SIPPING Working Group                                    A. Johnston
   Internet Draft                                              WorldCom
   Document: draft-ietf-sipping-pstn-call-flows-00.txt draft-ietf-sipping-pstn-call-flows-01.txt       S. Donovan
   Expires: February April 2003                                        R. Sparks
                                                          C. Cunningham
                                                            dynamicsoft
                                                             K. Summers
                                                                  Sonus
                                                            August
                                                          November 2002

                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 informational document gives 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 and Clients, 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), ANSI 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...................................4 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 Unsuccessful SIP to PSTN call: Treatment from PSTN........32
      2.5 Unsuccessful SIP to PSTN: REL w/Cause from PSTN...........39
      2.6 Unsuccessful SIP to PSTN: ANM Timeout.....................44
   3. PSTN to SIP Dialing...........................................50
      3.1 Successful PSTN to SIP call...............................52 call...............................51
      3.2 Successful PSTN to SIP call, Fast Answer..................59 Answer..................58
      3.3 Successful PBX to SIP call................................65 call................................64
      3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL.....72 REL.....71
      3.5 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL......74 REL......73
      3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones...78 tones...77
      3.7 Unsuccessful PSTN->SIP, ACM timeout.......................82 timeout.......................81
      3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy......86 Proxy......85
      3.9 Unsuccessful PSTN->SIP, Caller Abandonment................90 Abandonment................89
   4. PSTN to PSTN Dialing via SIP Network..........................96 Network..........................95
      4.1 Successful ISUP PSTN to ISUP PSTN call....................97 call....................96
      4.2 Successful FGB PBX to ISDN PBX call with overflow........105 overflow........104
   Security Considerations.........................................113
   References......................................................113
   Acknowledgments.................................................114 Considerations.........................................112
   Normative References............................................114
   Informative References..........................................114
   Acknowledgments.................................................115
   Author's Addresses..............................................115

1.   Overview

   The call flows shown in this document were developed in the design of
   a carrier-class SIP IP Telephony communications network.  They represent an example
   minimum set of functionality for SIP to be used in IP Telephony
   applications. functionality.

   It is the hope of the authors that this document will be useful for
   SIP implementors, designers, and protocol researchers alike and will
   help further the goal of a standard SIP implementation for IP
   Telephony.  It is envisioned that as changes to the standard of RFC 3261 [2].
   These flows represent carefully checked and
   additional RFCs are added that this document will reflect those working group reviewed

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                         SIP PSTN Call Flows              August            November 2002

   changes and represent the current state

   scenarios of the most common SIP/PSTN interworking examples as a standard interoperable
   SIP IP Telephony implementation.
   companion to the specifications.

   These call flows are based on the current version 2.0 of SIP in
   RFC 3261[2] 3261 [2] with SDP usage described in RFC 3264[3].

   Note that 3264 [3]. Other RFCs
   also comprise the SIP standard but are not used in this document is informational, and set of basic
   call flows. The SIP/ISUP mapping is NOT NORMATIVE based on any
   aspect of SIP or SIP/PSTN interworking. RFC zzzz [4].

   Various PSTN signaling protocols are illustrated in this document:
   ISDN (Integrated Services Digital Network), ANSI ISUP (ISDN User
   Part) and FGB (Feature Group B) circuit associated signaling.  They
   were chosen  This
   document shows mainly ANSI ISUP due to illustrate its practical origins.
   However, as used in this document, the nature of SIP/PSTN interworking - they
   are not a complete or even representative set.  Also, some details
   and parameters of these PSTN protocols have been omitted.  For full
   information about SIP usage is virtually identical
   to the ITU-T International ISUP mapping, refer to used as the reference in [4].

   Basic SIP call flow examples are contained in a companion document,
   RFC
   yyyy[5]. yyyy [10].

1.1     General Assumptions

   A number of architecture, network, and protocol assumptions underly 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 [6]. [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.  Other transport schemes  SCTP [6] could
   also be used.

   Throughout this document  See the call flows show a network where the
   proxy servers authenticate users discussion in RFC 3261 [2] for details on behalf of gateways.  Gateways may
   also authenticate users directly. Both of these are reasonable usages
   of the
   transport issues for SIP. If gateways do not authenticate directly they would be to
   refuse requests from entities other than trusted proxy servers with
   which they have effective channel security (for example [7] or [8])."

   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
                         SIP PSTN Call Flows              August 2002
   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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                         SIP PSTN Call Flows            November 2002

   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 [9].

1.2     Legend for Message Flows

   Dashed lines (---) represent signaling messages that are mandatory to [7].

   This document shows typical examples of SIP/ISUP interworking.
   Although in the call scenario. These messages can be SIP or PSTN 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 is informational only and is NOT NORMATIVE in any
   sense.
   sense, in that it does not prescribe the flows that are shown, indeed
   they MUST NOT be copied due to the reasons described in the next
   paragraph.  On the other hand, these SIP/PSTN call flows represent
   well-reviewed examples of SIP/PSTN interworking usage that are best
   common practice according to community 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
                         SIP PSTN Call Flows              August 2002
   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    BigGuy         UserA@atlanta.com    192.168.100.101    Alice          sip:alice@atlanta.com    192.0.2.101
   User Agent    LittleGuy      UserB@biloxi.com     192.168.200.201    Bob            sip:bob@biloxi.com       192.0.2.200
   Proxy Server                 ss1.atlanta.com      192.168.255.111                 sip:ss1.atlanta.com      192.0.2.111
   User Agent (Gateway)         gw1.atlanta.com      192.168.255.201         sip:gw1.atlanta.com      192.0.2.201
   User Agent (Gateway)         gw2.atlanta.com      192.168.255.202         sip:gw2.atlanta.com      192.0.2.202
   User Agent (Gateway)         gw3.atlanta.com      192.168.255.203         sip:gw3.atlanta.com      192.0.2.203
   User Agent (Gateway)         ngw1.atlanta.com     192.168.255.101         sip:ngw1.atlanta.com     192.0.2.103
   User Agent (Gateway)         ngw2.atlanta.com     192.168.255.102         sip:ngw2.atlanta.com     192.0.2.102

   Note that NGW 1 and NGW 2 also have a device URIs (Contacts) of
   sip:ngw1@atlanta.com and sip:ngw2@atlanta.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, User A (BigGuy sip:UserA@atlanta.com) Alice (Alice sip:alice@atlanta.com) is a
   SIP phone or other SIP-enabled device.  User B  Bob is reachable via the
   PSTN at global telephone number +19725552222. User A Alice places a call
   to User B Bob through a Proxy Server Proxy 1 and a Network Gateway.  In
   other scenarios, User A Alice places calls to User C, 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 (CgPN in ISUP). 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 CgPN calling party number
   in the PSTN.

