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Versions: 00 01 02 RFC 3666

   SIPPING Working Group                                    A. Johnston
   Internet Draft                                              WorldCom
   Document: draft-ietf-sipping-pstn-call-flows-00.txt       S. Donovan
   Expires: February 2003                                     R. Sparks
                                                          C. Cunningham
                                                            dynamicsoft
                                                             K. Summers
                                                                  Sonus
                                                            August 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 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, 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
   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
      3.2 Successful PSTN to SIP call, Fast Answer..................59
      3.3 Successful PBX to SIP call................................65
      3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL.....72
      3.5 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL......74
      3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones...78
      3.7 Unsuccessful PSTN->SIP, ACM timeout.......................82
      3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy......86
      3.9 Unsuccessful PSTN->SIP, Caller Abandonment................90
   4. PSTN to PSTN Dialing via SIP Network..........................96
      4.1 Successful ISUP PSTN to ISUP PSTN call....................97
      4.2 Successful FGB PBX to ISDN PBX call with overflow........105
   Security Considerations.........................................113
   References......................................................113
   Acknowledgments.................................................114
   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 network.  They represent an example
   minimum set of functionality for SIP to be used in IP Telephony
   applications.

   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 and
   additional RFCs are added that this document will reflect those


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   changes and represent the current state of a standard interoperable
   SIP IP Telephony implementation.

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

   Note that this document is informational, and is NOT NORMATIVE on any
   aspect of SIP or SIP/PSTN interworking.

   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 to illustrate 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 to ISUP mapping, refer to [4].

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

1.1     General Assumptions

   A number of architecture, network, and protocol assumptions underly
   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].

   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 UDP for transport.  Other transport schemes could
   also be used.

   Throughout this document the call flows show a network where the
   proxy servers authenticate users on behalf of gateways.  Gateways may
   also authenticate users directly. Both of these are reasonable usages
   of 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


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   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.

   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
   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. */

1.3     SIP Protocol Assumptions

   This document is informational only and is NOT NORMATIVE in any
   sense.

   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


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   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
   User Agent    LittleGuy      UserB@biloxi.com     192.168.200.201
   Proxy Server                 ss1.atlanta.com      192.168.255.111
   User Agent (Gateway)         gw1.atlanta.com      192.168.255.201
   User Agent (Gateway)         gw2.atlanta.com      192.168.255.202
   User Agent (Gateway)         gw3.atlanta.com      192.168.255.203
   User Agent (Gateway)         ngw1.atlanta.com     192.168.255.101
   User Agent (Gateway)         ngw2.atlanta.com     192.168.255.102


































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2.   SIP to PSTN Dialing


   In the following scenarios, User A (BigGuy sip:UserA@atlanta.com) is
   a SIP phone or other SIP-enabled device.  User B is reachable via the
   PSTN at global telephone number +19725552222. User A places a call
   to User B through a Proxy Server Proxy 1 and a Network Gateway.  In
   other scenarios, User A places calls to User C, 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).  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 in
   the PSTN.

   In these scenarios, User A is a SIP phone or other SIP-enabled
   device.  User A places a call to User B in the PSTN or User C 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 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          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 dials the globalized E.164 number +19725552222 to reach
   User B.  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 could use either their SIP address (sip:UserA@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 (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 answers the call then User A disconnects the
   call.  Signaling between NGW 1 and User B's telephone switch is ANSI
   ISUP.  For the details of SIP to ISUP mapping, refer to [4].


