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Versions: 00 01 02 03 04 05                                             
Internet Engineering Task Force                          Alan Johnston
Internet Draft                                                WorldCom
Document: draft-ietf-sip-call-flows-05.txt               Steve Donovan
Category: Informational                                  Robert Sparks
June 2001                                             Chris Cunningham
Expires: December 2001                                     Dean Willis
                                                    Jonathan Rosenberg
                                                           dynamicsoft
                                                         Kevin Summers
                                                                 Sonus
                                                   Henning Schulzrinne
                                                   Columbia University



                         SIP Call Flow Examples


   Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026[1].

   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 SIP (Session Initiation
   Protocol) call flows for IP telephony.  Elements in these call flows
   include SIP User Agents and Clients, SIP Proxy and Redirect Servers,
   and Gateways to the PSTN (Public Switch Telephone Network).  IP
   telephony scenarios include SIP Registration, SIP to SIP calling, SIP
   to Gateway, Gateway to SIP, and Gateway to Gateway via SIP.  Call
   flow diagrams and message details are shown.  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).  Example SIP messages used for
   testing during SIP "bakeoff" events include SIP "torture test"
   messages, and messages with invalid parameters, methods, and tags.

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Table of Contents

   1      Overview...................................................4
   1.1    General Assumptions........................................4
   1.2    Legend for Message Flows...................................6
   1.3    SIP Protocol Assumptions...................................7
   1.4    Changes to 00 draft .......................................7
   1.5    Changes to 01 draft .......................................8
   1.6    Changes to 02 draft .......................................8
   1.7    Changes to 03 draft .......................................9
   2      SIP Registration Services.................................10
   2.1    Success Scenarios.........................................10
    2.1.1 SIP Client New Registration...............................10
    2.1.2 User updates contact list.................................13
    2.1.3 User Requests Current Contact List........................15
    2.1.4 User Cancels Registration.................................17
   2.2    Failure Scenarios.........................................18
    2.2.1 Unsuccessful SIP registration.............................18
   3      SIP to SIP Dialing........................................20
   3.1    Success Scenarios.........................................20
    3.1.1 Successful Simple SIP to SIP..............................21
    3.1.2 Successful SIP to SIP through two proxies.................24
    3.1.3 Successful SIP to SIP with Multi Proxy Authentication.....33
    3.1.4 Successful SIP to SIP with Proxy failure..................42
    3.1.5 Successful SIP to SIP through SIP Firewall Proxy..........50
    3.1.6 Successful SIP to SIP via Redirect and Proxy SDP in ACK...58
    3.1.7 Successful SIP to SIP with re-INVITE......................64
   3.2    Failure Scenarios.........................................75
    3.2.1 Unsuccessful SIP to SIP no answer.........................75
    3.2.2 Unsuccessful SIP to SIP busy..............................82
    3.2.3 Unsuccessful SIP to SIP no response.......................86
    3.2.4 Unsuccessful SIP to SIP Temporarily Unavailable...........91
   4      SIP to Gateway Dialing....................................96
   4.1    Success Scenarios.........................................96
    4.1.1 Successful SIP to ISUP PSTN call..........................97
    4.1.2 Successful SIP to ISDN PBX call..........................104
    4.1.3 Successful SIP to ISUP PSTN call with overflow...........111
   4.2    Failure Scenarios........................................119
    4.2.1 Unsuccessful SIP to PSTN call: Treatment from PSTN.......120
    4.2.2 Unsuccessful SIP to PSTN: REL w/Cause from PSTN..........126
    4.2.3 Unsuccessful SIP to PSTN: ANM Timeout....................130
   5      Gateway to SIP Dialing...................................133
   5.1    Success Scenarios........................................133
    5.1.1 Successful PSTN to SIP call..............................134
    5.1.2 Successful PSTN to SIP call, Fast Answer.................140
    5.1.3 Successful PBX to SIP call...............................145
   5.2    Failure Scenarios........................................151
    5.2.1 Unsuccessful PSTN to SIP REL, SIP error mapped to REL....151
    5.2.2 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL.....153
    5.2.3 Unsuccessful PSTN->SIP, SIP error interworking to tones..157
    5.2.4 Unsuccessful PSTN->SIP, ACM timeout......................161



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    5.2.5 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy.....166
    5.2.6 Unsuccessful PSTN->SIP, ANM timeout......................173
   6      Gateway to Gateway Dialing via SIP Network...............179
   6.1    Success Scenarios........................................179
    6.1.1 Successful ISUP PSTN to ISUP PSTN call...................180
    6.1.2 Successful FGB PBX to ISDN PBX call with overflow........187
   7      SIP Test Messages........................................196
   7.1    INVITE Parser Torture Test Message.......................196
   7.2    INVITE with Proxy-Require and Require....................197
   7.3    INVITE with Unknown Schemes in URIs and URLs.............197
   7.4    REGISTER with Y2038 Test.................................198
   7.5    INVITE with inconsistent Accept and message body.........198
   7.6    INVITE with non-SDP message body.........................199
   7.7    Unknown Method Message...................................199
   7.8    Unknown Method with CSeq Error...........................200
   7.9    REGISTER with Unknown Authorization Scheme...............200
   7.10   Multiple SIP Request in a Single Message.................201
   7.11   INVITE missing Required Headers..........................201
   7.12   INVITE with Duplicate Required Headers...................202
   7.13   INVITE with Illegal Expires Header.......................202
   7.14   200 OK Response with Broadcast Via Header................203
   7.15   INVITE with Invalid Via and Contact Headers..............204
   7.16   INVITE with Incorrect Content-Length Header..............204
   7.17   INVITE with Invalid Value for Content-Length.............205
   7.18   INVITE with Garbage after Message Body...................205
   7.19   INVITE with Error in Display Name in To Header...........206
   7.20   INVITE with long tokens, Call-ID, and SIP URLs...........206
   7.21   INVITE with Illegal Enclosing of Request-URI in "<>".....207
   7.22   INVITE with Illegal LWS within Elements of Request-URI...207
   7.23   INVITE with Illegal >1 SP between elements Request-URI...208
   7.24   INVITE with a Legal SIP URL With Escaped Characters......209
   7.25   INVITE with the Illegal Escaped Headers Request-URI......209
   7.26   INVITE with Unknown in Request-URI.......................210
   7.27   OPTIONS with no LWS between display name and "<".........210
   7.28   OPTIONS with Extra WS between display name and "<".......211
   7.29   INVITE with an Illegal Date Format.......................211
   7.30   INVITE with Passed Expires Time..........................212
   7.31   INVITE with Max-Forwards Set to Zero.....................212
   7.32   REGISTER with Escaped Header in a Legal Contact URL......213
   7.33   REGISTER with Escaped Header in an Illegal Contact URL...213
   7.34   INVITE with Long Values in Headers.......................214
   7.35   OPTIONS with Multiple Headers............................215
   7.36   INVITE with Large Number of SDP Attributes and Long URI..216
   7.37   REGISTER with Contact Parameter..........................216
   7.38   REGISTER with a URL Parameter............................217
   7.39   INVITE with Unquoted Display Name with Tokens............217
   7.40   INVITE with Unquoted Display Name with Non-Token Chars...218
   7.41   INVITE with Unknown (Higher) Protocol Version Start Line.218
   8      Acknowledgements.........................................218
   9      References...............................................219




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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.  The message examples were developed during the SIP
   interoperability testing "bake-offs."