   In these scenarios, User A Alice is a SIP phone or other SIP-enabled
   device.  User A  Alice places a call to User B Bob in the PSTN or User C 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 User A 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

   User A

   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    |
     |                |                |<---------------|
     |                |                |                |

   User A

   Alice dials the globalized E.164 number +19725552222 to reach
   User B.
   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.

   User A

   Alice could use either their SIP address (sip:UserA@atlanta.com) (sip:alice@atlanta.com) or
   SIP telephone number (sip:+13145551111@ss1.atlanta.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 identification

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   through the Network Gateway (NGW 1) to User B 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, User B Bob answers the call then User A Alice disconnects the
   call.  Signaling between NGW 1 and User B's 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@atlanta.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.atlanta.com) which then
   resolves it to sip:ngw1.atlanta.com which is the address of NGW 1.

   This flow shows TCP transport.

   Message Details

   F1 INVITE A Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Proxy-Authorization: Digest username="UserA", username="alice", realm="atlanta.com",
    nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
    uri="sip:+19725552222@ss1.atlanta.com;user=phone",
    response="ccdca50cb091d587421457305d097458c"
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

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   F2 100 Trying Proxy 1 -> User A Alice

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
                         SIP PSTN Call Flows              August 2002
   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.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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 SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111

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

   F5 IAM NGW 1 -> User B
                         SIP PSTN Call Flows              August 2002 Bob

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

   F6 ACM User B Bob -> NGW 1

   ACM

   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 -> User A Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
                         SIP PSTN Call Flows              August 2002
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F9 ANM User B Bob -> NGW 1

   ANM

   F10 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

Johnston et al            Expires - May 2002                 [Page 11] 
                         SIP PSTN Call Flows            November 2002

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F11 200 OK Proxy 1 -> User A
                         SIP PSTN Call Flows              August 2002 Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F12 ACK A Alice -> Proxy 1

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

Johnston et al            Expires - May 2002                 [Page 12] 
                         SIP PSTN Call Flows            November 2002

   F13 ACK Proxy 1 -> NGW 1

   ACK sip:+19725552222@ngw1.atlanta.com;user=phone sip:ngw1@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
                         SIP PSTN Call Flows              August 2002
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

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

   F14 BYE A Alice -> Proxy 1

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

   F15 BYE Proxy 1 -> NGW 1

   BYE sip:+19725552222@ngw1.atlanta.com;user=phone sip:ngw1@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

Johnston et al            Expires - May 2002                 [Page 13] 
                         SIP PSTN Call Flows            November 2002

   F16 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
                         SIP PSTN Call Flows              August 2002
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F17 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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 - May 2002                 [Page 14] 
                         SIP PSTN Call Flows              August            November 2002

2.2    Successful SIP to ISDN PBX call

   User A

   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|
     |                |                |--------------->|
     |                |                |                |

   User A

   Alice is a SIP device while User C Carol is connected via a
   Gateway (GW 1) to a PBX.  The PBX connection is via a ISDN trunk
   group.  User A  Alice dials User C's 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 - May 2002                 [Page 15] 
                         SIP PSTN Call Flows              August            November 2002

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

   Message

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

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

   Message Details

   F1 INVITE A Alice -> Proxy 1

   INVITE sip:+19185553333@ss1.atlanta.com;user=phone sips:+19185553333@ss1.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@192.168.100.101> <sips:alice@client.atlanta.com>
   Proxy-Authorization: Digest username="UserA", username="alice",
    realm="atlanta.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",
    opaque="", uri="sip:+19185553333@ss1.atlanta.com;user=phone", uri="sips:+19185553333@ss1.atlanta.com;user=phone",
    response="6c792f5c9fa360358b93c7fb826bf550"
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F2 100 Trying Proxy 1 -> User A Alice

   SIP/2.0 100 Trying

Johnston et al            Expires - May 2002                 [Page 16] 
                         SIP PSTN Call Flows            November 2002

   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
                         SIP PSTN Call Flows              August 2002
   CSeq: 2 INVITE
   Content-Length: 0

   F3 INVITE Proxy 1 -> GW 1

   INVITE sip:4443333@gw1.atlanta.com sips:4443333@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@192.168.100.101> <sips:alice@client.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Content-Length: 0

Johnston et al            Expires - May 2002                 [Page 17] 
                         SIP PSTN Call Flows            November 2002

   F5 SETUP GW 1 -> User C Carol

   Protocol discriminator=Q.931
   Message type=SETUP
   Bearer capability: Information transfer capability=0 (Speech) or 16
   (3.1 kHz audio)
                         SIP PSTN Call Flows              August 2002
   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 User C -> Carol-> GW 1

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

   F7 PROGress User C -> 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/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:4443333@gw1.atlanta.com> <sips:4443333@gw1.atlanta.com>
   Content-Length: 0

   F9 180 Ringing Proxy 1 -> User A Alice

Johnston et al            Expires - May 2002                 [Page 18] 
                         SIP PSTN Call Flows            November 2002

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
                         SIP PSTN Call Flows              August 2002
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:4443333@gw1.atlanta.com> <sips:4443333@gw1.atlanta.com>
   Content-Length: 0

   F10 CONNect User C -> Carol-> GW 1

   Protocol discriminator=Q.931
   Message type=CONN

   F11 CONNect ACK GW 1 -> User C 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/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:4443333@gw1.atlanta.com> <sips:4443333@gw1.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.com
   t=0 0

Johnston et al            Expires - May 2002                 [Page 19] 
                         SIP PSTN Call Flows            November 2002

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

   F13 200 OK Proxy 1 -> User A
                         SIP PSTN Call Flows              August 2002 Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:4443333@gw1.atlanta.com> <sips:4443333@gw1.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F14 ACK A Alice -> Proxy 1

   ACK sip:4443333@gw1.atlanta.com sips:4443333@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 ACK
   Content-Length: 0

   F15 ACK Proxy 1 -> GW 1

   ACK sip:4443333@gw1.atlanta.com sips:4443333@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9

Johnston et al            Expires - May 2002                 [Page 20] 
                         SIP PSTN Call Flows            November 2002

    ;received=192.0.2.101
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
                         SIP PSTN Call Flows              August 2002
   CSeq: 2 ACK
   Content-Length: 0

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

   F16 BYE A Alice -> Proxy 1

   BYE sip:4443333@gw1.atlanta.com sips:4443333@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 3 BYE
   Content-Length: 0

   F17 BYE Proxy 1 -> GW 1

   BYE sip:4443333@gw1.atlanta.com sips:4443333@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 3 BYE
   Content-Length: 0