   Message Details


   F1 INVITE A -> 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 <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101>
   Proxy-Authorization: Digest username="UserA", realm="atlanta.com",
    nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
    uri="sip:+19725552222@ss1.atlanta.com;user=phone",
    response="ccdca50cb091d587421457305d097458c"
   Content-Type: application/sdp
   Content-Length: 147

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


   F2 100 Trying Proxy 1 -> User A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE


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

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101
   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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 IAM NGW 1 -> User B



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


   F6 ACM User B -> NGW 1

   ACM


   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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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

   F8 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159


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   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 141

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


   F9 ANM User B -> NGW 1

   ANM


   F10 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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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


   F11 200 OK Proxy 1 -> User A



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   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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


   F12 ACK A -> Proxy 1

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


   F13 ACK Proxy 1 -> NGW 1

   ACK 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
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159


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   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 ACK
   Content-Length: 0


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

   F14 BYE A -> Proxy 1

   BYE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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/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
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 BYE
   Content-Length: 0


   F16 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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159


Johnston et al         Expires - February 2002              [Page 13]

                         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 client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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 - February 2002              [Page 14]

                         SIP PSTN Call Flows              August 2002


2.2    Successful SIP to ISDN PBX call

   User A          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 is a SIP device while User C is connected via a
   Gateway (GW 1) to a PBX.  The PBX connection is via a ISDN trunk
   group.  User A dials User C'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 - February 2002              [Page 15]

                         SIP PSTN Call Flows              August 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 identified by its extension
   (444-3333) in the Request-URI sent to GW 1.


   Message Details


   F1 INVITE A -> Proxy 1

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

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


   F2 100 Trying Proxy 1 -> User A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com


Johnston et al         Expires - February 2002              [Page 16]

                         SIP PSTN Call Flows              August 2002


   CSeq: 2 INVITE
   Content-Length: 0


   F3 INVITE Proxy 1 -> GW 1

   INVITE sip:4443333@gw1.atlanta.com 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
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@192.168.100.101>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101
   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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Content-Length: 0


   F5 SETUP GW 1 -> User C

   Protocol discriminator=Q.931
   Message type=SETUP
   Bearer capability: Information transfer capability=0 (Speech) or 16
   (3.1 kHz audio)


Johnston et al         Expires - February 2002              [Page 17]

                         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 -> GW 1

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


   F7 PROGress User C -> 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
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:4443333@gw1.atlanta.com>
   Content-Length: 0


   F9 180 Ringing Proxy 1 -> User A

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl


Johnston et al         Expires - February 2002              [Page 18]

                         SIP PSTN Call Flows              August 2002


   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip:4443333@gw1.atlanta.com>
   Content-Length: 0


   F10 CONNect User C -> GW 1

   Protocol discriminator=Q.931
   Message type=CONN


   F11 CONNect ACK GW 1 -> User C

   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
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip: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
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F13 200 OK Proxy 1 -> User A



Johnston et al         Expires - February 2002              [Page 19]

                         SIP PSTN Call Flows              August 2002


   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 INVITE
   Contact: <sip: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
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 ACK A -> Proxy 1

   ACK sip:4443333@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+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 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
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com


Johnston et al         Expires - February 2002              [Page 20]

                         SIP PSTN Call Flows              August 2002


   CSeq: 2 ACK
   Content-Length: 0


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

   F16 BYE A -> Proxy 1

   BYE sip:4443333@gw1.atlanta.com SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+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 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
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com


Johnston et al         Expires - February 2002              [Page 21]

                         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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: OtherGuy <sip:+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

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


   F21 RELease User C -> GW 1

   Protocol discriminator=Q.931
   Message type=REL


   F22 RELease COMplete GW 1 -> User C

   Protocol discriminator=Q.931
   Message type=REL COM
















Johnston et al         Expires - February 2002              [Page 22]

                         SIP PSTN Call Flows              August 2002


2.3    Successful SIP to ISUP PSTN call with overflow

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

   Message Details



Johnston et al         Expires - February 2002              [Page 23]

                         SIP PSTN Call Flows              August 2002



   F1 INVITE A -> 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 <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101>
   Proxy-Authorization: Digest username="UserA",
    realm="atlanta.com", nonce="b59311c3ba05b401cf80b2a2c5ac51b0",
    opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
    response="ba6ab44923fa2614b28e3e3957789ab0"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101
   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
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101>
   Content-Type: application/sdp
   Content-Length: 147



Johnston et al         Expires - February 2002              [Page 24]

                         SIP PSTN Call Flows              August 2002


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


   F3 100 Trying Proxy 1 -> User A

   SIP/2.0 100 Trying
   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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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