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

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

   These call flows are based on the current version 2.0 of SIP in
   RFC2543[2].

   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.  The intent
   of this document was not to provide a complete and exact mapping of
   PSTN protocols to SIP.  Rather the emphasis is on the SIP signaling,
   the message interaction, and the modifications to SIP currently
   proposed to solve IP Telephony issues.

   Finally, some example messages are given along with expected
   behavior of clients and servers.

1.1 General Assumptions

   A number of architecture, network, and protocol assumptions underlie
   the call flows in this document. Note that these assumptions are NOT
   requirements.  They are outlined in this section so that they may be
   taken into consideration and to aide in the understanding of the call
   flow examples.

   The authentication of SIP User Agents in these example call flows is
   performed using SIP Digest[2].

   No authentication of Gateways is shown, since it is assumed that:
     . Gateways will only accept calls routed through a trusted Proxy.
     . Proxies will perform the Client authentication.



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     . The Proxy and the Gateway will authenticate each other using
       IPSec[3] or some other non-SIP scheme.

   The SIP Proxy Server has access to a Location Service and other
   databases.  Information present in the Request-URI and the context
   (From header) is sufficient to determine to which proxy or gateway
   the message should be routed.  In most cases, a primary and secondary
   route will be determined in case of Proxy or Gateway failure
   downstream.

   The 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.  This allows them to implement features
   later in the call, which are not shown in these flows.

   Gateways receive enough information in the Request-URI field to
   determine how to route a call, i.e. what trunk group or link to
   select, what digits to outpulse, etc.

   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.

   Two types of Gateways are described in this document.  The actual
   names of Gateways will be vendor and implementation specific.
   However, two catagories are described here since the type of Gateway
   determines the form of the SIP URL used to identify them.  The two
   types are:

     . Network Gateway.  This high port count PSTN gateway originates
       and terminates calls to the PSTN.  Its use is shared by many
       customers.  Incoming calls from the PSTN have the From header
       populated with a SIP URL containing the telephone number from
       the calling party telephone number, if available.  A Network
       Gateway typically uses carrier protocols such as SS7.

     . Enterprise Gateway.  This low port count PBX (Private Branch
       Exchange) gateway has trunks or lines for a single customer or
       user.  Incoming calls from the PBX have the From header
       populated with a provisionable string which uniquely identifies
       the customer, trunk group, or carrier.  This allows private
       numbers to be interpreted in their correct context.  An
       Enterprise Gateway typically uses SS7, ISDN, circuit associated
       signaling, or other PBX interfaces.

   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.



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

   Digit handling by the Gateways will be  as follows:

     . Dialed digits received from a Network or Enterprise Gateway will
       be put in a SIP URL with a telephone number.  The number will
       either be globalized (e.g. sip:+1-314-555-1111@ngw.wcom.com
       ;user=phone) or left as a private number (sip:555-
       6666@wcom.com) Alternatively, the "phone-context"
       qualifier could be used to interpret the private number.  It is
       defined in the telephony URL document [4].
       All Gateways will need to be provisioned to be able to parse the
       user portion of a Request-URI to determine the customer, trunk
       group, or circuit referenced.  Note that the visual separator "-"
       is used purely to aid in the readability of the examples; a real
       gateway would be unlikely to insert visual separators.

     . The From header will be populated with a SIP URL with a
       telephone number if it is Calling Party number (CgPN) from the
       PSTN.  If it is an Enterprise Gateway, a provisionable  string
       which uniquely identifies the customer, trunk group, or carrier
       will be used in the sip URI (e.g. From:
       sip:ProvisionableString@gw1.wcom.com;user=phone).

     . Note that an alternative to using a SIP URL for telephone
       numbers is the tel URL[4].  The major difference between using
       the SIP URL and the tel URL is that the SIP URL is routable in a
       SIP network (resolves down to an IP address) where the tel URL
       is not (it just represents digits).

   These flows show UDP for transport.  TCP could also be used.

1.2 Legend for Message Flows

   Dashed lines (---) represent control messages that are mandatory to
   the call scenario. These control messages can be SIP or PSTN
   signaling.

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

   Messages with parenthesis around name represent optional control
   messages.

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

    /* Comments. */



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1.3 SIP Protocol Assumptions

   Except for the following, this call flows document uses the April
   1999 version 2.0 of SIP defined by RFC 2543[2].  The following
   changes/extensions are assumed throughout:

     . A Contact header is included with every INVITE message.

     . A Contact header is included in every 200 OK Response.

     . The 183 Session Progress response message is used in SIP to
       Gateway and Gateway to Gateway via SIP calling (Sections 4 and
       6). The 183 response indicates to the User Agent that a Gateway
       has been contacted and is trying to complete the call to the PSTN
       and that one-way early media may be present which gives an
       indication of the progress of the call.  The User Agent will
       immediately play to the user any RTP media packets received to
       hear in-band call progress information such as ringtone or busy
       tone.  Note that SDP is not required in the 183 response since
       the media is uni-directional.  See Section 4 for more
       information.

     . A Content-Length header is present in every message, set to zero
       if there is no message body.  The content length calculations
       assume that each line of SDP ends with both a CR and a LF
       character.

     . The final entry in a Route header is always the Contact
       information obtained from the INVITE or 200 OK messages.

     . In the SDP message bodies, the time field is "t=0 0"  It is
       expected that an actual SDP message body would have a non-zero
       start timestamp.

     . Branch tags inserted by proxies are unique for each request.  In
       this document, they are not due to editing simplications.

     . Tags and Call-IDs are also reused for editing purposes.

     . Other simplifications have been made for editing purposes.  For
       example, the order of SIP headers is fairly consistent throughout
       the document.  With a few exceptions, the ordering of SIP headers
       is not significant.

   In any discrepency between this document and RFC 2543, follow RFC
   2543.  This document is informational only.

