 F12    |                |
     |                |<---------------|     REL F13    |
     |                |                |--------------->|
     |                |                |     RLC F14    |
     |                |     487 F15    |<---------------|
     |                |<---------------|                |
     |                |     ACK F16    |                |
     |     487 F17    |--------------->|                |
     |<---------------|                |                |
     |     ACK F18    |                |                |
     |--------------->|                |                |
     |                |                |                |

   Alice calls Bob in the PSTN through a proxy server Proxy 1 and a
   Network Gateway NGW 1.  The call is rejected by the PSTN with an in-
   band treatment (tone or recording) played.  Alice hears the
   treatment and then hangs up, which results in a CANCEL (F9) being
   sent to terminate the call. (A BYE is not sent since no final
   response was ever received by Alice.)


   Message Details




Johnston et al          Expires - October 2002               [Page 38]


                         SIP PSTN Call Flows               April 2003


   F1 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Proxy-Authorization: Digest username="alice",
    realm="a.example.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40",
    opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
    response="e178fbe430e6680a1690261af8831f40"
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 100 Trying Proxy 1 -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* Proxy 1 uses a Location Service function to determine where B is
   located.  Based upon location analysis the call is forwarded to NGW
   1.  Client for A prepares to receive data on port 49172 from the
   network. */

   F3 INVITE Proxy 1 -> NGW 1

   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1


Johnston et al          Expires - October 2002               [Page 39]


                         SIP PSTN Call Flows               April 2003


   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying NGW 1 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 IAM NGW 1 -> Bob

   IAM
   CdPN=972-555-2222,NPI=E.164,NOA=National
   CgPN=314-555-1111,NPI=E.164,NOA=National


   F6 ACM Bob -> NGW 1

   ACM




Johnston et al          Expires - October 2002               [Page 40]


                         SIP PSTN Call Flows               April 2003


   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 183 Session Progress Proxy 1 -> Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0


Johnston et al          Expires - October 2002               [Page 41]


                         SIP PSTN Call Flows               April 2003


   a=rtpmap:0 PCMU/8000


   /* Caller hears the recorded announcement, then hangs up */

   F9 CANCEL Alice -> Proxy 1

   CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F10 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F11 CANCEL Proxy 1 -> NGW 1

   CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F12 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111


Johnston et al          Expires - October 2002               [Page 42]


                         SIP PSTN Call Flows               April 2003


   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F13 REL NGW 1 -> B

   REL
   CauseCode=18 No user responding


   F14 RLC B -> NGW 1

   RLC


   F15 487 Request Terminated NGW 1 -> Proxy 1

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F16 ACK Proxy 1 -> NGW 1

   ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0




Johnston et al          Expires - October 2002               [Page 43]


                         SIP PSTN Call Flows               April 2003


   F17 487 Request Terminated Proxy 1 -> A

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F18 ACK Alice -> Proxy 1

   ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0

























Johnston et al          Expires - October 2002               [Page 44]


                         SIP PSTN Call Flows               April 2003


2.6    Unsuccessful SIP to PSTN: REL w/Cause from PSTN

   Alice            Proxy 1           NGW 1           Switch B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     100  F2    |                |                |
     |<---------------|   INVITE F3    |                |
     |                |--------------->|                |
     |                |     100  F4    |                |
     |                |<---------------|     IAM F5     |
     |                |                |--------------->|
     |                |                |    REL(1) F6   |
     |                |                |<---------------|
     |                |                |     RLC F7     |
     |                |     404 F8     |--------------->|
     |                |<---------------|                |
     |                |     ACK F9     |                |
     |                |--------------->|                |
     |     404 F10    |                |                |
     |<---------------|                |                |
     |     ACK F11    |                |                |
     |--------------->|                |                |
     |                |                |                |

   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 [4].


   Message Details


   F1 INVITE Alice -> Proxy 1

   INVITE sip:+44-1234@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Proxy-Authorization: Digest username="alice",
    realm="a.example.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",


Johnston et al          Expires - October 2002               [Page 45]


                         SIP PSTN Call Flows               April 2003


    opaque="", uri="sip:+44-1234@ss1.a.example.com;user=phone",
    response="a451358d46b55512863efe1dccaa2f42"
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 100 Trying Proxy 1 -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* Proxy 1 uses a Location Service function to determine where B is
   located.  Based upon location analysis the call is forwarded to NGW1.
   Client for A prepares to receive data on port 49172 from the network.
   */

   F3 INVITE Proxy 1 -> NGW 1

   INVITE sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 154



Johnston et al          Expires - October 2002               [Page 46]


                         SIP PSTN Call Flows               April 2003


   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying NGW 1 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 IAM NGW 1 -> Bob

   IAM
   CdPN=44-1234,NPI=E.164,NOA=International
   CgPN=314-555-1111,NPI=E.164,NOA=National


   F6 REL Bob -> NGW 1

   REL
   CauseValue=1 Unallocated number


   F7 RLC NGW 1 -> Bob

   RLC


   /* Network Gateway maps CauseValue=1 to the SIP message 404 Not
      Found */

   F8 404 Not Found NGW 1 -> Proxy 1

   SIP/2.0 404 Not Found
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1


Johnston et al          Expires - October 2002               [Page 47]


                         SIP PSTN Call Flows               April 2003


    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Error-Info: <sip:not-found-ann@ann.a.example.com>
   Content-Length: 0


   F9 ACK Proxy 1 -> NGW 1

   ACK sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0
   Max-Forwards: 70
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F10 404 Not Found Proxy 1 -> Alice

   SIP/2.0 404 Not Found
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Error-Info: <sip:not-found-ann@ann.a.example.com>
   Content-Length: 0


   F11 ACK Alice -> Proxy 1

   ACK sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK


Johnston et al          Expires - October 2002               [Page 48]


                         SIP PSTN Call Flows               April 2003


   Content-Length: 0


















































Johnston et al          Expires - October 2002               [Page 49]


                         SIP PSTN Call Flows               April 2003


2.7    Unsuccessful SIP to PSTN: ANM Timeout

   Alice           Proxy 1           NGW 1           Switch B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     100  F2    |                |                |
     |<---------------|   INVITE F3    |                |
     |                |--------------->|                |
     |                |     100  F4    |                |
     |                |<---------------|     IAM F5     |
     |                |                |--------------->|
     |                |                |     ACM F6     |
     |                |      183 F7    |<---------------|
     |     183 F8     |<---------------|                |
     |<---------------|                |                |
     |                |      Timer on NGW 1 Expires     |
     |                |                |                |
     |                |                |     REL F9     |
     |                |                |--------------->|
     |                |                |    RLC F10     |
     |                |     480 F11    |<---------------|
     |                |<---------------|                |
     |                |     ACK F12    |                |
     |                |--------------->|                |
     |     480 F13    |                |                |
     |<---------------|                |                |
     |     ACK F14    |                |                |
     |--------------->|                |                |

   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.