   F18 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111

Johnston et al            Expires - May 2002                 [Page 21] 
                         SIP PSTN Call Flows            November 2002

   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
                         SIP PSTN Call Flows              August 2002
   CSeq: 3 BYE
   Content-Length: 0

   F19 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101 SIP/2.0/TLS client.atlanta.com:5061;branch=z9hG4bK74bf9
    ;received=192.0.2.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone> Alice <sips:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone> Carol <sips:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 3 BYE
   Content-Length: 0

   F20 DISConnect GW 1 -> User C Carol

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

   F21 RELease User C -> Carol-> GW 1

   Protocol discriminator=Q.931
   Message type=REL

   F22 RELease COMplete GW 1 -> User C Carol

   Protocol discriminator=Q.931
   Message type=REL COM

Johnston et al            Expires - May 2002                 [Page 22] 
                         SIP PSTN Call Flows              August            November 2002

2.3    Successful SIP to ISUP PSTN call with overflow

   User A

   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   |
    |              |                             |<-------------|
    |              |                             |              |

   User A

   Alice calls User B 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.  User B  Bob answers the call.  The call is terminated when User A Alice
   disconnects the call.  NGW 2 and User B's Bob's telephone switch use ANSI
   ISUP signaling.

   Message Details

   NGW 2 also only accepts SIP PSTN Call Flows              August 2002

   F1 messages that come through Proxy 1, so
   the Contact URI sip:ngw2@atlanta.com is used in this flow.

Johnston et al            Expires - May 2002                 [Page 23] 
                         SIP PSTN Call Flows            November 2002

   This flow shows UDP transport.

   Message Details

   F1 INVITE A Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com>
   Proxy-Authorization: Digest username="UserA", username="alice",
    realm="atlanta.com", nonce="b59311c3ba05b401cf80b2a2c5ac51b0",
    opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
    response="ba6ab44923fa2614b28e3e3957789ab0"
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>

Johnston et al            Expires - May 2002                 [Page 24] 
                         SIP PSTN Call Flows            November 2002

   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 147
                         SIP PSTN Call Flows              August 2002 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F3 100 Trying Proxy 1 -> User A Alice

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 ACK Proxy 1 -> NGW 1

Johnston et al            Expires - May 2002                 [Page 25] 
                         SIP PSTN Call Flows            November 2002

   ACK sip:+19725552222@ngw1.atlanta.com;user=phone sip:ngw1@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com>;user=phone>
                         SIP PSTN Call Flows              August 2002
    ;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F7 IAM NGW 2 -> User B Bob

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

   F8 ACM User B Bob -> NGW 2

Johnston et al            Expires - May 2002                 [Page 26] 
                         SIP PSTN Call Flows            November 2002

   ACM

   F9 183 Session Progress NGW 2 -> Proxy 1

   SIP/2.0 183 Session Progress
                         SIP PSTN Call Flows              August 2002
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone> <sip:ngw2@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.102 ngw2.atlanta.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 -> User A Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone> <sip:ngw2@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0

Johnston et al            Expires - May 2002                 [Page 27] 
                         SIP PSTN Call Flows            November 2002

   o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.102 ngw2.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
                         SIP PSTN Call Flows              August 2002

   F11 ANM User B Bob -> NGW 2

   ANM

   F12 200 OK NGW 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone> <sip:ngw2@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.102 ngw2.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F13 200 OK Proxy 1 -> User A Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>

Johnston et al            Expires - May 2002                 [Page 28] 
                         SIP PSTN Call Flows            November 2002

    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone> <sip:ngw2@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141
                         SIP PSTN Call Flows              August 2002 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.102 ngw2.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F14 ACK A Alice -> Proxy 1

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

   F15 ACK Proxy 1 -> NGW 2

   ACK sip:+19725552222@ngw2.atlanta.com;user=phone sip:ngw2@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

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

Johnston et al            Expires - May 2002                 [Page 29] 
                         SIP PSTN Call Flows            November 2002

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

   F16 BYE A Alice -> Proxy 1

   BYE sip:+19725552222@ngw2.atlanta.com;user=phone sip:ngw2@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
                         SIP PSTN Call Flows              August 2002
   Max-Forwards: 70
   Route: <ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F17 BYE Proxy 1 -> NGW 2

   BYE sip:+19725552222@ngw2.atlanta.com;user=phone sip:ngw2@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

Johnston et al            Expires - May 2002                 [Page 30] 
                         SIP PSTN Call Flows            November 2002

   F19 200 OK Proxy 1 -> User A Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
                         SIP PSTN Call Flows              August 2002
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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 - May 2002                 [Page 31] 
                         SIP PSTN Call Flows              August            November 2002

2.4    Unsuccessful SIP to PSTN call: Treatment from PSTN

   User A

   Alice            Proxy 1           NGW 1            User B            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    |                |                |
     |--------------->|                |                |
     |                |                |                |

   User A

   Alice calls User B 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.  User A  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 User A.) Alice.)

   Message Details

Johnston et al            Expires - May 2002                 [Page 32] 
                         SIP PSTN Call Flows              August            November 2002

   F1 INVITE A Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Proxy-Authorization: Digest username="UserA", username="alice",
    realm="atlanta.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40",
    opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
    response="e178fbe430e6680a1690261af8831f40"
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1

Johnston et al            Expires - May 2002                 [Page 33] 
                         SIP PSTN Call Flows              August            November 2002

   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 IAM NGW 1 -> User B Bob

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

   F6 ACM User B Bob -> NGW 1

   ACM

Johnston et al            Expires - May 2002                 [Page 34] 
                         SIP PSTN Call Flows              August            November 2002

   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F8 183 Session Progress Proxy 1 -> User A Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0

Johnston et al            Expires - May 2002                 [Page 35] 
                         SIP PSTN Call Flows              August            November 2002

   a=rtpmap:0 PCMU/8000

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

   F9 CANCEL A Alice -> Proxy 1

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

   F10 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 CANCEL
   Content-Length: 0

   F11 CANCEL Proxy 1 -> NGW 1

   CANCEL sip:+19725552222@ngw1.atlanta.com;user=phone sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111

Johnston et al            Expires - May 2002                 [Page 36] 
                         SIP PSTN Call Flows              August            November 2002

   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F16 ACK Proxy 1 -> NGW 1

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

Johnston et al            Expires - May 2002                 [Page 37] 
                         SIP PSTN Call Flows              August            November 2002

   F17 487 Request Terminated Proxy 1 -> A

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F18 ACK A Alice -> Proxy 1

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

Johnston et al            Expires - May 2002                 [Page 38] 
                         SIP PSTN Call Flows              August            November 2002

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

   User A

   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    |                |                |
     |--------------->|                |                |
     |                |                |                |

   User A

   Alice calls PSTN User B 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 User A. Alice.  For more
   details of ISUP cause value to SIP responses response mapping refer to [4].