1.4 Changes to 00 draft





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   The major changes between this draft and the previous draft are
   listed below:

    - SIP Telephony Service Examples have been removed from the draft.
      They will be revised using the TRANSFER header in a separate
      draft.

    - Updated draft with RFC2543bis changes including: adding maddr to
      all Record-Route and Route headers, adding branch tags to Via
      headers inserted by proxies, added 487 response to CANCEL
      scenarios.

    - Added example of INFO method in 5.1.1.

    - Added Session: media to all 183 messages.

    - Corrected a number of typos including putting user=phone tags
      inside <>, fixing Request-URI on PRACK, added missing tags, fixed
      Request-URIs that did not match To header in initial INVITE.

    - Corrected all registrations to have same Call-ID.

1.5 Changes to 01 draft

   The major changes between this draft and the previous draft are
   listed below:

    - Added an example of multiple proxy authentication (3.1.3).

    - Removed INFO method from 5.1.1. The use of INFO to transport
      DTMF is controversial.

    - Fixed CSeq and Call-ID errors in Section 2.

    - Replaced Session: media with Content-Disposition: session

    - Added notes about CSeq numbering spaces.

    - Fixed maddr in some Route headers.

    - Corrected tag handling and Record-Routes in Section 4.


1.6 Changes to 02 draft

   The major changes between this draft and previous are listed below:

    - The word "Telephony" has been removed from the document title.

    - Selected messages have been successfully validated against a SIP
      parser for accuracy.



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    - Added word "informational" to Abstract, and added additional
      wording about this document being informational and not normative.

    - Added re-INVITE call flow (T.38 Fax)in 3.1.7.

    - Removed SDP from all 183 Session Progress messages since it is not
      needed.  Corrected figures to show early media path as one way.

    - Removed PRACK from example 4.1.2 since there is no longer any need
      for the 183 to be transported reliably since it no longer contains
      SDP.

    - Corrected Call-ID in 3.1.1.

    - Corrected domain in WWW-Authenticate and Proxy-Authenticate
      headers to be a SIP URL.

    - Section 7.4 SIP Date corrected to Sat, 01 Dec 2040 16:00:00 GMT

    - Section 7.7 Retransmission strategy for unknown method message
      should be based on BYE not INVITE

    - Section 7.10 multiple message element of this test removed as this
      behaviour is no longer allowed, higher SIP version number part
      moved into 7.42

    - Section 7.18 recommended response changed

    - Added Test Messages 7.21 through 7.41


1.7 Changes to the 03 draft

    - Replaced Authorization headers with Proxy-Authorization headers

    - Fixed Record-Route in example 3.1.3

    - Various CSeq and Content-Length corrections

1.8 Changes to the 04 draft

   - Removed domain names from Contact headers

   - Removed "phone-context" URI parameter.

   - Added expires parameter to 200 OK responses to REGISTERs.

   - Changed Proxy-Authorization headers in Section 2 to Authorization
     headers.




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   2  SIP Registration Services

2.1 Success Scenarios

   Registration either validates or invalidates a SIP client for user
   services provided by a SIP server.  Additionally, the client provides
   one or more contact locations to the SIP server with the registration
   request.  Registration is used by a Proxy to route incoming calls in
   an IP Telephony network.  Registration are shown with authentication
   in these call flows.  If authentication is not used, an imposter
   could "hijack" someone else's calls.

2.1.1 SIP Client New Registration

    User B                        SIP Server
     |                               |
     |          REGISTER F1          |
     |------------------------------>|
     |                               |
     |      401 Unauthorized F2      |
     |<------------------------------|
     |                               |
     |          REGISTER F3          |
     |------------------------------>|
     |                               |
     |            200 OK F4          |
     |<------------------------------|
     |                               |


   User B initiates a new SIP session with the SIP Server (i.e. the user
   "logs on to" the SIP server).  User B sends a SIP REGISTER request to
   the SIP server.  The request includes the user's contact list.  The
   SIP server provides a challenge to User B.  User B enters her/his
   valid user ID and password.  User B's SIP client encrypts the user
   information according to the challenge issued by the SIP server and
   sends the response to the SIP server.  The SIP server validates the
   user's credentials.  It registers the user in its contact database
   and returns a response (200 OK) to User B's SIP client.  The response
   includes the user's current contact list in Contact headers.  The
   format of the authentication shown is SIP digest as described by
   RFC 2543[2].  It is assumed that User B has not previously registered
   with this Server.


   Message Details


   F1 REGISTER B -> SIP Server

   REGISTER sip:ss2.wcom.com SIP/2.0



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   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   Contact: <sip:UserB@110.111.112.113>
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>
   Contact: tel:+1-972-555-2222
   Content-Length: 0


   F2 401 Unauthorized SIP Server -> User B

   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   WWW-Authenticate: Digest realm="MCI WorldCom SIP",
    domain="sip:ss2.wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0


   F3 REGISTER B -> SIP Server

   REGISTER sip:ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 2 REGISTER
   Contact: <sip:UserB@110.111.112.113>
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>
   Contact: tel:+1-972-555-2222
   Authorization:Digest username="UserB",
    realm="MCI WorldCom SIP",
    nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
    uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc"
   Content-Length: 0


   F4 200 OK SIP Server -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 2 REGISTER



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   Contact: <sip:UserB@110.111.112.113>;expires=3600
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>;expires=3600
   Contact: <tel:+1-972-555-2222>;expires=4294967295
   Content-Length: 0


















































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   2.1.2 User updates contact list


   User B                        SIP Server
     |                               |
     |          REGISTER F1          |
     |------------------------------>|
     |                               |
     |            200 OK F2          |
     |<------------------------------|
     |                               |


   User B wishes to update the list of addresses where the SIP server
   will redirect or forward INVITE requests.

   User B sends a SIP REGISTER request to the SIP server.  User B's
   request includes an updated contact list.  Since the user already has
   authenticated with the server, the user supplies authentication
   credentials with the request and is not challenged by the server.
   The SIP server validates the user's credentials.  It registers the
   user in its contact database, updates the user's contact list, and
   returns a response (200 OK) to User B's SIP client.  The response
   includes the user's current contact list in Contact headers.