   Message Details


   F1 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com


Johnston et al          Expires - October 2002               [Page 50]


                         SIP PSTN Call Flows               April 2003


   CSeq: 1 INVITE
   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Proxy-Authorization: Digest username="alice",
    realm="a.example.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40",
    opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
    response="579cb9db184cdc25bf816f37cbc03c7d"
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine where B is
   located.  Based upon location analysis the call is forwarded to NGW
   1.  Client for A prepares to receive data on port 49172 from the
   network.*/

   F2 100 Trying Proxy 1 -> A

   SIP/2.0  100 Trying
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F3 INVITE Proxy 1 -> NGW 1

   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE


Johnston et al          Expires - October 2002               [Page 51]


                         SIP PSTN Call Flows               April 2003


   Contact: <sip:alice@client.a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 154

   v=0
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com
   s=-
   c=IN IP4 client.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying NGW 1 -> Proxy 1

   SIP/2.0  100 Trying
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 IAM NGW 1 -> Bob

   IAM
   CdPN=972-555-2222,NPI=E.164,NOA=National
   CgPN=314-555-1111,NPI=E.164,NOA=National


   F6 ACM Bob -> NGW 1

   ACM


   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>


Johnston et al          Expires - October 2002               [Page 52]


                         SIP PSTN Call Flows               April 2003


    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 183 Session Progress Proxy 1 -> Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.a.example.com;lr>
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* After NGW 1's timer expires, Network Gateway sends REL to ISUP
   network and 480 to SIP network */

   F9 REL NGW 1 -> Bob

   REL


Johnston et al          Expires - October 2002               [Page 53]


                         SIP PSTN Call Flows               April 2003


   CauseCode=18 No user responding


   F10 RLC Bob -> NGW 1

   RLC


   F11 480 Temporarily Unavailable NGW 1 -> Proxy 1

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 INVITE
   Error-Info: <sip:temp-unavail-ann@ann.a.example.com>
   Content-Length: 0


   F12 ACK Proxy 1 -> NGW 1

   ACK sip:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F13 480 Temporarily Unavailable F13 Proxy 1 -> Alice

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com


Johnston et al          Expires - October 2002               [Page 54]


                         SIP PSTN Call Flows               April 2003


   CSeq: 1 INVITE
   Error-Info: <sip:temp-unavail-ann@ann.a.example.com>
   Content-Length: 0


   F14 ACK Alice -> Proxy 1

   ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Max-Forwards: 70
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
    ;tag=9fxced76sl
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com
   CSeq: 1 ACK
   Content-Length: 0


































Johnston et al          Expires - October 2002               [Page 55]


                         SIP PSTN Call Flows               April 2003


3.   PSTN to SIP Dialing


   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.

   Since the called SIP User Agent does not send in-band signaling
   information, no early media path needs to be established on the IP
   side.  As a result, the 183 Session Progress response is not used.
   However, NGW 1 will establish a one way speech path prior to call
   completion, and generate ringing for the PSTN caller.  Any tones or
   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.
































Johnston et al          Expires - October 2002               [Page 56]


                         SIP PSTN Call Flows               April 2003


3.1    Successful PSTN to SIP call

   Switch A          NGW 1          Proxy 1           Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |     100  F4    |--------------->|
     |                |<---------------|                |
     |                |                |      180 F5    |
     |                |    180 F6      |<---------------|
     |     ACM F7     |<---------------|                |
     |<---------------|                |                |
     |  One Way Voice |                |                |
     |<===============|                |                |
     |  Ringing Tone  |                |      200 F8    |
     |<===============|    200 F9      |<---------------|
     |                |<---------------|                |
     |                |     ACK F10    |                |
     |     ANM F12    |--------------->|     ACK F11    |
     |<---------------|                |--------------->|
     | Both Way Voice |        Both Way RTP Media       |
     |<==============>|<===============================>|
     |     REL F13    |                |                |
     |--------------->|                |                |
     |     RLC F14    |                |                |
     |<---------------|     BYE F15    |                |
     |                |--------------->|     BYE F16    |
     |                |                |--------------->|
     |                |                |     200 F17    |
     |                |     200 F18    |<---------------|
     |                |<---------------|                |
     |                |                |                |

   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 which is mapped to a BYE by NGW 1.

   Message Details


   F1 IAM Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National



Johnston et al          Expires - October 2002               [Page 57]


                         SIP PSTN Call Flows               April 2003



   F2 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine where B is
   located.  Based upon location analysis the call is forwarded to NGW
   1.  NGW 1  prepares to receive data on port 3456 from Alice.*/

   F3 INVITE Proxy 1 -> Bob

   INVITE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0


Johnston et al          Expires - October 2002               [Page 58]


                         SIP PSTN Call Flows               April 2003


   a=rtpmap:0 PCMU/8000


   F4 100 Trying Bob -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 180 Ringing Bob -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com>
   Content-Length: 0


   F6 180 Ringing Proxy 1 -> NGW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com>
   Content-Length: 0


   F7 ACM NGW 1 -> Alice



Johnston et al          Expires - October 2002               [Page 59]


                         SIP PSTN Call Flows               April 2003


   ACM


   F8 200 OK Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   Contact: <sip:bob@client.b.example.com>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 151

   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 client.b.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com>
   Content-Type: application/sdp
   Content-Length: 151

   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 client.b.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


Johnston et al          Expires - October 2002               [Page 60]


                         SIP PSTN Call Flows               April 2003




   F10 ACK NGW 1 -> Proxy 1

   ACK sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F11 ACK Proxy 1 -> Bob

   ACK sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F12 ANM Bob -> NGW 1

   ANM


   /* RTP streams are established between A and B (via the GW) */

   /* Alice Hangs Up with Bob. */

   F13 REL Alice -> NGW 1

   REL
   CauseCode=16 Normal


   F14 RLC NGW 1 -> Alice

   RLC




Johnston et al          Expires - October 2002               [Page 61]