   ACK sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com>;user=phone>


Johnston et al         Expires - February 2002              [Page 25]

                         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
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101>
   Content-Type: application/sdp
   Content-Length: 147

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


   F7 IAM NGW 2 -> User B

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


   F8 ACM User B -> NGW 2

   ACM


   F9 183 Session Progress NGW 2 -> Proxy 1

   SIP/2.0 183 Session Progress


Johnston et al         Expires - February 2002              [Page 26]

                         SIP PSTN Call Flows              August 2002


   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
    ;received=192.168.255.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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


Johnston et al         Expires - February 2002              [Page 27]

                         SIP PSTN Call Flows              August 2002




   F11 ANM User B -> 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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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


   F13 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141


Johnston et al         Expires - February 2002              [Page 28]

                         SIP PSTN Call Flows              August 2002



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


   F14 ACK A -> Proxy 1

   ACK sip:+19725552222@ngw2.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   Route: <ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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/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
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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) */

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

   F16 BYE A -> Proxy 1

   BYE sip:+19725552222@ngw2.atlanta.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9


Johnston et al         Expires - February 2002              [Page 29]

                         SIP PSTN Call Flows              August 2002


   Max-Forwards: 70
   Route: <ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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/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
   Max-Forwards: 69
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 2 BYE
   Content-Length: 0


   F19 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>


Johnston et al         Expires - February 2002              [Page 30]

                         SIP PSTN Call Flows              August 2002


    ;tag=9fxced76sl
   To: LittleGuy <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 - February 2002              [Page 31]

                         SIP PSTN Call Flows              August 2002


2.4    Unsuccessful SIP to PSTN call: Treatment from PSTN

   User A          Proxy 1           NGW 1            User B
     |                |                |                |
     |   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 calls User B 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 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.)


   Message Details




Johnston et al         Expires - February 2002              [Page 32]

                         SIP PSTN Call Flows              August 2002


   F1 INVITE A -> 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 <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@192.168.100.101>
   Proxy-Authorization: Digest username="UserA",
    realm="atlanta.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40",
    opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
    response="e178fbe430e6680a1690261af8831f40"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101
   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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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 - February 2002              [Page 33]

                         SIP PSTN Call Flows              August 2002


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

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


   F5 IAM NGW 1 -> User B

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


   F6 ACM User B -> NGW 1

   ACM




Johnston et al         Expires - February 2002              [Page 34]

                         SIP PSTN Call Flows              August 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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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


   F8 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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


Johnston et al         Expires - February 2002              [Page 35]

                         SIP PSTN Call Flows              August 2002


   a=rtpmap:0 PCMU/8000


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

   F9 CANCEL A -> 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 <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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


Johnston et al         Expires - February 2002              [Page 36]

                         SIP PSTN Call Flows              August 2002


   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 ACK
   Content-Length: 0




Johnston et al         Expires - February 2002              [Page 37]

                         SIP PSTN Call Flows              August 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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE
   Content-Length: 0


   F18 ACK A -> 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 <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
    ;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 ACK
   Content-Length: 0

























Johnston et al         Expires - February 2002              [Page 38]

                         SIP PSTN Call Flows              August 2002


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

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


   Message Details


   F1 INVITE A -> Proxy 1

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


Johnston et al         Expires - February 2002              [Page 39]

                         SIP PSTN Call Flows              August 2002


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

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



Johnston et al         Expires - February 2002              [Page 40]

                         SIP PSTN Call Flows              August 2002


   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101
   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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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

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


   F6 REL User B -> NGW 1

   REL
   CauseValue=1 Unallocated number


   F7 RLC NGW 1 -> User B

   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 ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1


Johnston et al         Expires - February 2002              [Page 41]

                         SIP PSTN Call Flows              August 2002


    ;received=192.168.255.111
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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 ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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

   SIP/2.0 404 Not Found
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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 -> Proxy 1

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


Johnston et al         Expires - February 2002              [Page 42]

                         SIP PSTN Call Flows              August 2002


   Content-Length: 0


















































Johnston et al         Expires - February 2002              [Page 43]

                         SIP PSTN Call Flows              August 2002


2.6    Unsuccessful SIP to PSTN: ANM Timeout

   User A          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 calls User B 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 in the SIP network.