   Message Details


   F1 REGISTER B -> SIP Server

   REGISTER sip:ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   Contact: mailto:UserB@there.com
   Authorization:Digest username="UserB",
    realm="MCI WorldCom SIP",
    nonce="1cec4341ae6cbe5a359ea9c8e88df84f", opaque="",
    uri="sip:ss2.wcom.com", response="71ba27c64bd01de719686aa4590d5824"
   Content-Length: 0


   F2 200 OK SIP Server -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com



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   CSeq: 1 REGISTER
   Contact: <sip:UserB@110.111.112.113>;expires=3600
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>;expires=3600
   Contact: <tel:+1-972-555-2222>;expires=4294967295
   Contact: <mailto:UserB@there.com>;expires=4294967295
   Content-Length: 0
















































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2.1.3 User Requests Current Contact List


   User B                        SIP Server
     |                               |
     |          REGISTER F1          |
     |------------------------------>|
     |                               |
     |            200 OK F2          |
     |<------------------------------|
     |                               |

   User B sends a register request to the Proxy Server containing no
   Contact headers, indicating the user wishes to query the server for
   the user's current contact list.  Since the user already has
   authenticated with the server, the user supplies authentication
   credentials with the request and is not challenged by the server. The
   SIP server validates the user's credentials.  The server returns a
   response (200 OK) which includes the user's current registration list
   in Contact headers.


   Message Details

   F1 REGISTER B -> SIP Server

   REGISTER sip:ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   Authorization:Digest username="UserB",
    realm="MCI WorldCom SIP",
    nonce="df84f1cec4341ae6cbe5ap359a9c8e88", opaque="",
    uri="sip:ss2.wcom.com", response="aa7ab4678258377c6f7d4be6087e2f60"
   Content-Length: 0


   F2 200 OK SIP Server -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   Contact: <sip:UserB@110.111.112.113>;expires=3600
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>;expires=3600
   Contact: <tel:+1-972-555-2222>;expires=4294967295
   Contact: <mailto:UserB@there.com>;expires=4294967295



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   Content-Length: 0





















































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   2.1.4 User Cancels Registration

   User B                         SIP Server
     |                               |
     |          REGISTER F1          |
     |------------------------------>|
     |                               |
     |            200 OK F2          |
     |<------------------------------|
     |                               |

   User B wishes to cancel their registration with the SIP
   server. User B sends a SIP REGISTER request to the
   SIP server.  The request has an expiration period of 0 and applies to
   all existing contact locations.  Since the user already has
   authenticated with the server, the user supplies authentication
   credentials with the request and is not challenged by the server.
   The SIP server validates the user's credentials.  It clears the
   user's contact list, and returns a response (200 OK) to User B's SIP
   client.

   Message Details


   F1 REGISTER B -> SIP Server

   REGISTER sip:ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   Expires: 0
   Contact: *
   Authorization:Digest username="UserB", realm="MCI WorldCom SIP",
    nonce="88df84f1cac4341aea9c8ee6cbe5a359", opaque="",
    uri="sip:ss2.wcom.com", response="ff0437c51696f9a76244f0cf1dbabbea"
   Content-Length: 0


   F2 200 OK SIP Server -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   Content-Length: 0





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2.2 Failure Scenarios

2.2.1 Unsuccessful SIP registration

   User B                        SIP Server
     |                               |
     |          REGISTER F1          |
     |------------------------------>|
     |                               |
     |      401 Unauthorized F2      |
     |<------------------------------|
     |                               |
     |          REGISTER F3          |
     |------------------------------>|
     |                               |
     |      401 Unauthorized F4      |
     |<------------------------------|
     |                               |



   User B sends a SIP REGISTER request to the SIP Server.  The SIP
   server provides a challenge to User B.  User B enters her/his user ID
   and password.  User B's SIP client encrypts the user information
   according to the challenge issued by the SIP server and sends the
   response to the SIP server.  The SIP server attempts to validate the
   user's credentials, but they are not valid (the user's password does
   not match the password established for the user's account).  The
   server returns a response (401 Unauthorized) to User B's SIP client.



   Message Details


   F1 REGISTER B -> SIP Server

   REGISTER sip:ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   Contact: <sip:UserB@110.111.112.113>
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>
   Contact: tel:+1-972-555-2222
   Content-Length: 0


   F2 Unauthorized SIP Server -> User B




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   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 1 REGISTER
   WWW-Authenticate: Digest realm="MCI WorldCom SIP",
    domain="sip:ss2.wcom.com", once="f1cec4341ae6ca9c8e88df84be55a359",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0


   F3 REGISTER B -> SIP Server

   REGISTER sip:ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 2 REGISTER
   Contact: <sip:UserB@110.111.112.113>
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>
   Contact: tel:+1-972-555-2222
   Authorization:Digest username="UserB", realm="MCI WorldCom SIP",
    nonce="f1cec4341ae6ca9c8e88df84be55a359", opaque="",
    uri="sip:ss2.wcom.com", response="61f8470ceb87d7ebf508220214ed438b"
   Content-Length: 0


   /*  The response above encodes the incorrect password */

   F4 401 Unauthorized SIP Server -> User B

   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 123456789@there.com
   CSeq: 2 REGISTER
   WWW-Authenticate: Digest realm="MCI WorldCom SIP",
    domain="sip:ss2.wcom.com", nonce="84f1c1ae6cbe5ua9c8e88dfa3ecm3459",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0











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3  SIP to SIP Dialing

3.1  Success Scenarios

   This section details calls between two SIP User Agents (UAs):
   User A and User B.  User A (LittleGuy sip:UserA@here.com) and User
   B (BigGuy sip:UserB@there.com) are assumed to be SIP phones or SIP-
   enabled devices.  The successful calls show the initial signaling,
   the exchange of media information in the form of SDP payloads, the
   establishment of the media session, then finally the termination of
   the call.

   SIP digest authentication is used by Proxy Servers to
   authenticate the caller User A. It is assumed that User B has
   registered with Proxy Server Proxy 2 as per Section 2.1 to be able to
   receive the calls via the Proxy.






































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3.1.1 Successful Simple SIP to SIP

   User A                  User B
     |                        |
     |       INVITE F1        |
     |----------------------->|
     |    (100 Trying) F2     |
     |<-----------------------|
     |    180 Ringing F3      |
     |<-----------------------|
     |                        |
     |       200 OK F4        |
     |<-----------------------|
     |         ACK F5         |
     |----------------------->|
     |   Both Way RTP Media   |
     |<======================>|
     |                        |
     |         BYE F6         |
     |<-----------------------|
     |       200 OK F7        |
     |----------------------->|
     |                        |

   In this scenario, User A completes a call to User B directly.