                         SIP PSTN Call Flows               April 2003


   F15 BYE NGW 1-> Proxy 1

   BYE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F16 BYE Proxy 1 -> Bob

   BYE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F17 200 OK Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F18 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com


Johnston et al          Expires - October 2002               [Page 62]


                         SIP PSTN Call Flows               April 2003


   CSeq: 2 BYE
   Content-Length: 0

















































Johnston et al          Expires - October 2002               [Page 63]


                         SIP PSTN Call Flows               April 2003


3.2    Successful PSTN to SIP call, Fast Answer

   Switch A           NGW 1          Proxy 1           Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |     100  F4    |--------------->|
     |                |<---------------|                |
     |                |                |      200 F5    |
     |                |     200 F6     |<---------------|
     |                |<---------------|                |
     |                |     ACK F7     |                |
     |     ANM F9     |--------------->|     ACK F8     |
     |<---------------|                |--------------->|
     | Both Way Voice |        Both Way RTP Media       |
     |<==============>|<===============================>|
     |     REL F10    |                |                |
     |--------------->|                |                |
     |     RLC F11    |                |                |
     |<---------------|     BYE F12    |                |
     |                |--------------->|     BYE F13    |
     |                |                |--------------->|
     |                |                |     200 F14    |
     |                |     200 F15    |<---------------|
     |                |<---------------|                |
     |                |                |                |

   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.

   Message Details


   F1 IAM Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National


   F2 INVITE NGW 1 -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2


Johnston et al          Expires - October 2002               [Page 64]


                         SIP PSTN Call Flows               April 2003


   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine where B is
   located.  Based upon location analysis the call is forwarded to User
   B.  Bob  prepares to receive data on port 3456 from Alice.*/

   F3 INVITE Proxy 1 -> Bob

   INVITE bob@b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying Proxy 1 -> NGW 1



Johnston et al          Expires - October 2002               [Page 65]


                         SIP PSTN Call Flows               April 2003


   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 200 OK Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 151

   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 client.b.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F6 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 151



Johnston et al          Expires - October 2002               [Page 66]


                         SIP PSTN Call Flows               April 2003


   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 client.b.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 ACK NGW 1 -> Proxy 1

   ACK bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F8 ACK Proxy 1 -> Bob

   ACK bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=130.131.132.14
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F9 ANM Bob -> NGW 1

   ANM


   /* RTP streams are established between A and B (via the GW) */

   /* Alice Hangs Up with Bob. */

   F10 REL ser Alice -> NGW 1

   REL
   CauseCode=16 Normal


Johnston et al          Expires - October 2002               [Page 67]


                         SIP PSTN Call Flows               April 2003




   F11 RLC NGW 1 -> Alice

   RLC


   F12 BYE NGW 1 -> Proxy 1

   BYE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F13 BYE Proxy 1 -> Bob

   BYE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F14 200 OK Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F15 200 OK Proxy 1 -> NGW 1


Johnston et al          Expires - October 2002               [Page 68]


                         SIP PSTN Call Flows               April 2003



   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0










































Johnston et al          Expires - October 2002               [Page 69]


                         SIP PSTN Call Flows               April 2003


3.3    Successful PBX to SIP call

   PBX A            GW 1           Proxy 1           Bob
     |                |                |                |
     |    Seizure     |                |                |
     |--------------->|                |                |
     |      Wink      |                |                |
     |<---------------|                |                |
     |  MF Digits F1  |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |     100  F4    |--------------->|
     |                |<---------------|                |
     |                |                |      180 F5    |
     |                |    180 F6      |<---------------|
     |                |<---------------|                |
     |  One Way Voice |                |                |
     |<===============|                |                |
     |  Ringing Tone  |                |      200 F7    |
     |<===============|     200 F8     |<---------------|
     |                |<---------------|                |
     |                |     ACK F9     |                |
     |     Seizure    |--------------->|     ACK F10    |
     |<---------------|                |--------------->|
     | Both Way Voice |        Both Way RTP Media       |
     |<==============>|<===============================>|
     | Seizure Removal|                |                |
     |--------------->|                |                |
     | Seizure Removal|                |                |
     |<---------------|     BYE F11    |                |
     |                |--------------->|     BYE F12    |
     |                |                |--------------->|
     |                |                |     200 F13    |
     |                |     200 F14    |<---------------|
     |                |<---------------|                |
     |                |                |                |

   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 instead is 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 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.



Johnston et al          Expires - October 2002               [Page 70]


                         SIP PSTN Call Flows               April 2003


   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.

   Message Details


   PBX Alice -> GW 1

   Seizure


   GW 1 -> PBX A

   Wink


   F1 MF Digits PBX Alice -> GW 1

   KP 1 972 555 2222 ST


   F2 INVITE GW 1 -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.a.example.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine where the
   phone number +19725552222 is located.  Based upon location
   analysis the call is forwarded to SIP Bob. */



Johnston et al          Expires - October 2002               [Page 71]


                         SIP PSTN Call Flows               April 2003


   F3 INVITE Proxy 1 -> Bob

   INVITE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.a.example.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying Proxy 1 -> GW 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 180 Ringing Bob -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE


Johnston et al          Expires - October 2002               [Page 72]


                         SIP PSTN Call Flows               April 2003


   Contact: <sip:bob@client.b.example.com>
   Content-Length: 0


   F6 180 Ringing Proxy 1 -> GW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com>
   Content-Length: 0


   /* One way Voice path is established between GW and the PBX for
   ringing. */

   F7 200 OK Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   Contact: <sip:bob@client.b.example.com>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 151

   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 client.b.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 200 OK Proxy 1 -> GW 1

   SIP/2.0 200 OK


Johnston et al          Expires - October 2002               [Page 73]


                         SIP PSTN Call Flows               April 2003


   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com>
   Content-Type: application/sdp
   Content-Length: 151

   v=0
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com
   s=-
   c=IN IP4 client.b.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 ACK GW 1 -> Proxy 1

   ACK sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F10 ACK Proxy 1 -> Bob

   ACK sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Max-Forwards: 69
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between A and B (via the GW) */



Johnston et al          Expires - October 2002               [Page 74]


                         SIP PSTN Call Flows               April 2003


   /* Alice Hangs Up with Bob. */

   F11 BYE GW 1 -> Proxy 1

   BYE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F12 BYE Proxy 1 -> Bob

   BYE sip:bob@client.b.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Max-Forwards: 69
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F13 200 OK Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0


   F14 200 OK Proxy 1 -> GW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159


Johnston et al          Expires - October 2002               [Page 75]


                         SIP PSTN Call Flows               April 2003


   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 2 BYE
   Content-Length: 0
















































Johnston et al          Expires - October 2002               [Page 76]


                         SIP PSTN Call Flows               April 2003


3.4    Unsuccessful PSTN to SIP REL, SIP error mapped to REL

   Switch A            GW 1          Proxy 1           Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|                |
     |                |     604 F3     |                |
     |                |<---------------|                |
     |                |     ACK F4     |                |
     |                |--------------->|                |
     |     REL F5     |                |                |
     |<---------------|                |                |
     |     RLC F6     |                |                |
     |--------------->|                |                |
     |                |                |                |

   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.