   Message Details


   F1 INVITE A -> 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 <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com


Johnston et al         Expires - February 2002              [Page 44]

                         SIP PSTN Call Flows              August 2002


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

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
   s=-
   c=IN IP4 192.168.100.101
   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 client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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 ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
   CSeq: 1 INVITE


Johnston et al         Expires - February 2002              [Page 45]

                         SIP PSTN Call Flows              August 2002


   Contact: <sip:UserA@192.168.100.101>
   Content-Type: application/sdp
   Content-Length: 147

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


   F5 IAM NGW 1 -> User B

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


   F6 ACM User B -> NGW 1

   ACM


   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
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>


Johnston et al         Expires - February 2002              [Page 46]

                         SIP PSTN Call Flows              August 2002


    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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


   F8 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
    ;received=192.168.100.101
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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>
   Content-Type: application/sdp
   Content-Length: 141

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

   REL


Johnston et al         Expires - February 2002              [Page 47]

                         SIP PSTN Call Flows              August 2002


   CauseCode=18 No user responding


   F10 RLC User B -> NGW 1

   RLC


   F11 480 Temporarily Unavailable NGW 1 -> Proxy 1

   SIP/2.0 480 Temporarily Unavailable
   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
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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/2.0
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
   Max-Forwards: 70
   From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
    ;tag=9fxced76sl
   To: LittleGuy <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

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


Johnston et al         Expires - February 2002              [Page 48]

                         SIP PSTN Call Flows              August 2002


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


   F14 ACK User A -> Proxy 1

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


































Johnston et al         Expires - February 2002              [Page 49]

                         SIP PSTN Call Flows              August 2002


3.   PSTN to SIP Dialing


   In these scenarios, User A is placing calls from the PSTN to User B
   in a SIP network.  User A'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.
































Johnston et al         Expires - February 2002              [Page 50]

                         SIP PSTN Call Flows              August 2002





















































Johnston et al         Expires - February 2002              [Page 51]

                         SIP PSTN Call Flows              August 2002


3.1    Successful PSTN to SIP call

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

   Message Details


   F1 IAM User A -> 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 - February 2002              [Page 52]

                         SIP PSTN Call Flows              August 2002



   F2 INVITE A -> 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>
   Content-Type: application/sdp
   Content-Length: 141

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

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@192.168.200.201 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
   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>
   Content-Type: application/sdp
   Content-Length: 141

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


Johnston et al         Expires - February 2002              [Page 53]

                         SIP PSTN Call Flows              August 2002


   a=rtpmap:0 PCMU/8000


   F4 100 Trying User B -> Proxy 1

   SIP/2.0 100 Trying
   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
   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 -> 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 ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
   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>
   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
   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>
   Content-Length: 0


   F7 ACM NGW 1 -> User A



Johnston et al         Expires - February 2002              [Page 54]

                         SIP PSTN Call Flows              August 2002


   ACM


   F8 200 OK User B -> 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
   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>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 145

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201
   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
   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>
   Content-Type: application/sdp
   Content-Length: 145

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


Johnston et al         Expires - February 2002              [Page 55]

                         SIP PSTN Call Flows              August 2002




   F10 ACK NGW 1 -> Proxy 1

   ACK sip:UserB@192.168.200.201 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

   ACK sip:UserB@192.168.200.201 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
   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 -> NGW 1

   ANM


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

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

   F13 REL User A -> NGW 1

   REL
   CauseCode=16 Normal


   F14 RLC NGW 1 -> User A

   RLC




Johnston et al         Expires - February 2002              [Page 56]

                         SIP PSTN Call Flows              August 2002


   F15 BYE NGW 1-> Proxy 1

   BYE sip:UserB@192.168.200.201 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

   BYE sip:UserB@192.168.200.201 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
   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 -> 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
   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
   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 - February 2002              [Page 57]