   Message Details


   F1 INVITE User A -> User B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000





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   F2 (100 Trying) User B -> User A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F3 180 Ringing User B -> User A

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=8321234356
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F4 200 OK User B -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=8321234356
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 ACK User A -> User B

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=8321234356
   Call-ID: 12345601@here.com
   CSeq: 1 ACK



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   Content-Length: 0


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

   /* User B Hangs Up with User A. Note that the CSeq is NOT 2, since
      User A and User B maintain their own independent CSeq counts.
      (The INVITE was request 1 generated by User A, and the BYE is
      request 1 generated by User B) */

   F6 BYE User B -> User A

   BYE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=8321234356
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F7 200 OK User A -> User B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=8321234356
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0
























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3.1.2 Successful SIP to SIP through two proxies

   User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     407 F2     |                |                |
     |<---------------|                |                |
     |     ACK F3     |                |                |
     |--------------->|                |                |
     |   INVITE F4    |                |                |
     |--------------->|   INVITE F5    |                |
     |    (100) F6    |--------------->|   INVITE F7    |
     |<---------------|    (100) F8    |--------------->|
     |                |<---------------|                |
     |                |                |     180 F9     |
     |                |    180 F10     |<---------------|
     |     180 F11    |<---------------|                |
     |<---------------|                |     200 F12    |
     |                |    200 F13     |<---------------|
     |     200 F14    |<---------------|                |
     |<---------------|                |                |
     |     ACK F15    |                |                |
     |--------------->|    ACK F16     |                |
     |                |--------------->|     ACK F17    |
     |                |                |--------------->|
     |                Both Way RTP Media                |
     |<================================================>|
     |                |                |     BYE F18    |
     |                |    BYE F19     |<---------------|
     |     BYE F20    |<---------------|                |
     |<---------------|                |                |
     |     200 F21    |                |                |
     |--------------->|     200 F22    |                |
     |                |--------------->|     200 F23    |
     |                |                |--------------->|
     |                |                |                |

   In this scenario, User A completes a call to User B using two proxies
   Proxy 1 and Proxy 2.  The initial INVITE (F1) does not contain the
   Authorization credentials Proxy 1 requires, so a 407 Proxy
   Authorization response is sent containing the challenge information.
   A new INVITE (F4) is then sent containing the correct credentials and
   the call proceeds.  The call terminates when User B disconnects by
   initiating a BYE message.

   Proxy 1 inserts a Record-Route header into the INVITE message to
   ensure that it is present in all subsequent message exchanges.  Proxy
   2 also inserts itself into the Record-Route header.  The ACK (F15)
   and BYE (F18) both have a Route header.




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   A tag is inserted by User B in message F9 since the initial INVITE
   message contains more than one Via header and may have been forked.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 challenges User A for authentication */

   F2 407 Proxy Authorization Required Proxy 1 -> User A

   SIP/2.0 407 Proxy Authorization Required
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Proxy-Authenticate: Digest realm="MCI WorldCom SIP",
    domain="sip:ss1.wcom.com", nonce="f84f1cec41e6cbe5aea9c8e88d359",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0


   F3 ACK A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com



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   CSeq: 1 ACK
   Content-Length: 0


   /* User A responds be re-sending the INVITE with authentication
      credentials in it.  A new Call-ID is used, so the CSeq is reset
      back to 1 */

   F4 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359", opaque="",
    uri="sip:ss1.wcom.com", response="42ce3cef44b22f50c6a6071bc8"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE to Proxy
   2.  Proxy 1 is assumed to have been authenticated by Proxy 2 using
   IPSec.  Client for A prepares to receive data on port 49172 from the
   network. */

   F5 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147



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   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F6 (100 Trying) Proxy 1 -> User A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 INVITE Proxy 2 -> B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 (100 Trying) Proxy 2 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060



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   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 180 Ringing B -> Proxy 2

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F10 180 Ringing Proxy 2 -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F11 180 Ringing Proxy 1 -> A

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F12 200 OK B -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>



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   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F13 200 OK Proxy 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE



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   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F15 ACK A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F16 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F17 ACK Proxy 2 -> B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0




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

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

   /* Again, note that the CSeq is NOT 2.  User A and User B maintain
      their own separate CSeq counts */

   F18 BYE User B -> Proxy 2

   BYE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com;maddr=ss1.wcom.com>,
    <sip:UserA@100.101.102.103>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F19 BYE Proxy 2 -> Proxy 1

   BYE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@100.101.102.103>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F20 BYE Proxy 1 -> User A

   BYE sip:UserA@100.101.102.103 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F21 200 OK User A -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1



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   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F22 200 OK Proxy 1 -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F23 200 OK Proxy 2 -> User B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 BYE
   Content-Length: 0
























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3.1.3 SIP to SIP with Multi Proxy Authentication

     User A       Proxy 1     Proxy 2       User B
       |  INVITE F1 |           |             |
       |----------->|           |             |
       |  407 Proxy Authorization Required F2 |
       |<-----------|           |             |
       |   ACK F3   |           |             |
       |----------->|           |             |
       |  INVITE F4 |           |             |
       |----------->|           |             |
       |  (100 F5)  |           |             |
       |<-----------| INVITE F6 |             |
       |            |---------->|             |
       |            |  407 Proxy Authorization Required F7
       |            |<----------|             |
       |            |   ACK F8  |             |
       |            |---------->|             |
       |  407 Proxy Authorization Required F9 |
       |<-----------|           |             |
       |   ACK F10  |           |             |
       |----------->|           |             |
       |  INVITE F11|           |             |
       |----------->|           |             |
       |  (100 F12) |           |             |
       |<-----------| INVITE F13|             |
       |            |---------->|             |
       |            | (100 F14) |             |
       |            |<----------|  INVITE F15 |
       |            |           |------------>|
       |            |           | 200 OK F16  |
       |            | 200 OK F17|<------------|
       | 200 OK F18 |<----------|             |
       |<-----------|           |             |
       |   ACK F19  |           |             |
       |----------->|  ACK F20  |             |
       |            |---------->|   ACK F21   |
       |            |           |------------>|
       |           RTP Media Path             |
       |<====================================>|

   In this scenario, User A completes a call to User B using two proxies
   Proxy 1 and Proxy 2.  The initial INVITE (F1) does not contain the
   Authorization credentials Proxy 1 requires, so a 407 Proxy
   Authorization response is sent containing the challenge information.
   A new INVITE (F4) is then sent containing the correct credentials and
   the call proceeds after Proxy 2 challenges and receives valid
   credentials.  The call terminates when User B disconnects by
   initiating a BYE message.

   Proxy 1 inserts a Record-Route header into the INVITE message to



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   ensure that it is present in all subsequent message exchanges.  Proxy
   2 also inserts itself into the Record-Route header.