   Message Details

   F1 INVITE A Alice -> Proxy 1

   INVITE sip:+44-1234@ss1.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+44-1234@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Proxy-Authorization: Digest username="UserA", username="alice",
    realm="atlanta.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",

Johnston et al            Expires - May 2002                 [Page 39] 
                         SIP PSTN Call Flows              August            November 2002

    opaque="", uri="sip:+44-1234@ss1.atlanta.com;user=phone",
    response="a451358d46b55512863efe1dccaa2f42"
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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 SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+44-1234@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.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.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+44-1234@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 147 150

Johnston et al            Expires - May 2002                 [Page 40] 
                         SIP PSTN Call Flows              August            November 2002

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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 SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+44-1234@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 IAM NGW 1 -> User B Bob

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

   F6 REL User B Bob -> NGW 1

   REL
   CauseValue=1 Unallocated number

   F7 RLC NGW 1 -> User B 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/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1

Johnston et al            Expires - May 2002                 [Page 41] 
                         SIP PSTN Call Flows              August            November 2002

    ;received=192.168.255.111

    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+44-1234@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Error-Info: <sip:not-found-ann@ann.atlanta.com>
   Content-Length: 0

   F9 ACK Proxy 1 -> NGW 1

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

   F10 404 Not Found Proxy 1 -> User A Alice

   SIP/2.0 404 Not Found
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+44-1234@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Error-Info: <sip:not-found-ann@ann.atlanta.com>
   Content-Length: 0

   F11 ACK User A Alice -> Proxy 1

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

Johnston et al            Expires - May 2002                 [Page 42] 
                         SIP PSTN Call Flows              August            November 2002

   Content-Length: 0

Johnston et al            Expires - May 2002                 [Page 43] 
                         SIP PSTN Call Flows              August            November 2002

2.6    Unsuccessful SIP to PSTN: ANM Timeout

   User A

   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    |                |                |
     |--------------->|                |                |

   User A

   Alice calls User B 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 User A Alice in the SIP network.

   Message Details

   F1 INVITE A Alice -> Proxy 1

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

Johnston et al            Expires - May 2002                 [Page 44] 
                         SIP PSTN Call Flows              August            November 2002

   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Proxy-Authorization: Digest username="UserA", username="alice",
    realm="atlanta.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40",
    opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
    response="579cb9db184cdc25bf816f37cbc03c7d"
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F3 INVITE Proxy 1 -> NGW 1

   INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE

Johnston et al            Expires - May 2002                 [Page 45] 
                         SIP PSTN Call Flows              August            November 2002

   Contact: <sip:UserA@192.168.100.101> <sip:alice@client.atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 147 150

   v=0
   o=UserA
   o=alice 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101 client.atlanta.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 SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 IAM NGW 1 -> User B Bob

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

   F6 ACM User B Bob -> NGW 1

   ACM

   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>

Johnston et al            Expires - May 2002                 [Page 46] 
                         SIP PSTN Call Flows              August            November 2002

    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F8 183 Session Progress Proxy 1 -> User A Alice

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 ngw1.atlanta.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 -> User B Bob

   REL

Johnston et al            Expires - May 2002                 [Page 47] 
                         SIP PSTN Call Flows              August            November 2002

   CauseCode=18 No user responding

   F10 RLC User B Bob -> NGW 1

   RLC

   F11 480 Temporarily Unavailable NGW 1 -> Proxy 1

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Error-Info: <sip:temp-unavail-ann@ann.atlanta.com>
   Content-Length: 0

   F12 ACK Proxy 1 -> NGW 1

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

   F13 480 Temporarily Unavailable F13 Proxy 1 -> User A Alice

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP SIP/2.0/TCP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
    ;received=192.0.2.101
   From: BigGuy Alice <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy Bob <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com

Johnston et al            Expires - May 2002                 [Page 48] 
                         SIP PSTN Call Flows              August            November 2002

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

   F14 ACK User A Alice -> Proxy 1

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

Johnston et al            Expires - May 2002                 [Page 49] 
                         SIP PSTN Call Flows              August            November 2002

3.   PSTN to SIP Dialing

   In these scenarios, User A Alice is placing calls from the PSTN to User B Bob
   in a SIP network.  User A's  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 atlanta.com proxy server has
   access to the biloxi.com location service.

                         SIP PSTN Call Flows              August

Johnston et al            Expires - May 2002                 [Page 50] 
                         SIP PSTN Call Flows              August            November 2002

3.1    Successful PSTN to SIP call

   Switch A          NGW 1          Proxy 1           User B           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, User A Alice from the PSTN calls User B Bob through a Network
   Gateway NGW1 and Proxy Server Proxy 1.  When User B Bob answers the call
   the media path is setup end-to-end. The call terminates when User A Alice
   hangs up the call, with User A's Alice's telephone switch sending an ISUP
   RELease message which is mapped to a BYE by NGW 1.

   Message Details

   F1 IAM User A 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 - May 2002                 [Page 51] 
                         SIP PSTN Call Flows              August            November 2002

   F2 INVITE A Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 User A.*/ Alice.*/

   F3 INVITE Proxy 1 -> User B Bob

   INVITE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0

Johnston et al            Expires - May 2002                 [Page 52] 
                         SIP PSTN Call Flows              August            November 2002

   a=rtpmap:0 PCMU/8000

   F4 100 Trying User B Bob -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 180 Ringing User B Bob -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   Content-Length: 0

   F6 180 Ringing Proxy 1 -> NGW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   Content-Length: 0

   F7 ACM NGW 1 -> User A Alice

Johnston et al            Expires - May 2002                 [Page 53] 
                         SIP PSTN Call Flows              August            November 2002

   ACM

   F8 200 OK User B Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 145 146

   v=0
   o=UserB
   o=bob 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201 client.biloxi.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.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   Content-Type: application/sdp
   Content-Length: 145 146

   v=0
   o=UserB
   o=bob 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201 client.biloxi.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

Johnston et al            Expires - May 2002                 [Page 54] 
                         SIP PSTN Call Flows              August            November 2002

   F10 ACK NGW 1 -> Proxy 1

   ACK sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F11 ACK Proxy 1 -> User B Bob

   ACK sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F12 ANM User B Bob -> NGW 1

   ANM

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

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

   F13 REL User A Alice -> NGW 1

   REL
   CauseCode=16 Normal

   F14 RLC NGW 1 -> User A Alice

   RLC

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                         SIP PSTN Call Flows              August            November 2002