   Message Details


   F1 IAM Alice -> GW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-9999,NPI=E.164,NOA=National


   F2 INVITE Alice -> Proxy 1

   INVITE sip:+1972559999@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
   To: <sip:+1972559999@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE
   Contact:
   <sip:+13145551111@gw1.a.example.com;user=phone;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 144

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
   s=-


Johnston et al          Expires - October 2002               [Page 77]


                         SIP PSTN Call Flows               April 2003


   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service to find a route to +1-972-555-
   9999.  A route is not found, so Proxy 1 rejects the call. */

   F3 604 Does Not Exist Anywhere Proxy 1 -> GW 1

   SIP/2.0 604 Does Not Exist Anywhere
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
   To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 INVITE
   Error-Info: <sip:does-not-exist@ann.a.example.com>
   Content-Length: 0


   F4 ACK GW 1 -> Proxy 1

   ACK sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
   To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F5 REL GW 1 -> Alice

   REL
   CauseCode=1


   F6 RLC Alice -> GW 1

   RLC








Johnston et al          Expires - October 2002               [Page 78]


                         SIP PSTN Call Flows               April 2003


3.5    Unsuccessful PSTN to SIP REL, SIP busy mapped to REL

   Switch A          NGW 1           Proxy 1          Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |     100  F4    |--------------->|
     |                |<---------------|                |
     |                |                |      600 F5    |
     |                |                |<---------------|
     |                |                |      ACK F6    |
     |                |     600 F7     |--------------->|
     |                |<---------------|                |
     |                |     ACK F8     |                |
     |                |--------------->|                |
     |   REL(17) F9   |                |                |
     |<---------------|                |                |
     |     RLC F10    |                |                |
     |<-------------->|                |                |
     |                |                |                |

   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 [4].


   Message Details


   F1 IAM Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National


   F2 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70


Johnston et al          Expires - October 2002               [Page 79]


                         SIP PSTN Call Flows               April 2003


   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 144

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +19725552222.  The call is then forwarded to Bob. */

   F3 INVITE F3 Proxy 1 -> Bob

   INVITE bob@b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 144

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying Proxy 1 -> NGW 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2


Johnston et al          Expires - October 2002               [Page 80]


                         SIP PSTN Call Flows               April 2003


    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 600 Busy Everywhere Bob -> Proxy 1

   SIP/2.0 600 Busy Everywhere
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 ACK Proxy 1 -> Bob

   ACK bob@b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F7 600 Busy Everywhere Proxy 1 -> NGW 1

   SIP/2.0 600 Busy Everywhere
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 ACK NGW 1 -> Proxy 1

   ACK bob@b.example.com SIP/2.0


Johnston et al          Expires - October 2002               [Page 81]


                         SIP PSTN Call Flows               April 2003


   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F9 REL NGW 1 -> Alice

   REL
   CauseCode=17 Busy


   F10 RLC Alice -> NGW 1

   RLC

































Johnston et al          Expires - October 2002               [Page 82]


                         SIP PSTN Call Flows               April 2003


3.6    Unsuccessful PSTN->SIP, SIP error interworking to tones

   Switch A          NGW 1           Proxy 1          Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |     100  F4    |--------------->|
     |                |<---------------|                |
     |                |                |      600 F5    |
     |                |                |<---------------|
     |                |                |      ACK F6    |
     |                |     600 F7     |--------------->|
     |                |<---------------|                |
     |                |     ACK F8     |                |
     |     ACM F9     |--------------->|                |
     |<---------------|                |                |
     | One Way Voice  |                |                |
     |<===============|                |                |
     |    Busy Tone   |                |                |
     |<===============|                |                |
     |   REL(16) F10  |                |                |
     |--------------->|                |                |
     |     RLC F11    |                |                |
     |<---------------|                |                |
     |                |                |                |


   In this scenario, Alice calls Bob through Network Gateway NGW1
   and Proxy 1.  The call is routed to Bob by Proxy 1.  The call is
   rejected by the Bob client.  NGW 1 sets up a two way voice path to
   Alice and plays busy tone.  The caller then disconnects

   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.


   Message Details


   F1 IAM Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National


Johnston et al          Expires - October 2002               [Page 83]


                         SIP PSTN Call Flows               April 2003


   CdPN=972-555-2222,NPI=E.164,NOA=National
   Interworking=encountered


   F2 INVITE NGW1 -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +19725552222.  The call is then forwarded to Bob. */

   F3 INVITE Proxy 1 -> Bob

   INVITE bob@b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com


Johnston et al          Expires - October 2002               [Page 84]


                         SIP PSTN Call Flows               April 2003


   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying Bob -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 600 Busy Everywhere Bob -> Proxy 1

   SIP/2.0 600 Busy Everywhere
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 ACK Proxy 1 -> Bob

   ACK bob@b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F7 600 Busy Everywhere Proxy 1 -> NGW 1

   SIP/2.0 600 Busy Everywhere
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2


Johnston et al          Expires - October 2002               [Page 85]


                         SIP PSTN Call Flows               April 2003


    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 ACK NGW 1 -> Proxy 1

   ACK sip:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F9 ACM NGW 1 -> Alice

   ACM


   /* A one way speech path is established between NGW 1 and Alice. */

   /* Call Released after Alice hangs up. */

   F10 REL Alice -> NGW 1

   REL
   CauseCode=16


   F11 RLC NGW 1 -> Alice

   RLC













Johnston et al          Expires - October 2002               [Page 86]