   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 challenges User A for authentication */

   F2 407 Proxy Authorization Required Proxy 1 -> User A

   SIP/2.0 407 Proxy Authorization Required
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Proxy-Authenticate: Digest realm="WorldCom SIP",
    domain="sip:ss1.wcom.com", nonce="wf84f1cczx41ae6cbeaea9ce88d359",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0


   F3 ACK A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>



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   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* User A responds be re-sending the INVITE with authentication
      credentials in it.  The same Call-ID is used, so the CSeq is
      increased. */

   F4 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA", realm="WorldCom SIP",
    nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359", opaque="",
    uri="sip:ss1.wcom.com", response="42ce3cef44b22f50c6a6071bc8"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE to Proxy
   2.  Proxy 1 is assumed to have been authenticated by Proxy 2 using
   IPSec.  Client for A prepares to receive data on port 49172 from the
   network. */

   F5 (100 Trying Proxy 1 -> User A)

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   F6 INVITE Proxy 1 -> Proxy 2




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   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 2 challenges User A for authentication */

   F7 407 Proxy Authorization Required Proxy 2 -> Proxy 1

   SIP/2.0 407 Proxy Authorization Required
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=838209
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Proxy-Authenticate: Digest realm="MCI SIP",
    domain="sip:ss2.mci.com", nonce="c1e22c41ae6cbe5ae983a9c8e88d359",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0


   F8 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=838209
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0






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   /* Proxy 1 forwards the challenge to User A for authentication from
   Proxy 2 */

   F9 407 Proxy Authorization Required Proxy 1 -> User A

   SIP/2.0 407 Proxy Authorization Required
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=2341d
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Proxy-Authenticate: Digest realm="MCI SIP",
    domain="sip:ss2.mci.com", nonce="c1e22c41ae6cbe5ae983a9c8e88d359",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0


   F10 ACK User A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=2341d
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   /* User A responds be re-sending the INVITE with authentication
   credentials for Proxy 1 AND Proxy 2.  */

   F11 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 3 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA", realm="WorldCom SIP",
    nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359", opaque="",
    uri="sip:ss1.wcom.com", response="42ce3cef44b22f50c6a6071bc8"
   Proxy-Authorization:Digest username="UserA", realm="MCI SIP",
    nonce="c1e22c41ae6cbe5ae983a9c8e88d359", opaque="",
    uri="sip:ss2.mci.com", response="f44ab22f150c6a56071bce8"
   Content-Type: application/sdp
   Content-Length: 147

   v=0



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   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 finds its credentials and authorizes User A, forwarding
   the INVITE to Proxy.  */


   F12 (100 Trying Proxy 1 -> User A)

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 3 INVITE
   Content-Length: 0


   F13 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 3 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA", realm="MCI SIP",
    nonce="c1e22c41ae6cbe5ae983a9c8e88d359", opaque="",
    uri="sip:ss2.mci.com", response="f44ab22f150c6a56071bce8"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 2 finds its credentials and authorizes User A, forwarding
   the INVITE to User B.  */



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   F14 (100 Trying Proxy 2 -> Proxy 1)

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 3 INVITE
   Content-Length: 0


   F15 INVITE Proxy 2 -> User B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.mci.com:5060;branch=31972.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.mci.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 3 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* User B answers the call immediately */

   F16 200 OK User B -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.mci.com:5060;branch=31972.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.mci.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9103874
   Call-ID: 12345600@here.com



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   CSeq: 3 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F17 200 OK Proxy 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.mci.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9103874
   Call-ID: 12345600@here.com
   CSeq: 3 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F18 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.mci.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9103874
   Call-ID: 12345600@here.com
   CSeq: 3 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 149



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   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F19 ACK User A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com;maddr=ss2.mci.com>,
    <sip:UserB@there.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9103874
   Call-ID: 12345600@here.com
   CSeq: 3 ACK
   Content-Length: 0


   F20 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9103874
   Call-ID: 12345600@here.com
   CSeq: 3 ACK
   Contact: <sip:UserB@110.111.112.113>
   Content-Length: 0


   F21 ACK Proxy 2 -> User A

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.mci.com:5060;branch=31972.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=230f2.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9103874
   Call-ID: 12345600@here.com
   CSeq: 3 ACK
   Contact: <sip:UserB@110.111.112.113>
   Content-Length: 0





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3.1.4 Successful SIP to SIP with Proxy failure

   User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |   INVITE F2    |                |                |
     |--------------->|                |                |
     |   INVITE F3    |                |                |
     |--------------->|                |                |
     |   INVITE F4    |                |                |
     |--------------->|                |                |
     |   INVITE F5    |                |                |
     |--------------->|                |                |
     |   INVITE F6    |                |                |
     |--------------->|                |                |
     |   INVITE F7    |                |                |
     |--------------->|                |                |
     |    CANCEL F8   |                |                |
     |--------------->|                |                |
     |     INVITE F9                   |                |
     |-------------------------------->|                |
     |            407 F10              |                |
     |<--------------------------------|                |
     |             ACK F11             |                |
     |-------------------------------->|                |
     |     INVITE F12                  |                |
     |-------------------------------->|   INVITE F13   |
     |            (100) F14            |--------------->|
     |<--------------------------------|                |
     |                                 |     180 F15    |
     |             180 F16             |<---------------|
     |<--------------------------------|                |
     |                                 |     200 F17    |
     |             200 F18             |<---------------|
     |<--------------------------------|                |
     |             ACK F19             |                |
     |-------------------------------->|     ACK F20    |
     |                                 |--------------->|
     |                Both Way RTP Media                |
     |<================================================>|
     |                                 |     BYE F21    |
     |             BYE F22             |<---------------|
     |<--------------------------------|                |
     |             200 F23             |                |
     |-------------------------------->|     200 F24    |
     |                                 |--------------->|
     |                                 |                |






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   In this scenario, User A completes a call to User B via a Proxy
   Server. User A is configured for a primary SIP Proxy Server Proxy 1
   and a secondary SIP Proxy Server Proxy 2 (Or is able to use DNS SRV
   records to locate Proxy 1 and Proxy 2). Proxy 1 is out of service and
   does not respond to INVITEs (it is reachable, but unresponsive).
   After sending a CANCEL to Proxy 1, User A then completes the call to
   User B using Proxy 2.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 INVITE A -> Proxy 1

   Same as Message F1


   F3 INVITE A -> Proxy 1

   Same as Message F1


   F4 INVITE A -> Proxy 1

   Same as Message F1


   F5 INVITE A -> Proxy 1

   Same as Message F1



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   F6 INVITE A -> Proxy 1

   Same as Message F1


   F7 INVITE A -> Proxy 1

   Same as Message F1


   /* User A gives up on the unresponsive proxy and sends a CANCEL. */

   F8 CANCEL A -> Proxy 1

   CANCEL sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL


   F9 INVITE A -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 2 challenges User A for authentication */