   F15 BYE NGW 1-> Proxy 1

   BYE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F16 BYE Proxy 1 -> User B Bob

   BYE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F17 200 OK User B Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.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.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com

Johnston et al            Expires - May 2002                 [Page 56] 
                         SIP PSTN Call Flows              August            November 2002

   CSeq: 2 BYE
   Content-Length: 0
                         SIP PSTN Call

Johnston et al            Expires - May 2002                 [Page 57] 
                         SIP PSTN Call Flows              August            November 2002

3.2    Successful PSTN to SIP call, Fast Answer

   Switch A           NGW 1          Proxy 1           User B           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 5.1.1 3.1 except that User B 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 User A 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@ngw1.atlanta.com;user=phone sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2

Johnston et al            Expires - May 2002                 [Page 58] 
                         SIP PSTN Call Flows              August            November 2002

   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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.  User B  Bob  prepares to receive data on port 3456 from User A.*/ Alice.*/

   F3 INVITE Proxy 1 -> User B Bob

   INVITE UserB@biloxi.com bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 - May 2002                 [Page 59] 
                         SIP PSTN Call Flows              August            November 2002

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 200 OK User B Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 145 146

   v=0
   o=UserB
   o=bob 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201 client.biloxi.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/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 145 147

Johnston et al            Expires - May 2002                 [Page 60] 
                         SIP PSTN Call Flows              August            November 2002

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.200.201 client.biloxi.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F7 ACK NGW 1 -> Proxy 1

   ACK UserB@192.168.200.201 bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F8 ACK Proxy 1 -> User B Bob

   ACK UserB@192.168.200.201 bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=130.131.132.14
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F9 ANM User B Bob -> NGW 1

   ANM

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

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

   F10 REL ser A Alice -> NGW 1

   REL
   CauseCode=16 Normal

Johnston et al            Expires - May 2002                 [Page 61] 
                         SIP PSTN Call Flows              August            November 2002

   F11 RLC NGW 1 -> User A Alice

   RLC

   F12 BYE NGW 1 -> Proxy 1

   BYE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F13 BYE Proxy 1 -> User B Bob

   BYE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F14 200 OK User B Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F15 200 OK Proxy 1 -> NGW 1

Johnston et al            Expires - May 2002                 [Page 62] 
                         SIP PSTN Call Flows              August            November 2002

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

Johnston et al            Expires - May 2002                 [Page 63] 
                         SIP PSTN Call Flows              August            November 2002

3.3    Successful PBX to SIP call

   PBX A            GW 1           Proxy 1           User B           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, User A Alice dials from PBX A to User B 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 User B, Bob, GW 1 generates ringing
   tone for User A.

   User B Alice.

   Bob answers the call by sending a 200 OK.  The call terminates
   when User A Alice hangs up, causing GW1 to send a BYE.

Johnston et al            Expires - May 2002                 [Page 64] 
                         SIP PSTN Call Flows              August            November 2002

   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.atlanta.com.

   Message Details

   PBX A Alice -> GW 1

   Seizure

   GW 1 -> PBX A

   Wink

   F1 MF Digits PBX A Alice -> GW 1

   KP 1 972 555 2222 ST

   F2 INVITE GW 1 -> Proxy 1

   INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.atlanta.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.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 User B. Bob. */

Johnston et al            Expires - May 2002                 [Page 65] 
                         SIP PSTN Call Flows              August            November 2002

   F3 INVITE Proxy 1 -> User B Bob

   INVITE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.atlanta.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.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.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 180 Ringing User B Bob -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE

Johnston et al            Expires - May 2002                 [Page 66] 
                         SIP PSTN Call Flows              August            November 2002

   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   Content-Length: 0

   F6 180 Ringing Proxy 1 -> GW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   Content-Length: 0

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

   F7 200 OK User B Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 145 146

   v=0
   o=UserB
   o=bob 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201 client.biloxi.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 - May 2002                 [Page 67] 
                         SIP PSTN Call Flows              August            November 2002

   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   Content-Type: application/sdp
   Content-Length: 145 146

   v=0
   o=UserB
   o=bob 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201 client.biloxi.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F9 ACK GW 1 -> Proxy 1

   ACK sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F10 ACK Proxy 1 -> User B Bob

   ACK sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

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

Johnston et al            Expires - May 2002                 [Page 68] 
                         SIP PSTN Call Flows              August            November 2002

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

   F11 BYE GW 1 -> Proxy 1

   BYE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F12 BYE Proxy 1 -> User B Bob

   BYE sip:UserB@192.168.200.201 sip:bob@client.biloxi.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 2 BYE
   Content-Length: 0

   F13 200 OK User B Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.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.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159

Johnston et al            Expires - May 2002                 [Page 69] 
                         SIP PSTN Call Flows              August            November 2002

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

Johnston et al            Expires - May 2002                 [Page 70] 
                         SIP PSTN Call Flows              August            November 2002

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

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

   User A

   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 User A Alice -> GW 1

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

   F2 INVITE A Alice -> Proxy 1

   INVITE sip:+1972559999@ss1.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@gw1.atlanta.com;user=phone>;tag=076342s
   To: <sip:+1972559999@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@gw1.atlanta.com;user=phone> <sip:+13145551111@gw1.atlanta.com;user=phone;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.com

Johnston et al            Expires - May 2002                 [Page 71] 
                         SIP PSTN Call Flows              August            November 2002

   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/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:+13145551111@gw1.atlanta.com;user=phone>;tag=076342s
   To: <sip:+1972559999@ss1.atlanta.com;user=phone>;tag=6a34d410
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
   CSeq: 1 INVITE
   Error-Info: <sip:does-not-exist@ann.atlanta.com>
   Content-Length: 0

   F4 ACK GW 1 -> Proxy 1

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

   F5 REL GW 1 -> User A Alice

   REL
   CauseCode=1

   F6 RLC User A Alice -> GW 1

   RLC

Johnston et al            Expires - May 2002                 [Page 72] 
                         SIP PSTN Call Flows              August            November 2002

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

   Switch A          NGW 1           Proxy 1          User B          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, User A Alice calls User B Bob through Network Gateway NGW 1
   and Proxy 1.  The call is routed to User B Bob by Proxy 1.  The call is
   rejected by User B 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 User A's Alice's telephone switch.  In scenario 5.2.3, 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 User A Alice -> NGW 1

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

   F2 INVITE A Alice -> Proxy 1

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

Johnston et al            Expires - May 2002                 [Page 73] 
                         SIP PSTN Call Flows              August            November 2002

   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.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 User B. Bob. */