                         SIP PSTN Call Flows               April 2003


3.7    Unsuccessful PSTN->SIP, ACM timeout

   Switch A          NGW 1           Proxy 1          Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |     100  F4    |--------------->|
     |                |<---------------|                |
     |                |                |   INVITE F5    |
     |                |                |--------------->|
     |                |                |   INVITE F6    |
     |                |                |--------------->|
     |                |                |   INVITE F7    |
     |                |                |--------------->|
     |                |                |   INVITE F8    |
     |                |                |--------------->|
     |                |                |   INVITE F9    |
     |                |                |--------------->|
     |     REL F10    |                |                |
     |--------------->|                |                |
     |     RLC F11    |                |                |
     |<---------------|                |                |
     |                |   CANCEL F12   |                |
     |                |--------------->|                |
     |                |     200 F13    |                |
     |                |<---------------|                |

   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.
   Message Details

   F1 IAM Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National

   F2 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>


Johnston et al          Expires - October 2002               [Page 87]


                         SIP PSTN Call Flows               April 2003


   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +19725552222.  The call is then forwarded to Bob. */

   F3 INVITE Proxy 1 -> Bob

   INVITE sip:bob@b.example.com  SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   c c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying Proxy 1 -> NGW 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>


Johnston et al          Expires - October 2002               [Page 88]


                         SIP PSTN Call Flows               April 2003


   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 INVITE Proxy 1 -> Bob

   Same as Message F3


   F6 INVITE Proxy 1 -> Bob

   Same as Message F3


   F7 INVITE Proxy 1 -> Bob

   Same as Message F3


   F8 INVITE Proxy 1 -> Bob

   Same as Message F3


   F9 INVITE Proxy 1 -> Bob

   Same as Message F3


   /* Timer expires in Alice's access network. */

   F10 REL Alice -> NGW 1

   REL
   CauseCode=16 Normal


   F11 RLC NGW 1 -> Alice

   RLC


   F12 CANCEL NGW 1 -> Proxy 1

   CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals


Johnston et al          Expires - October 2002               [Page 89]


                         SIP PSTN Call Flows               April 2003


   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F13 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0



































Johnston et al          Expires - October 2002               [Page 90]


                         SIP PSTN Call Flows               April 2003


3.8    Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy

   Switch A          NGW 1      Stateless Proxy 1     Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |   INVITE F4    |--------------->|
     |                |--------------->|   INVITE F5    |
     |                |   INVITE F6    |--------------->|
     |                |--------------->|   INVITE F7    |
     |                |   INVITE F8    |--------------->|
     |                |--------------->|   INVITE F9    |
     |                |   INVITE F10   |--------------->|
     |                |--------------->|   INVITE F11   |
     |                |   INVITE F12   |--------------->|
     |                |--------------->|   INVITE F13   |
     |                |                |--------------->|
     |     REL F14    |                |                |
     |--------------->|                |                |
     |     RLC F15    |                |                |
     |<---------------|                |                |

   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).


   Message Details


   F1 IAM Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National


   F2 INVITE NGW 1 -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com


Johnston et al          Expires - October 2002               [Page 91]


                         SIP PSTN Call Flows               April 2003


   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +19725552222.  The call is then forwarded to Bob. */

   F3 INVITE Proxy 1 -> Bob

   INVITE sip:bob@b.example.com  SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F5 INVITE Proxy 1 -> Bob

   Same as Message F3


Johnston et al          Expires - October 2002               [Page 92]


                         SIP PSTN Call Flows               April 2003




   F6 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F7 INVITE Proxy 1 -> Bob

   Same as Message F3


   F8 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F9 INVITE Proxy 1 -> Bob

   Same as Message F3


   F10 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F11 INVITE Proxy 1 -> Bob

   Same as Message F3


   F12 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F13 INVITE Proxy 1 -> Bob

   Same as Message F3


   /* A timer expires in Alice's access network. */

   F14 REL Alice -> NGW 1

   REL
   CauseCode=102 Timeout



Johnston et al          Expires - October 2002               [Page 93]


                         SIP PSTN Call Flows               April 2003



   F15 RLC NGW 1 -> Alice

   RLC















































Johnston et al          Expires - October 2002               [Page 94]


                         SIP PSTN Call Flows               April 2003


3.9    Unsuccessful PSTN->SIP, Caller Abandonment

   Switch A          NGW 1          Proxy 1           Bob
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |     100  F4    |--------------->|
     |                |<---------------|                |
     |                |                |      180 F5    |
     |                |    180 F6      |<---------------|
     |     ACM F7     |<---------------|                |
     |<---------------|                |                |
     |  One Way Voice |                |                |
     |<===============|                |                |
     |  Ringing Tone  |                |                |
     |<===============|                |                |
     |                |                |                |
     |     REL F8     |                |                |
     |--------------->|                |                |
     |     RLC F9     |                |                |
     |<---------------|   CANCEL F10   |                |
     |                |--------------->|                |
     |                |     200 F11    |                |
     |                |<---------------|                |
     |                |                |   CANCEL F12   |
     |                |                |--------------->|
     |                |                |     200 F13    |
     |                |                |<---------------|
     |                |                |     487 F14    |
     |                |                |<---------------|
     |                |                |     ACK F15    |
     |                |     487 F16    |--------------->|
     |                |<---------------|                |
     |                |     ACK F17    |                |
     |                |--------------->|                |
     |                |                |                |


   In this scenario, Alice calls Bob through NGW 1 and Proxy 1.
   Bob does not respond with 200 OK.  NGW 1 plays ringing tone since
   the ACM indicates that interworking has been encountered.  Alice
   disconnects the call with a Release message REL which is mapped by
   NGW 1 to a CANCEL.  Note that if Bob had sent a 200 OK response
   after the REL, NGW 1 would have sent an ACK then a BYE to properly
   terminate the call.


   Message Details


Johnston et al          Expires - October 2002               [Page 95]


                         SIP PSTN Call Flows               April 2003




   F1 IAM Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National


   F2 INVITE Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +19725552222.  The call is then forwarded to Bob. */

   F3 INVITE Proxy 1 -> Bob

   INVITE sip:bob@b.example.com  SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:ngw1@a.example.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 146


Johnston et al          Expires - October 2002               [Page 96]


                         SIP PSTN Call Flows               April 2003



   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 100 Trying Bob -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 180 Ringing Bob -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com;transport=tcp>
   Content-Length: 0


   F6 180 Ringing Proxy 1 -> NGW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com


Johnston et al          Expires - October 2002               [Page 97]


                         SIP PSTN Call Flows               April 2003


   CSeq: 1 INVITE
   Contact: <sip:bob@client.b.example.com>
   Content-Length: 0


   F7 ACM NGW 1 -> Alice

   ACM


   /* Alice hangs up */

   F8 REL Alice -> NGW 1

   REL
   CauseCode=16 Normal


   F9 RLC NGW 1 -> Alice

   RLC


   F10 CANCEL NGW 1 -> Proxy 1

   CANCEL sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F11 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F12 CANCEL Proxy 1 -> Bob