   F10 407 Proxy Authorization Required Proxy 2 -> User A

   SIP/2.0 407 Proxy Authorization Required
   Via: SIP/2.0/UDP here.com:5060



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   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Proxy-Authenticate: Digest realm="MCI SIP",
    domain="sip:ss2.wcom.com", nonce="1ae6cbe5ea9c8e8df84fqnlec434a359",
    opaque="", stale=FALSE, algorithm=MD5
   Content-Length: 0


   F11 ACK A -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* User A responds be re-sending the INVITE with authentication
   credentials in it.  */

   F12 INVITE A -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA", realm="MCI SIP",
    nonce="1ae6cbe5ea9c8e8df84fqnlec434a359", opaque="",
    uri="sip:ss2.wcom.com", response="8a880c919d1a52f20a1593e228adf599"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 2 accepts the credentials and forwards the INVITE to User B.
   Client for A prepares to receive data on port 49172 from the network.
   */



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   F13 INVITE Proxy 2 -> B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 (100 Trying) Proxy 2 -> User A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F15 180 Ringing B -> Proxy 2

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F16 180 Ringing Proxy 2 -> A

   SIP/2.0 180 Ringing



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   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F17 200 OK B -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F18 200 OK Proxy 2 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000



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   F19 ACK A -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345602@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F20 ACK Proxy 2 -> B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345602@here.com
   CSeq: 1 ACK
   Content-Length: 0


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

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

   F21 BYE User B -> Proxy 2

   BYE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@100.101.102.103>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345602@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F22 BYE Proxy 2 -> User A

   BYE sip:UserA@100.101.102.103 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345602@here.com
   CSeq: 1 BYE



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   Content-Length: 0


   F23 200 OK User A -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345602@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F24 200 OK Proxy 2 -> User B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345602@here.com
   CSeq: 1 BYE
   Content-Length: 0






























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3.1.5 Successful SIP to SIP through SIP Firewall Proxy

   User A          FW Proxy         Proxy 1          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|   INVITE F2    |                |
     |    (100) F3    |--------------->|   INVITE F4    |
     |<---------------|    (100) F5    |--------------->|
     |                |<---------------|      180 F6    |
     |                |     180 F7     |<---------------|
     |     180 F8     |<---------------|                |
     |<---------------|                |      200 F9    |
     |                |    200 F10     |<---------------|
     |     200 F11    |<---------------|                |
     |<---------------|                |                |
     |     ACK F12    |                |                |
     |--------------->|     ACK F13    |                |
     |                |--------------->|     ACK F14    |
     |                |                |--------------->|
     |    RTP Media   |        Both Way RTP Media       |
     |<==============>|<===============================>|
     |     BYE F15    |                |                |
     |--------------->|     BYE F16    |                |
     |                |--------------->|     BYE F17    |
     |                |                |--------------->|
     |                |                |     200 F18    |
     |                |     200 F19    |<---------------|
     |     200 F20    |<---------------|                |
     |<---------------|                |                |
     |                |                |                |

   User A completes a call to User B through a Firewall Proxy and a SIP
   Proxy.  The signaling message exchange is identical to 3.1.1 but the
   media stream setup is not end-to-end - the Firewall proxy terminates
   both media streams and bridges them.  This is done by the Proxy
   modifying the SDP in the INVITE (F1) and 200 OK (F10) messages, and
   possibly any 18x or ACK messages containing SDP.

   In addition to firewall traversal, this Back-to-Back User Agent
   (B2BUA) could be used as part of an anonymizer
   service (in which all identifying information on User A would be
   removed), or to perform codec media conversion, such as mu-law to A-
   law conversion of PCM on an international call.

   Message Details


   F1 INVITE A -> SIP FW

   INVITE sip:UserB@there.com;maddr=fwp1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060



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   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="85b4f1cen4341ae6cbe5a3a9c8e88df9", opaque="",
    uri="sip:ss1.wcom.com", response="b3f392f9218a328b9294076d708e6815"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Client for A prepares to receive data on port 49172 from the
   network. */

   F2 INVITE SIP FW -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=fwp1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="85b4f1cen4341ae6cbe5a3a9c8e88df9", opaque="",
    uri="sip:ss1.wcom.com", response="b3f392f9218a328b9294076d708e6815"
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 200.201.202.203
   t=0 0
   m=audio 1000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000





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   F3 (100 Trying) SIP FW -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* SIP FW prepares to proxy data from port 200.201.202.203/1000 to
   100.101.102.103/49172.   Proxy 1 uses a Location Service function to
   determine where B is located. Based upon location analysis the call
   is forwarded to User B */

   F4 INVITE Proxy 1 -> B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>,
    <sip:UserB@there.com;maddr=fwp1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 200.201.202.203
   t=0 0
   m=audio 1000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 (100 Trying) Proxy 1 -> SIP FW

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0



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   F6 180 Ringing B -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 180 Ringing Proxy 1 -> SIP FW

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 180 Ringing SIP FW -> A

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>,
    <sip:UserB@there.com;maddr=fwp1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>



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   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F10 200 OK Proxy 1 -> SIP FW

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>,
    <sip:UserB@there.com;maddr=fwp1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 200 OK SIP FW -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>,
    <sip:UserB@there.com;maddr=fwp1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 147

   v=0



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   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 200.201.202.203
   t=0 0
   m=audio 1002 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* The Firewall Proxy prepares to proxy packets from 200.201.202.203/
      1002 to 110.111.112.113/3456 */

   F12 ACK A -> SIP FW

   ACK sip:UserB@there.com;maddr=fwp1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com;maddr=ss1.wcom.com>,
    <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F13 ACK SIP FW -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F14 ACK Proxy 1 -> B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between A and the Firewall Proxy and



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   between the Firewall Proxy and B*/

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

   F15 BYE A -> SIP FW

   BYE sip:UserB@fwp1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com;maddr=ss1.wcom.com>,
    <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F16 BYE SIP FW -> Proxy 1

   BYE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F17 BYE F18 Proxy 1 -> B

   BYE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F18 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159



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   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F19 200 OK Proxy 1 -> SIP FW

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP fwp1.wcom.com:5060;branch=9471385739578.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F20 200 OK SIP FW -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0




























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3.1.6 Successful SIP to SIP via Redirect and Proxy with SDP in ACK