   F3 INVITE F3 Proxy 1 -> User B Bob

   INVITE UserB@biloxi.com bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.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 SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2

Johnston et al            Expires - May 2002                 [Page 74] 
                         SIP PSTN Call Flows              August            November 2002

    ;received=192.168.255.201

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

   F5 600 Busy Everywhere User B Bob -> Proxy 1

   SIP/2.0 600 Busy Everywhere
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F6 ACK Proxy 1 -> User B Bob

   ACK UserB@biloxi.com bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.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/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F8 ACK NGW 1 -> Proxy 1

   ACK UserB@biloxi.com bob@biloxi.com SIP/2.0

Johnston et al            Expires - May 2002                 [Page 75] 
                         SIP PSTN Call Flows              August            November 2002

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

   F9 REL NGW 1 -> User A Alice

   REL
   CauseCode=17 Busy

   F10 RLC User A Alice -> NGW 1

   RLC

Johnston et al            Expires - May 2002                 [Page 76] 
                         SIP PSTN Call Flows              August            November 2002

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

   Switch A          NGW 1           Proxy 1          User B          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, User A Alice calls User B Bob through Network Gateway NGW1
   and Proxy 1.  The call is routed to User B Bob by Proxy 1.  The call is
   rejected by the User B Bob client.  NGW 1 sets up a two way voice path to
   User A
   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 User A Alice -> NGW 1

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

Johnston et al            Expires - May 2002                 [Page 77] 
                         SIP PSTN Call Flows              August            November 2002

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

   F2 INVITE NGW1 -> Proxy 1

   INVITE sip:+19725552222@ngw1.atlanta.com;user=phone sip:+19725552222@ss1.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 User B. Bob. */

   F3 INVITE Proxy 1 -> User B Bob

   INVITE UserB@biloxi.com bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com

Johnston et al            Expires - May 2002                 [Page 78] 
                         SIP PSTN Call Flows              August            November 2002

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

   F4 100 Trying User B Bob -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 600 Busy Everywhere User B Bob -> Proxy 1

   SIP/2.0 600 Busy Everywhere
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F6 ACK Proxy 1 -> User B Bob

   ACK UserB@biloxi.com bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.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/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2

Johnston et al            Expires - May 2002                 [Page 79] 
                         SIP PSTN Call Flows              August            November 2002

    ;received=192.168.255.101

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

   F8 ACK NGW 1 -> Proxy 1

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

   F9 ACM NGW 1 -> User A Alice

   ACM

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

   /* Call Released after User A Alice hangs up. */

   F10 REL User A Alice -> NGW 1

   REL
   CauseCode=16

   F11 RLC NGW 1 -> User A Alice

   RLC

Johnston et al            Expires - May 2002                 [Page 80] 
                         SIP PSTN Call Flows              August            November 2002

3.7    Unsuccessful PSTN->SIP, ACM timeout

   Switch A          NGW 1           Proxy 1          User B          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    |                |
     |                |<---------------|                |

   User A

   Alice calls User B 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 User B.  User B Bob.  Bob never responds with 180 Ringing or any
   other response (it is reachable but unresponsive).  After the
   expiration of a timer, User A's Alice's network disconnects the call by
   sending a Release message REL.  The Gateway maps this to a CANCEL.
   Message Details

   F1 IAM User A Alice -> NGW 1

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

   F2 INVITE A Alice -> Proxy 1

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

Johnston et al            Expires - May 2002                 [Page 81] 
                         SIP PSTN Call Flows              August            November 2002

   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 User B. Bob. */

   F3 INVITE Proxy 1 -> User B Bob

   INVITE sip:UserB@biloxi.com sip:bob@biloxi.com  SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   c c=IN IP4 192.168.255.101 ngw1.atlanta.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.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com

Johnston et al            Expires - May 2002                 [Page 82] 
                         SIP PSTN Call Flows              August            November 2002

   CSeq: 1 INVITE
   Content-Length: 0

   F5 INVITE Proxy 1 -> User B Bob

   Same as Message F3

   F6 INVITE Proxy 1 -> User B Bob

   Same as Message F3

   F7 INVITE Proxy 1 -> User B Bob

   Same as Message F3

   F8 INVITE Proxy 1 -> User B Bob

   Same as Message F3

   F9 INVITE Proxy 1 -> User B Bob

   Same as Message F3

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

   F10 REL User A Alice -> NGW 1

   REL
   CauseCode=16 Normal

   F11 RLC NGW 1 -> User A Alice

   RLC

   F12 CANCEL NGW 1 -> Proxy 1

   CANCEL sip:+19725552222@ss11.atlanta.com;user=phone sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>

Johnston et al            Expires - May 2002                 [Page 83] 
                         SIP PSTN Call Flows              August            November 2002

   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.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.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 CANCEL
   Content-Length: 0

Johnston et al            Expires - May 2002                 [Page 84] 
                         SIP PSTN Call Flows              August            November 2002

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

   Switch A          NGW 1      Stateless Proxy 1     User B     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, User A Alice calls User B 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.  User B  Bob does not respond with 180 Ringing.  User A's  Alice's
   network disconnects the call with a release REL (CauseCode=102
   Timeout).

   Message Details

   F1 IAM User A 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.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com

Johnston et al            Expires - May 2002                 [Page 85] 
                         SIP PSTN Call Flows              August            November 2002

   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 User B. Bob. */

   F3 INVITE Proxy 1 -> User B Bob

   INVITE sip:UserB@biloxi.com sip:bob@biloxi.com  SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 -> User B Bob

   Same as Message F3

Johnston et al            Expires - May 2002                 [Page 86] 
                         SIP PSTN Call Flows              August            November 2002

   F6 INVITE NGW 1 -> Proxy 1

   Same as Message F2

   F7 INVITE Proxy 1 -> User B Bob

   Same as Message F3

   F8 INVITE NGW 1 -> Proxy 1

   Same as Message F2

   F9 INVITE Proxy 1 -> User B Bob

   Same as Message F3

   F10 INVITE NGW 1 -> Proxy 1

   Same as Message F2

   F11 INVITE Proxy 1 -> User B Bob

   Same as Message F3

   F12 INVITE NGW 1 -> Proxy 1

   Same as Message F2

   F13 INVITE Proxy 1 -> User B Bob

   Same as Message F3

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

   F14 REL User A Alice -> NGW 1

   REL
   CauseCode=102 Timeout

Johnston et al            Expires - May 2002                 [Page 87] 
                         SIP PSTN Call Flows              August            November 2002

   F15 RLC NGW 1 -> User A Alice

   RLC

Johnston et al            Expires - May 2002                 [Page 88] 
                         SIP PSTN Call Flows              August            November 2002