                         SIP PSTN Call Flows              August 2002


   CSeq: 2 BYE
   Content-Length: 0

















































Johnston et al         Expires - February 2002              [Page 58]

                         SIP PSTN Call Flows              August 2002


3.2    Successful PSTN to SIP call, Fast Answer

   Switch A           NGW 1          Proxy 1           User B
     |                |                |                |
     |     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 except that User B
   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 -> 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/2.0
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2


Johnston et al         Expires - February 2002              [Page 59]

                         SIP PSTN Call Flows              August 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>
   Content-Type: application/sdp
   Content-Length: 141

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

   F3 INVITE Proxy 1 -> User B

   INVITE UserB@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
   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>
   Content-Type: application/sdp
   Content-Length: 141

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101
   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 - February 2002              [Page 60]

                         SIP PSTN Call Flows              August 2002


   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
   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 -> 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
   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>
   Content-Type: application/sdp
   Content-Length: 145

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201
   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 ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
   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>
   Content-Type: application/sdp
   Content-Length: 145



Johnston et al         Expires - February 2002              [Page 61]

                         SIP PSTN Call Flows              August 2002


   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.200.201
   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 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


   F8 ACK Proxy 1 -> User B

   ACK UserB@192.168.200.201 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=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 -> NGW 1

   ANM


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

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

   F10 REL ser A -> NGW 1

   REL
   CauseCode=16 Normal


Johnston et al         Expires - February 2002              [Page 62]

                         SIP PSTN Call Flows              August 2002




   F11 RLC NGW 1 -> User A

   RLC


   F12 BYE NGW 1 -> Proxy 1

   BYE sip:UserB@192.168.200.201 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


   F13 BYE Proxy 1 -> User B

   BYE sip:UserB@192.168.200.201 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
   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 -> 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
   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 - February 2002              [Page 63]

                         SIP PSTN Call Flows              August 2002



   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
   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 - February 2002              [Page 64]

                         SIP PSTN Call Flows              August 2002


3.3    Successful PBX to SIP call

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

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



Johnston et al         Expires - February 2002              [Page 65]

                         SIP PSTN Call Flows              August 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 -> GW 1

   Seizure


   GW 1 -> PBX A

   Wink


   F1 MF Digits PBX A -> 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

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



Johnston et al         Expires - February 2002              [Page 66]

                         SIP PSTN Call Flows              August 2002


   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@192.168.200.201 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
   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

   v=0
   o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.201
   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
   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 -> 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 ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
   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 - February 2002              [Page 67]

                         SIP PSTN Call Flows              August 2002


   Contact: <sip:UserB@192.168.200.201>
   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
   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>
   Content-Length: 0


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

   F7 200 OK User B -> 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 gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
   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>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 145

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201
   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 - February 2002              [Page 68]

                         SIP PSTN Call Flows              August 2002


   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
   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>
   Content-Type: application/sdp
   Content-Length: 145

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
   s=-
   c=IN IP4 192.168.200.201
   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/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

   ACK sip:UserB@192.168.200.201 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
   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 - February 2002              [Page 69]

                         SIP PSTN Call Flows              August 2002


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

   F11 BYE GW 1 -> Proxy 1

   BYE sip:UserB@192.168.200.201 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

   BYE sip:UserB@192.168.200.201 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
   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 -> 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 gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
   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
   From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
   To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159


Johnston et al         Expires - February 2002              [Page 70]

                         SIP PSTN Call Flows              August 2002


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
















































Johnston et al         Expires - February 2002              [Page 71]

                         SIP PSTN Call Flows              August 2002


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

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

   User A 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 -> GW 1

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


   F2 INVITE A -> Proxy 1

   INVITE sip:+1972559999@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@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>
   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


Johnston et al         Expires - February 2002              [Page 72]

                         SIP PSTN Call Flows              August 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 ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
   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 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

   REL
   CauseCode=1


   F6 RLC User A -> GW 1

   RLC









Johnston et al         Expires - February 2002              [Page 73]