Johnston et al          Expires - October 2002               [Page 98]


                         SIP PSTN Call Flows               April 2003


   CANCEL sip:bob@b.example.com SIP/2.0
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F13 200 OK Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F14 487 Request Terminated Bob -> Proxy 1

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F15 ACK Proxy 1 -> Bob

   ACK sip:bob@b.example.com SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F16 487 Request Terminated Proxy 1 -> NGW 1


Johnston et al          Expires - October 2002               [Page 99]


                         SIP PSTN Call Flows               April 2003



   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F17 ACK NGW 1 -> Proxy 1

   ACK sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0






























Johnston et al          Expires - October 2002              [Page 100]


                         SIP PSTN Call Flows               April 2003


4.   PSTN to PSTN Dialing via SIP Network

   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 [4].

   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 in-band alerting may
   flow from the called party telephone switch to the caller.


































Johnston et al          Expires - October 2002              [Page 101]


                         SIP PSTN Call Flows               April 2003


4.1    Successful ISUP PSTN to ISUP PSTN call

   Switch A       NGW 1         Proxy 1         GW 2         Switch C
    |              |              |              |              |
    |     IAM F1   |              |              |              |
    |------------->|              |              |              |
    |              |  INVITE F2   |              |              |
    |              |------------->|  INVITE F3   |              |
    |              |              |------------->|     IAM F4   |
    |              |              |              |------------->|
    |              |              |              |     ACM F5   |
    |              |              |   183 F6     |<-------------|
    |              |    183 F7    |<-------------|              |
    |    ACM F8    |<-------------|              |              |
    |<-------------|              |              |              |
    | One Way Voice|      Two Way RTP Media      | One Way Voice|
    |<=============|<===========================>|<=============|
    |              |              |              |    ANM F9    |
    |              |              |   200 F10    |<-------------|
    |              |    200 F11   |<-------------|              |
    |    ANM F12   |<-------------|              |              |
    |<-------------|              |              |              |
    |              |    ACK F13   |              |              |
    |              |------------->|    ACK F14   |              |
    |              |              |------------->|              |
    |Both Way Voice|     Both Way RTP Media      |Both Way Voice|
    |<=============|<===========================>|<=============|
    |              |              |              |    REL F15   |
    |              |              |              |<-------------|
    |              |              |   BYE F16    |              |
    |              |    BYE F18   |<-------------|    RLC F17   |
    |              |<-------------|              |------------->|
    |              |              |              |              |
    |              |    200 F19   |              |              |
    |              |------------->|    200 F20   |              |
    |              |              |------------->|              |
    |    REL F21   |              |              |              |
    |<-------------|              |              |              |
    |    RLC F22   |              |              |              |
    |------------->|              |              |              |
    |              |              |              |              |


   In this scenario, Alice in the PSTN calls Carol who is an extension
   on a PBX.  Alice's telephone switch signals via SS7 to the Network
   Gateway NGW 1, while Carol's PBX signals via SS7 with the
   Gateway GW 2.  The CdPN and CgPN are mapped by GW1 into SIP URIs and
   placed in the To and From headers.  Proxy 1 looks up the dialed
   digits in the Request-URI and maps the digits to the PBX extension of


Johnston et al          Expires - October 2002              [Page 102]


                         SIP PSTN Call Flows               April 2003


   Carol which 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.

   Message Details


   F1 IAM Switch Alice -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=918-555-3333,NPI=E.164,NOA=National


   F2 INVITE NGW 1 -> Proxy 1

   INVITE sips:+19185553333@ss1.a.example.com;user=phone  SIP/2.0
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sips:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 consults Location Service and translates the dialed number
   to a private number in the Request-URI*/

   F3 INVITE Proxy 1 -> GW 2

   INVITE sips:4443333@gw2.a.example.com SIP/2.0
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKwqwee65


Johnston et al          Expires - October 2002              [Page 103]


                         SIP PSTN Call Flows               April 2003


    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sips:ss1.a.example.com;lr>
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sips:ngw1@a.example.com>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com
   s=-
   c=IN IP4 ngw1.a.example.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 IAM GW 2 -> Switch C

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=444-3333,NPI=Private,NOA=Subscriber


   F5 ACM Switch C -> GW 2

   ACM


   /* Based on the ACM message, GW 2 returns a 183 response.  In-band
   call progress indications are sent to Alice through NGW 1. */

   F6 183 Session Progress GW 2 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sips:ss1.a.example.com;lr>
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sips:4443333@gw2.a.example.com>
   Content-Type: application/sdp


Johnston et al          Expires - October 2002              [Page 104]


                         SIP PSTN Call Flows               April 2003


   Content-Length: 143

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com
   s=-
   c=IN IP4 gw2.a.example.com
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 183 Session Progress Proxy 1 -> GW 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sips:ss1.a.example.com;lr>
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sips:4443333@gw2.a.example.com>
   Content-Type: application/sdp
   Content-Length: 143

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com
   s=-
   c=IN IP4 gw2.a.example.com
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* NGW 1 receives packets from GW 2 with encoded ringback, tones or
   other audio.  NGW 1 decodes this and places it on the originating
   trunk. */

   F8 ACM NGW 1 -> Switch A

   ACM


   /* Bob answers */

   F9 ANM Switch C -> GW 2

   ANM



Johnston et al          Expires - October 2002              [Page 105]


                         SIP PSTN Call Flows               April 2003



   F10 200 OK GW 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sips:ss1.a.example.com;lr>
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sips:4443333@gw2.a.example.com>
   Content-Type: application/sdp
   Content-Length: 143

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com
   s=-
   c=IN IP4 gw2.a.example.com
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sips:ss1.a.example.com;lr>
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sips:4443333@gw2.a.example.com>
   Content-Type: application/sdp
   Content-Length: 143

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com
   s=-
   c=IN IP4 gw2.a.example.com
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000




Johnston et al          Expires - October 2002              [Page 106]


                         SIP PSTN Call Flows               April 2003



   F12 ANM NGW 1 -> Switch A

   ANM


   F13 ACK NGW 1 -> Proxy 1

   ACK sips:4443333@gw2.a.example.com SIP/2.0
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sips:ss1.a.example.com;lr>
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F14 ACK Proxy 1 -> GW 2