3.9    Unsuccessful PSTN->SIP, Caller Abandonment

   Switch A          NGW 1          Proxy 1           User B           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, User A Alice calls User B Bob through NGW 1 and Proxy 1.
   User B
   Bob does not respond with 200 OK.  NGW 1 plays ringing tone since
   the ACM indicates that interworking has been encountered.  User A  Alice
   disconnects the call with a Release message REL which is mapped by
   NGW 1 to a CANCEL.  Note that if User B 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 - May 2002                 [Page 89] 
                         SIP PSTN Call Flows              August            November 2002

   F1 IAM User A Alice -> NGW 1

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

   F2 INVITE A Alice -> Proxy 1

   INVITE sip:+19725552222@ss1.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.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 User B. Bob. */

   F3 INVITE Proxy 1 -> User B Bob

   INVITE sip:UserB@biloxi.com sip:bob@biloxi.com  SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sip:ngw1@atlanta.com;transport=tcp>
   Content-Type: application/sdp
   Content-Length: 141 142

Johnston et al            Expires - May 2002                 [Page 90] 
                         SIP PSTN Call Flows              August            November 2002

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   F4 100 Trying User B Bob -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 180 Ringing User B Bob -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com;transport=tcp>
   Content-Length: 0

   F6 180 Ringing Proxy 1 -> NGW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com

Johnston et al            Expires - May 2002                 [Page 91] 
                         SIP PSTN Call Flows              August            November 2002

   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201> <sip:bob@client.biloxi.com>
   Content-Length: 0

   F7 ACM NGW 1 -> User A Alice

   ACM

   /* User A Alice hangs up */

   F8 REL User A Alice -> NGW 1

   REL
   CauseCode=16 Normal

   F9 RLC NGW 1 -> User A Alice

   RLC

   F10 CANCEL NGW 1 -> Proxy 1

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

   F11 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 CANCEL
   Content-Length: 0

   F12 CANCEL Proxy 1 -> User B Bob

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                         SIP PSTN Call Flows              August            November 2002

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

   F13 200 OK User B Bob -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 CANCEL
   Content-Length: 0

   F14 487 Request Terminated User B Bob -> Proxy 1

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP SIP/2.0/TCP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F15 ACK Proxy 1 -> User B Bob

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

   F16 487 Request Terminated Proxy 1 -> NGW 1

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                         SIP PSTN Call Flows              August            November 2002

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
    ;received=192.0.2.103
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F17 ACK NGW 1 -> Proxy 1

   ACK sip:+19725552222@ss11.atlanta.com;user=phone sip:+19725552222@ss1.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP SIP/2.0/TCP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

Johnston et al            Expires - May 2002                 [Page 94] 
                         SIP PSTN Call Flows              August            November 2002

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.

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                         SIP PSTN Call Flows              August            November 2002

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, User A Alice in the PSTN calls User C Carol who is an extension
   on a PBX.  User A's  Alice's telephone switch signals via SS7 to the Network
   Gateway NGW 1, while User C's 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

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                         SIP PSTN Call Flows              August            November 2002

   User C

   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
   User A Alice can hear
   the ringing tone generated by PBX C.

   User C

   Carol answers the call and the media path is cut through in both
   directions.  User B  Bob hangs up terminating the call.

   Message Details

   F1 IAM Switch A 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 sip:+19185553333@ss1.atlanta.com;user=phone sips:+19185553333@ss1.atlanta.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2 SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKlueha2
   Max-Forwards: 70
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone> <sips:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sips:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 49172 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 sip:4443333@gw2.atlanta.com sips:4443333@gw2.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKwqwee65 SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKwqwee65

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                         SIP PSTN Call Flows              August            November 2002

    ;received=192.168.255.101

    ;received=192.0.2.103
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone> <sips:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone> <sips:ngw1@atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141 142

   v=0
   o=GW 2890844526 2890844526 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101 ngw1.atlanta.com
   t=0 0
   m=audio 49172 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 User A Alice through NGW 1. */

   F6 183 Session Progress GW 2 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:4443333@gw2.atlanta.com> <sips:4443333@gw2.atlanta.com>
   Content-Type: application/sdp

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                         SIP PSTN Call Flows              August            November 2002

   Content-Length: 149

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.202 gw2.atlanta.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/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101 SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:4443333@gw2.atlanta.com> <sips:4443333@gw2.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.202 gw2.atlanta.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

   /* User B Bob answers */

   F9 ANM Switch C -> GW 2

   ANM

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                         SIP PSTN Call Flows              August            November 2002

   F10 200 OK GW 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:4443333@gw2.atlanta.com> <sips:4443333@gw2.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.202 gw2.atlanta.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/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101 SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Record-Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:4443333@gw2.atlanta.com> <sips:4443333@gw2.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.202 gw2.atlanta.com
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

Johnston et al            Expires - May 2002                [Page 100] 
                         SIP PSTN Call Flows              August            November 2002

   F12 ANM NGW 1 -> Switch A

   ANM

   F13 ACK NGW 1 -> Proxy 1

   ACK sip:4443333@gw2.atlanta.com sips:4443333@gw2.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2 SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKlueha2
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F14 ACK Proxy 1 -> GW 2

   ACK sip:4443333@gw2.atlanta.com sips:4443333@gw2.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101 SIP/2.0/TLS ngw1.atlanta.com:5061;branch=z9hG4bKlueha2
    ;received=192.0.2.103
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

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

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

   F15 REL Switch C -> GW 2

   REL
   CauseCode=16 Normal

   F16 BYE GW 2 -> Proxy 1

   BYE sip:+13145551111@ngw1.atlanta.com;user=phone sips:ngw1@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6 SIP/2.0/TLS gw2.atlanta.com:5061;branch=z9hG4bKtexx6
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr> <sips:ss1.atlanta.com;lr>

Johnston et al            Expires - May 2002                [Page 101] 
                         SIP PSTN Call Flows              August            November 2002

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

   F17 RLC GW 2 -> Switch C

   RLC

   F18 BYE Proxy 1 -> NGW 1

   BYE sip:+13145551111@ngw1.atlanta.com;user=phone sips:ngw1@atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6
    ;received=192.168.255.202 SIP/2.0/TLS gw2.atlanta.com:5061;branch=z9hG4bKtexx6
    ;received=192.0.2.202
   Max-Forwards: 69
   From: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   To: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 4 BYE
   Content-Length: 0