                         SIP PSTN Call Flows              August 2002


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

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

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


   F2 INVITE A -> Proxy 1

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


Johnston et al         Expires - February 2002              [Page 74]

                         SIP PSTN Call Flows              August 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>
   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
   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. */

   F3 INVITE F3 Proxy 1 -> User B

   INVITE UserB@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
   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>
   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
   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


Johnston et al         Expires - February 2002              [Page 75]

                         SIP PSTN Call Flows              August 2002


    ;received=192.168.255.201
   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 -> Proxy 1

   SIP/2.0 600 Busy Everywhere
   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=192.168.255.201
   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

   ACK UserB@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP 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 ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.201
   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 SIP/2.0


Johnston et al         Expires - February 2002              [Page 76]

                         SIP PSTN Call Flows              August 2002


   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>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0


   F9 REL NGW 1 -> User A

   REL
   CauseCode=17 Busy


   F10 RLC User A -> NGW 1

   RLC

































Johnston et al         Expires - February 2002              [Page 77]

                         SIP PSTN Call Flows              August 2002


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

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

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


Johnston et al         Expires - February 2002              [Page 78]

                         SIP PSTN Call Flows              August 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/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>
   Content-Type: application/sdp
   Content-Length: 141

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

   F3 INVITE Proxy 1 -> User B

   INVITE UserB@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
   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>
   Content-Type: application/sdp
   Content-Length: 141

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


Johnston et al         Expires - February 2002              [Page 79]

                         SIP PSTN Call Flows              August 2002


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


   F4 100 Trying User B -> Proxy 1

   SIP/2.0 100 Trying
   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
   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 -> Proxy 1

   SIP/2.0 600 Busy Everywhere
   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
   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

   ACK UserB@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP 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 ngw1.atlanta.com:5060;branch=z9hG4bKlueha2


Johnston et al         Expires - February 2002              [Page 80]

                         SIP PSTN Call Flows              August 2002


    ;received=192.168.255.101
   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/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>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0


   F9 ACM NGW 1 -> User A

   ACM


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

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

   F10 REL User A -> NGW 1

   REL
   CauseCode=16


   F11 RLC NGW 1 -> User A

   RLC













Johnston et al         Expires - February 2002              [Page 81]

                         SIP PSTN Call Flows              August 2002


3.7    Unsuccessful PSTN->SIP, ACM timeout

   Switch A          NGW 1           Proxy 1          User B
     |                |                |                |
     |     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 calls User B 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 never responds with 180 Ringing or any
   other response (it is reachable but unresponsive).  After the
   expiration of a timer, User A's network disconnects the call by
   sending a Release message REL.  The Gateway maps this to a CANCEL.
   Message Details

   F1 IAM User A -> NGW 1

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

   F2 INVITE A -> 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 - February 2002              [Page 82]

                         SIP PSTN Call Flows              August 2002


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

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

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@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
   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>
   Content-Type: application/sdp
   Content-Length: 141

   v=0
   o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
   c c=IN IP4 192.168.255.101
   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
   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 - February 2002              [Page 83]

                         SIP PSTN Call Flows              August 2002


   CSeq: 1 INVITE
   Content-Length: 0


   F5 INVITE Proxy 1 -> User B

   Same as Message F3


   F6 INVITE Proxy 1 -> User B

   Same as Message F3


   F7 INVITE Proxy 1 -> User B

   Same as Message F3


   F8 INVITE Proxy 1 -> User B

   Same as Message F3


   F9 INVITE Proxy 1 -> User B

   Same as Message F3


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

   F10 REL User A -> NGW 1

   REL
   CauseCode=16 Normal


   F11 RLC NGW 1 -> User A

   RLC


   F12 CANCEL NGW 1 -> Proxy 1

   CANCEL sip:+19725552222@ss11.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 - February 2002              [Page 84]

                         SIP PSTN Call Flows              August 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
   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 - February 2002              [Page 85]

                         SIP PSTN Call Flows              August 2002


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

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


   Message Details


   F1 IAM User A -> 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 - February 2002              [Page 86]

                         SIP PSTN Call Flows              August 2002


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

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

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@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
   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>
   Content-Type: application/sdp
   Content-Length: 141