   ACK sips:4443333@gw2.a.example.com SIP/2.0
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between NGW 1 and GW 2. */

   /* Bob Hangs Up with Alice. */

   F15 REL Switch C -> GW 2

   REL
   CauseCode=16 Normal


   F16 BYE GW 2 -> Proxy 1

   BYE sips:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
   Max-Forwards: 70
   Route: <sips:ss1.a.example.com;lr>


Johnston et al          Expires - October 2002              [Page 107]


                         SIP PSTN Call Flows               April 2003


   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 4 BYE
   Content-Length: 0


   F17 RLC GW 2 -> Switch C

   RLC


   F18 BYE Proxy 1 -> NGW 1

   BYE sips:ngw1@a.example.com SIP/2.0
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
    ;received=192.0.2.202
   Max-Forwards: 69
   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 4 BYE
   Content-Length: 0


   F19 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
    ;received=192.0.2.202
   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 4 BYE
   Content-Length: 0


   F20 200 OK Proxy 1 -> GW 2

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
    ;received=192.0.2.202
   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
   CSeq: 4 BYE


Johnston et al          Expires - October 2002              [Page 108]


                         SIP PSTN Call Flows               April 2003


   Content-Length: 0


   F21 REL Switch C -> GW 2

   REL
   CauseCode=16 Normal


   F22 RLC GW 2 -> Switch C

   RLC







































Johnston et al          Expires - October 2002              [Page 109]


                         SIP PSTN Call Flows               April 2003


4.2    Successful FGB PBX to ISDN PBX call with overflow

 PBX A       GW 1        Proxy 1        GW 2         GW 3        PBX C
   |            |            |            |            |            |
   |  Seizure   |            |            |            |            |
   |----------->|            |            |            |            |
   |    Wink    |            |            |            |            |
   |<-----------|            |            |            |            |
   |MF Digits F1|            |            |            |            |
   |----------->|            |            |            |            |
   |            | INVITE F2  |            |            |            |
   |            |----------->| INVITE F3  |            |            |
   |            |            |----------->|            |            |
   |            |            |   503 F4   |            |            |
   |            |            |<-----------|            |            |
   |            |            |   ACK F5   |            |            |
   |            |            |----------->|            |            |
   |            |            |  INVITE F6              |            |
   |            |            |------------------------>|  SETUP F7  |
   |            |            |          100  F8        |----------->|
   |            |            |<------------------------|CALL PROC F9|
   |            |            |                         |<-----------|
   |            |            |                         | ALERT F10  |
   |            |            |          180 F11        |<-----------|
   |            |  180 F12   |<------------------------|            |
   |            |<-----------|                         |            |
   | Ringtone   |            |                         |OneWay Voice|
   |<===========|            |                         |<===========|
   |            |            |                         | CONNect F13|
   |            |            |         200 F14         |<-----------|
   |            |  200 F15   |<------------------------|            |
   |  Seizure   |<-----------|                         |            |
   |<-----------|  ACK F16   |                         |            |
   |            |----------->|         ACK F17         |            |
   |            |            |------------------------>|CONN ACK F18|
   |            |            |                         |----------->|
   |BothWayVoice|          Both Way RTP Media          |BothWayVoice|
   |<==========>|<====================================>|<==========>|
   |            |            |                         |  DISC F19  |
   |            |            |                         |<-----------|
   |            |            |         BYE F20         |            |
   |            |  BYE F21   |<------------------------|  REL F22   |
   |Seiz Removal|<-----------|                         |----------->|
   |<-----------|  200 F23   |                         |            |
   |Seiz Removal|----------->|         200 F24         |            |
   |----------->|            |------------------------>| REL COM F25|
   |            |            |                         |<-----------|
   |            |            |                         |            |



Johnston et al          Expires - October 2002              [Page 110]


                         SIP PSTN Call Flows               April 2003




   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.


   Message Details

   PBX Alice -> GW 1

   Seizure


   GW 1 -> PBX A

   Wink


   F1 MF Digits PBX Alice -> GW 1

   KP 444 3333 ST


   F2 INVITE GW 1 -> Proxy 1

   INVITE sip:4443333@ss1.a.example.com SIP/2.0
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.a.example.com>
   Content-Type: application/sdp
   Content-Length: 155

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


Johnston et al          Expires - October 2002              [Page 111]


                         SIP PSTN Call Flows               April 2003




   /* Proxy 1 uses a Location Service function to determine where B is
   located.  Response is returned listing alternative routes, GW2 and
   GW3, which are then tried sequentially. */

   F3 INVITE Proxy 1 -> GW 2

   INVITE sip:4443333@gw2.a.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.a.example.com>
   Content-Type: application/sdp
   Content-Length: 155

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 503 Service Unavailable GW 2 -> Proxy 1

   SIP/2.0 503 Service Unavailable
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 ACK Proxy 1 -> GW 2

   ACK sip:4443333@ss1.a.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1


Johnston et al          Expires - October 2002              [Page 112]


                         SIP PSTN Call Flows               April 2003


   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Max-Forward: 70
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F6 INVITE Proxy 1 -> GW 3

   INVITE sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.a.example.com>
   Content-Type: application/sdp
   Content-Length: 155

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com
   s=-
   c=IN IP4 gw1.a.example.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 SETUP GW 3 -> PBX C

   Protocol discriminator=Q.931
   Message type=SETUP
   Bearer capability: Information transfer capability=0 (Speech) or 16
   (3.1 kHz audio)
   Channel identification=Preferred or exclusive B-channel
   Progress indicator=1 (Call is not end-to-end ISDN; further call
   progress information may be available inband)
   Called party number:
   Type of number and numbering plan ID=33 (National number in ISDN
   numbering plan)
   Digits=918-555-3333



Johnston et al          Expires - October 2002              [Page 113]


                         SIP PSTN Call Flows               April 2003



   F8 100 Trying GW 3 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 CALL PROCeeding PBX C -> GW 3

   Protocol discriminator=Q.931
   Message type=CALL PROC


   F10 ALERT PBX C -> GW 3

   Protocol discriminator=Q.931
   Message type=PROG


   /* Based on ALERT message, GW 3 returns a 180 response. */

   F11 180 Ringing GW 3 -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
   Content-Length: 0


   F12 180 Ringing Proxy 1 -> GW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>


Johnston et al          Expires - October 2002              [Page 114]