   F19 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111 SIP/2.0/TLS ss1.atlanta.com:5061;branch=z9hG4bK2d4790.1
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6
    ;received=192.168.255.202 SIP/2.0/TLS gw2.atlanta.com:5061;branch=z9hG4bKtexx6
    ;received=192.0.2.202
   From: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   To: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 4 BYE
   Content-Length: 0

   F20 200 OK Proxy 1 -> GW 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6
    ;received=192.168.255.202 SIP/2.0/TLS gw2.atlanta.com:5061;branch=z9hG4bKtexx6
    ;received=192.0.2.202
   From: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159 <sips:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   To: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals <sips:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 4 BYE

Johnston et al            Expires - May 2002                [Page 102] 
                         SIP PSTN Call Flows              August            November 2002

   Content-Length: 0

   F21 REL Switch C -> GW 2

   REL
   CauseCode=16 Normal

   F22 RLC GW 2 -> Switch C

   RLC

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                         SIP PSTN Call Flows              August            November 2002

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 - May 2002                [Page 104] 
                         SIP PSTN Call Flows              August            November 2002

   PBX User A Alice calls PBX User C Carol via Gateway GW 1 and Proxy 1.  During the
   attempt to reach User C 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 User C Carol answering the call.  The call is terminated when
   User C
   Carol hangs up.

   Message Details

   PBX A Alice -> GW 1

   Seizure

   GW 1 -> PBX A

   Wink

   F1 MF Digits PBX A Alice -> GW 1

   KP 444 3333 ST

   F2 INVITE GW 1 -> Proxy 1

   INVITE sip:4443333@ss1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

Johnston et al            Expires - May 2002                [Page 105] 
                         SIP PSTN Call Flows              August            November 2002

   /* 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.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0

   F5 ACK Proxy 1 -> GW 2

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

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                         SIP PSTN Call Flows              August            November 2002

   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forward: 70
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F6 INVITE Proxy 1 -> GW 3

   INVITE sip:+19185553333@gw3.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:551313@gw1.atlanta.com>
   Content-Type: application/sdp
   Content-Length: 141

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201 gw1.atlanta.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 - May 2002                [Page 107] 
                         SIP PSTN Call Flows              August            November 2002

   F8 100 Trying GW 3 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19185553333@gw3.atlanta.com;user=phone>
   Content-Length: 0

   F12 180 Ringing Proxy 1 -> GW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>

Johnston et al            Expires - May 2002                [Page 108] 
                         SIP PSTN Call Flows              August            November 2002

   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19185553333@gw3.atlanta.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.atlanta.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19185553333@gw3.atlanta.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=GW 987654321 987654321 IN IP4 gw3.atlanta.com
   s=-
   c=IN IP4 192.168.255.203 gw3.atlanta.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.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 INVITE

Johnston et al            Expires - May 2002                [Page 109] 
                         SIP PSTN Call Flows              August            November 2002

   Contact: <sip:+19185553333@gw3.atlanta.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=GW 987654321 987654321 IN IP4 gw3.atlanta.com
   s=-
   c=IN IP4 192.168.255.203 gw3.atlanta.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.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F17 ACK Proxy 1 -> GW 3

   ACK sip:+19185553333@gw3.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
    ;received=192.0.2.201
   Max-Forwards: 69
   From: <sip:551313@gw1.atlanta.com>;tag=63412s
   To: <sip:4443333@ss1.atlanta.com>;tag=123456789
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

   F18 CONNect ACK GW 3 -> PBX C

   Protocol discriminator=Q.931
   Message type=CONN ACK

Johnston et al            Expires - May 2002                [Page 110] 
                         SIP PSTN Call Flows              August            November 2002

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

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

   F21 BYE Proxy 1 -> GW 1

   BYE sip:551313@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
   Via: SIP/2.0/UDP gw3.atlanta.com:5060;branch=z9hG4bKkdjuwq
    ;received=192.168.255.203
    ;received=192.0.2.203
   Max-Forwards: 69
   From: <sip:4443333@ss1.atlanta.com>;tag=123456789
   To: <sip:551313@gw1.atlanta.com>;tag=63412s
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.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

Johnston et al            Expires - May 2002                [Page 111] 
                         SIP PSTN Call Flows              August            November 2002

   F23 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.168.255.111
    ;received=192.0.2.111
   Via: SIP/2.0/UDP gw3.atlanta.com:5060;branch=z9hG4bKkdjuwq
    ;received=192.168.255.203
    ;received=192.0.2.203
   From: <sip:4443333@ss1.atlanta.com>;tag=123456789
   To: <sip:551313@gw1.atlanta.com>;tag=63412s
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.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.atlanta.com:5060;branch=z9hG4bKkdjuwq
    ;received=192.168.255.203
    ;received=192.0.2.203
   From: <sip:4443333@ss1.atlanta.com>;tag=123456789
   To: <sip:551313@gw1.atlanta.com>;tag=63412s
   Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
   CSeq: 1 BYE
   Content-Length: 0

   F25 RELease COMplete PBX C -> GW 3

   Protocol discriminator=Q.931
   Message type=REL COM

   PBX A Alice -> GW 1

   Seizure removal

Security Considerations

   Since this

   This document represents NON NORMATIVE 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.

Johnston et al            Expires - May 2002                [Page 112] 
                         SIP PSTN Call Flows            November 2002

   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 session
   establishment, the security considerations 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] apply.

References
   [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              August            November 2002

Normative References

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

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

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

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

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

   6

   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.

   7 Dierks, T.

   6 J. Rosenberg, H. Schulzrinne, and C. Allen, G. Camarillo, "The TLS Stream
      Control Transmission Protocol Version 1.0", RFC 2246,
      January 1999.

   8 S. Kent, R. Atkinson, "Security Architecture as a Transport for the Session
      Initiation Protocol," Internet
      Protocol", RFC 2401, November 1998.

   9 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.

Informative References

   10 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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                         SIP PSTN Call Flows            November 2002

Acknowledgments

   Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings,
   and Tom Taylor for their detailed comments during the final 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 Neil Deason for his additions to the
   Torture Test messages and Kundan Singh for performing parser
   validation of messages.

                         SIP PSTN Call Flows              August 2002

   The authors wish to thank the following individuals for their
   participation in the final 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

Johnston et al            Expires - May 2002                [Page 115] 
                         SIP PSTN Call Flows            November 2002

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

      EMail:  rsparks@dynamicsoft.com

      Chris Cunningham
      dynamicsoft, Inc.

                         SIP PSTN Call Flows              August 2002
      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

   Copyright Notice

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   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

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                         SIP PSTN Call Flows            November 2002

   This document and the information contained herein is provided on an
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Acknowledgement
                         SIP PSTN Call Flows              August 2002

   Funding for the RFC Editor function is currently provided by the
   Internet Society.

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