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

   Same as Message F3


Johnston et al         Expires - February 2002              [Page 87]

                         SIP PSTN Call Flows              August 2002




   F6 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F7 INVITE Proxy 1 -> User B

   Same as Message F3


   F8 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F9 INVITE Proxy 1 -> User B

   Same as Message F3


   F10 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F11 INVITE Proxy 1 -> User B

   Same as Message F3


   F12 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F13 INVITE Proxy 1 -> User B

   Same as Message F3


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

   F14 REL User A -> NGW 1

   REL
   CauseCode=102 Timeout



Johnston et al         Expires - February 2002              [Page 88]

                         SIP PSTN Call Flows              August 2002



   F15 RLC NGW 1 -> User A

   RLC















































Johnston et al         Expires - February 2002              [Page 89]

                         SIP PSTN Call Flows              August 2002


3.9    Unsuccessful PSTN->SIP, Caller Abandonment

   Switch A          NGW 1          Proxy 1           User B
     |                |                |                |
     |     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 calls User B through NGW 1 and Proxy 1.
   User B does not respond with 200 OK.  NGW 1 plays ringing tone since
   the ACM indicates that interworking has been encountered.  User A
   disconnects the call with a Release message REL which is mapped by
   NGW 1 to a CANCEL.  Note that if User B 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 - February 2002              [Page 90]

                         SIP PSTN Call Flows              August 2002




   F1 IAM User A -> NGW 1

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


   F2 INVITE A -> 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>
   Content-Type: application/sdp
   Content-Length: 141

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

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@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
   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>
   Content-Type: application/sdp
   Content-Length: 141


Johnston et al         Expires - February 2002              [Page 91]

                         SIP PSTN Call Flows              August 2002



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


   F4 100 Trying User B -> Proxy 1

   SIP/2.0 100 Trying
   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=192.168.255.201
   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 -> 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 ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
   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>
   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
   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 - February 2002              [Page 92]

                         SIP PSTN Call Flows              August 2002


   CSeq: 1 INVITE
   Contact: <sip:UserB@192.168.200.201>
   Content-Length: 0


   F7 ACM NGW 1 -> User A

   ACM


   /* User A hangs up */

   F8 REL User A -> NGW 1

   REL
   CauseCode=16 Normal


   F9 RLC NGW 1 -> User A

   RLC


   F10 CANCEL NGW 1 -> Proxy 1

   CANCEL 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 CANCEL
   Content-Length: 0


   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
   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



Johnston et al         Expires - February 2002              [Page 93]

                         SIP PSTN Call Flows              August 2002


   CANCEL sip:UserB@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP 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 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
    ;received=192.168.255.111
   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 -> 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
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
   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

   ACK sip:UserB@biloxi.com 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>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0


   F16 487 Request Terminated Proxy 1 -> NGW 1


Johnston et al         Expires - February 2002              [Page 94]

                         SIP PSTN Call Flows              August 2002



   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
    ;received=192.168.255.101
   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/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>;tag=314159
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
   CSeq: 1 ACK
   Content-Length: 0






























Johnston et al         Expires - February 2002              [Page 95]

                         SIP PSTN Call Flows              August 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.


































Johnston et al         Expires - February 2002              [Page 96]

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


Johnston et al         Expires - February 2002              [Page 97]

                         SIP PSTN Call Flows              August 2002


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

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

   Message Details


   F1 IAM Switch A -> 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  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:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 141

   v=0
   o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101
   t=0 0
   m=audio 49172 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 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=z9hG4bKwqwee65


Johnston et al         Expires - February 2002              [Page 98]

                         SIP PSTN Call Flows              August 2002


    ;received=192.168.255.101
   Max-Forwards: 69
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 141

   v=0
   o=GW 2890844526 2890844526 IN IP4 ngw1.atlanta.com
   s=-
   c=IN IP4 192.168.255.101
   t=0 0
   m=audio 49172 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 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
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip:4443333@gw2.atlanta.com>
   Content-Type: application/sdp