                         SIP PSTN Call Flows               April 2003


   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
   Content-Length: 0


   F13 CONNect PBX C -> GW 3

   Protocol discriminator=Q.931
   Message type=CONN


   F14 200 OK GW 3 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE
   Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 143

   v=0
   o=GW 987654321 987654321 IN IP4 gw3.a.example.com
   s=-
   c=IN IP4 gw3.a.example.com
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F15 200 OK Proxy 1 -> GW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Record-Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 INVITE


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                         SIP PSTN Call Flows               April 2003


   Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 143

   v=0
   o=GW 987654321 987654321 IN IP4 gw3.a.example.com
   s=-
   c=IN IP4 gw3.a.example.com
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   GW 1 -> PBX A

   Seizure


   F16 ACK GW 1 -> Proxy 1

   ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F17 ACK Proxy 1 -> GW 3

   ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
    ;received=192.0.2.201
   Max-Forwards: 69
   From: <sip:551313@gw1.a.example.com>;tag=63412s
   To: <sip:4443333@ss1.a.example.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 ACK
   Content-Length: 0


   F18 CONNect ACK GW 3 -> PBX C

   Protocol discriminator=Q.931
   Message type=CONN ACK


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                         SIP PSTN Call Flows               April 2003




   /* RTP streams are established between GW 1 and GW 3. */

   /* Bob Hangs Up with Alice. */

   F19 DISConnect PBX C -> GW 3

   Protocol discriminator=Q.931
   Message type=DISC
   Cause=16 (Normal clearing)


   F20 BYE GW 3 -> Proxy 1

   BYE sip:551313@gw1.a.example.com SIP/2.0
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
   Max-Forwards: 70
   Route: <sip:ss1.a.example.com;lr>
   From: <sip:4443333@ss1.a.example.com>;tag=123456789
   To: <sip:551313@gw1.a.example.com>;tag=63412s
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 BYE
   Content-Length: 0


   F21 BYE Proxy 1 -> GW 1

   BYE sip:551313@gw1.a.example.com SIP/2.0
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
    ;received=192.0.2.203
   Max-Forwards: 69
   From: <sip:4443333@ss1.a.example.com>;tag=123456789
   To: <sip:551313@gw1.a.example.com>;tag=63412s
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 BYE
   Content-Length: 0


   GW 1 -> PBX A

   Seizure removal


   F22 RELease GW 3 -> PBX C

   Protocol discriminator=Q.931
   Message type=REL


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                         SIP PSTN Call Flows               April 2003




   F23 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
    ;received=192.0.2.203
   From: <sip:4443333@ss1.a.example.com>;tag=123456789
   To: <sip:551313@gw1.a.example.com>;tag=63412s
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 BYE
   Content-Length: 0


   F24 200 OK Proxy 1 -> GW 3

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
    ;received=192.0.2.203
   From: <sip:4443333@ss1.a.example.com>;tag=123456789
   To: <sip:551313@gw1.a.example.com>;tag=63412s
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
   CSeq: 1 BYE
   Content-Length: 0


   F25 RELease COMplete PBX C -> GW 3

   Protocol discriminator=Q.931
   Message type=REL COM


   PBX Alice -> GW 1

   Seizure removal




Security Considerations

   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 zzzz [4] which is summarized
   here.




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                         SIP PSTN Call Flows               April 2003


   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 including 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 [8] 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
   [2]) 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.



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                         SIP PSTN Call Flows               April 2003


Normative References


   1  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997

    2 Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
      Peterson, J., Sparks, R., Handley, M., and Schooler, E., "SIP:
      Session Initiation Protocol", RFC 3261, June 2002.

    3 Rosenberg, J. and Schulzrinne, H., "An Offer/Answer Model with
      SDP", Internet Engineering Task Force, RFC 3264, April 2002.

   4 G. Camarillo, A. Roach, J. Peterson, L. Ong, "ISUP to SIP
      Mapping", Internet Draft, Internet Engineering Task Force, Work in
      progress. August 2002.

   5 Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach,
      P., Luotonen, A. and L. Stewart, "HTTP authentication: Basic and
      Digest Access Authentication", RFC 2617, June 1999.

   6 J. Rosenberg, H. Schulzrinne, and G. Camarillo, "The Stream
      Control Transmission Protocol as a Transport for the Session
      Initiation Protocol," Internet Draft, Internet Engineering Task
      Force, Work in progress. June 2002.

   7 A. Vaha-Sipila, "URLs for Telephone Calls", Internet Draft,
      Internet Engineering Task Force, RFC 2806, April 2000.

   8 A. Vemuri and J. Peterson, "Session Initiation Protocol for
      Telephones (SIP-T): Context and Architectures," RFC 3372,
      September 2002.

   9  E. Zimmerer, J. Peterson, A. Vemuri, L. Ong, F. Audet, M. Watson,
      M. Zonoun, "MIME media types for ISUP and QSIG Objects," RFC 3204,
      December 2001.

   10 P. Faltstrom, "E.164 Numbers and DNS," RFC 2916, September 2000.

Informative References


   11 Johnston, A., Donovan, S., Sparks, R., Cunningham, C., Summers,
      K., "Session Initiation Protocol Basic Call Flow Examples", RFC
      yyyv, August 2002.






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Acknowledgments

   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.

Author's Addresses

   All listed authors actively contributed large amounts of text to this
   document.

      Alan Johnston
      WorldCom
      100 South 4th Street
      St. Louis, MO 63102
      USA

      EMail:  alan.johnston@wcom.com


      Steve Donovan
      dynamicsoft, Inc.
      5100 Tennyson Parkway
      Suite 1200
      Plano, Texas 75024
      USA

      EMail:  sdonovan@dynamicsoft.com




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                         SIP PSTN Call Flows               April 2003


      Robert Sparks
      dynamicsoft, Inc.
      5100 Tennyson Parkway
      Suite 1200
      Plano, Texas 75024
      USA

      EMail:  rsparks@dynamicsoft.com

      Chris Cunningham
      dynamicsoft, Inc.
      5100 Tennyson Parkway
      Suite 1200
      Plano, Texas 75024
      USA

      EMail: ccunningham@dynamicsoft.com


      Kevin Summers
      Sonus
      1701 North Collins Blvd, Suite 3000
      Richardson, TX 75080
      USA

      Email: kevin.summers@sonusnet.com


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   this standard. Please address the information to the IETF Executive
   Director.


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                         SIP PSTN Call Flows               April 2003




Full Copyright Statement

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   This document and translations of it may be copied and furnished to
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