Johnston et al         Expires - February 2002              [Page 99]

                         SIP PSTN Call Flows              August 2002


   Content-Length: 149

   v=0
   o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
   s=-
   c=IN IP4 192.168.255.202
   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
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip: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
   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 answers */

   F9 ANM Switch C -> GW 2

   ANM



Johnston et al         Expires - February 2002             [Page 100]

                         SIP PSTN Call Flows              August 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
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip: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
   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
   Record-Route: <sip:ss1.atlanta.com;lr>
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
   CSeq: 1 INVITE
   Contact: <sip: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
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000




Johnston et al         Expires - February 2002             [Page 101]

                         SIP PSTN Call Flows              August 2002



   F12 ANM NGW 1 -> Switch A

   ANM


   F13 ACK NGW 1 -> Proxy 1

   ACK sip:4443333@gw2.atlanta.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:+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 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
   Max-Forwards: 69
   From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
   To: <sip:+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 Hangs Up with User A. */

   F15 REL Switch C -> GW 2

   REL
   CauseCode=16 Normal


   F16 BYE GW 2 -> Proxy 1

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


Johnston et al         Expires - February 2002             [Page 102]

                         SIP PSTN Call Flows              August 2002


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


Johnston et al         Expires - February 2002             [Page 103]

                         SIP PSTN Call Flows              August 2002


   Content-Length: 0


   F21 REL Switch C -> GW 2

   REL
   CauseCode=16 Normal


   F22 RLC GW 2 -> Switch C

   RLC







































Johnston et al         Expires - February 2002             [Page 104]

                         SIP PSTN Call Flows              August 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 - February 2002             [Page 105]

                         SIP PSTN Call Flows              August 2002




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


   Message Details

   PBX A -> GW 1

   Seizure


   GW 1 -> PBX A

   Wink


   F1 MF Digits PBX A -> 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
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


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                         SIP PSTN Call Flows              August 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
   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
   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
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
   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


Johnston et al         Expires - February 2002             [Page 107]

                         SIP PSTN Call Flows              August 2002


   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
   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
   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
   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 - February 2002             [Page 108]

                         SIP PSTN Call Flows              August 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
   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
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
   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
   Record-Route: <sip:ss1.atlanta.com;lr>


Johnston et al         Expires - February 2002             [Page 109]

                         SIP PSTN Call Flows              August 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
   Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
    ;received=192.168.255.201
   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
   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
   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 - February 2002             [Page 110]

                         SIP PSTN Call Flows              August 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
   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
   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 - February 2002             [Page 111]

                         SIP PSTN Call Flows              August 2002




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

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

   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
   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 - February 2002             [Page 112]

                         SIP PSTN Call Flows              August 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
   Via: SIP/2.0/UDP gw3.atlanta.com:5060;branch=z9hG4bKkdjuwq
    ;received=192.168.255.203
   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
   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 -> GW 1

   Seizure removal




Security Considerations

   Since this document represents NON NORMATIVE examples of SIP session
   establishment, the security considerations in RFC 3261 [2] apply.

References




Johnston et al         Expires - February 2002             [Page 113]

                         SIP PSTN Call Flows              August 2002



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

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

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

   4 G. Camarillo, "Best Current Practice for 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 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. and C. Allen, "The TLS Protocol Version 1.0", RFC 2246,
      January 1999.

   8 S. Kent, R. Atkinson, "Security Architecture for the Internet
      Protocol", RFC 2401, November 1998.

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






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.

   The authors wish to thank Neil Deason for his additions to the
   Torture Test messages and Kundan Singh for performing parser
   validation of messages.




Johnston et al         Expires - February 2002             [Page 114]

                         SIP PSTN Call Flows              August 2002


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


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

      EMail:  rsparks@dynamicsoft.com

      Chris Cunningham
      dynamicsoft, Inc.


Johnston et al         Expires - February 2002             [Page 115]

                         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

   "Copyright (C) The Internet Society 2002. All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

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

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.


Acknowledgement



Johnston et al         Expires - February 2002             [Page 116]

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
















































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