   User A       Redirect Proxy      Proxy 2           User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     302 F2     |                |                |
     |<---------------|                |                |
     |     ACK F3     |                |                |
     |--------------->|                |                |
     |     INVITE F4                   |                |
     |-------------------------------->|    INVITE F5   |
     |            (100) F6             |--------------->|
     |<--------------------------------|     (100) F7   |
     |                                 |<---------------|
     |                                 |      180 F8    |
     |             180 F9              |<---------------|
     |<--------------------------------|                |
     |                                 |     200 F10    |
     |             200 F11             |<---------------|
     |<--------------------------------|                |
     |             ACK F12             |                |
     |-------------------------------->|     ACK F13    |
     |                                 |--------------->|
     |                Both Way RTP Media                |
     |<================================================>|
     |                                 |     BYE F14    |
     |             BYE F15             |<---------------|
     |<--------------------------------|                |
     |             200 F16             |                |
     |-------------------------------->|     200 F17    |
     |                                 |--------------->|
     |                                 |                |

   In this scenario, User A places a call to User B using first a
   Redirect server then a Proxy Server.  The INVITE message is first
   sent to the Redirect Server.  The Server returns a 302 Moved
   Temporarily response (F2) containing a Contact header with User B's
   current SIP address.  User A then generates a new INVITE and sends to
   User B via the Proxy Server and the call proceeds normally.  In this
   example, no SDP is present in the INVITE, so the SDP is carried in
   the ACK message.

   The call is terminated when User B sends a BYE message.


   Message Details


   F1 INVITE A -> Redirect Proxy




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   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Length: 0


   F2 302 Moved Temporarily Redirect Proxy -> A

   SIP/2.0 302 Moved Temporarily
   Contact: sip:UserB@everywhere.com
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F3 ACK A -> Redirect Proxy

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F4 INVITE A -> Proxy 2

   INVITE sip:UserB@everywhere.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Length: 0


   F5 INVITE Proxy 2 -> B

   INVITE sip:UserB@111.112.113.114 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@everywhere.com;maddr=ss2.wcom.com>



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   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Length: 0


   F6 (100 Trying) Proxy 2 -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   F7 (100 Trying) B -> Proxy 2

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   F8 180 Ringing B -> Proxy 2

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   F9 180 Ringing Proxy 2 -> A

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE



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   Content-Length: 0


   F10 200 OK B -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@everywhere.com;maddr=ss2.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserB@111.112.113.114>
   Content-Type: application/sdp
   Content-Length: 152

   v=0
   o=UserB 2890844527 2890844527 IN IP4 everywhere.com
   s=Session SDP
   c=IN IP4 111.112.113.114
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 200 OK Proxy -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserB@111.112.113.114>
   Content-Type: application/sdp
   Content-Length: 152

   v=0
   o=UserB 2890844527 2890844527 IN IP4 everywhere.com
   s=Session SDP
   c=IN IP4 111.112.113.114
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   /* ACK contains SDP of A since none present in INVITE */

   F12 ACK A -> Proxy 2




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   ACK sip:UserB@everyhere.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@111.112.113.114>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 ACK
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F13 ACK Proxy 2 -> B

   ACK sip:UserB@111.112.112.114 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 ACK
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

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

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

   F14 BYE B -> Proxy 2

   BYE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP everywhere.com:5060
   Route: <sip:UserA@100.101.102.103>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345600@here.com



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   CSeq: 1 BYE
   Content-Length: 0


   F15 BYE Proxy 2 -> A

   BYE sip:UserA@100.101.102.103 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP everywhere.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345600@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F16 200 OK A -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP everywhere.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345600@here.com
   CSeq: 1 BYE
   Content-Length: 0


   F17 200 OK Proxy 2 -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP everywhere.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345600@here.com
   CSeq: 1 BYE
   Content-Length: 0

















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3.1.7 Successful SIP to SIP with re-INVITE (T.38 Fax)


       IFT UA                 Proxy              IFTGW UA
           |   F1 INVITE        |                    |
           |------------------->|      F2 INVITE     |
           |                    |------------------->|
           |   F3  100 Trying   |                    |
           |<-------------------|   F4  100 Trying   |
           |                    |<-------------------|
           |                    |   F5  180 Ringing  |
           |   F6  180 Ringing  |<-------------------|
           |<-------------------|                    |
           |                    |   F7  200 OK       |
           |   F8  200 OK       |<-------------------|
           |<-------------------|                    |
           |   F9  ACK          |                    |
           |------------------->|   F10 ACK          |
           |                     ------------------->|
           |      Both Way RTP Media Established     |
           |<=======================================>|
           |                    |   F11 INVITE       |
           |    F12 INVITE      |<-------------------|
           |<-------------------|                    |
           |   F13 200 OK       |                    |
           |------------------->|    F14 200 OK      |
           |                    |------------------->|
           |                    |   F15  ACK         |
           |      F16  ACK      |<-------------------|
           |<-------------------|                    |
           |          T.38/UDPT Fax Flow Established |
           |<=======================================>|
           |                    |   F17 INVITE       |
           |     F18 INVITE     |<-------------------|
           |<-------------------|                    |
           |   F19 200 OK       |                    |
           |------------------->|     F20 200 OK     |
           |                    |------------------->|
           |                    |   F21  ACK         |
           |           F22  ACK |<-------------------|
           |<-------------------|                    |
           |          RTP Media Re-Established       |
           |<=======================================>|
           |   F23 BYE          |                    |
           |------------------->|       F24 BYE      |
           |                    |------------------->|
           |                    |   F25  200 OK      |
           |    F26  200 OK     |<-------------------|
           |<-------------------|                    |





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   This example shows a session whose media session changes twice during
   the session.  The initial PCM media session is established between
   the two User Agents in messages F1 through F10.  The called party
   then wishes to change the type of media session between them. (In
   this example, the called party detects a fax transmission and wishes
   to change the media session to a T.38 Fax over IP session).  The
   called party then sends a re-INVITE in message F11 which is accepted
   in the 200 OK of message F13.  After message F16, the original PCM
   media session is terminated and the new T.38 session established.
   After the fax transmission is over, the called party again wishes to
   switch the media session back to a PCM media session and initiates a
   re-INVITE sequence with message F17.


   Message Details

   F1 INVITE  IFT UA -> PROXY

   INVITE sip:+1-650-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Contact: <sip:+1-303-555-1111@ift.here.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 146

   v=0
   o=IFAXTERMINAL01 2890844527 2890844527 IN IP4 ift.here.com
   s=Session SDP
   c=IN IP4 iftmg.here.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 INVITE  PROXY -> IFTGW UA

   INVITE sip:+1-650-555-2222@iftgw.there.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d007.1
   Via: SIP/2.0/UDP ift.here.com:5060
   Record-Route: <sip:+1-650-555-2222@iftgw.there.com;
       maddr=ss1.wcom.com>
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Contact: <sip:+1-303-555-1111@ift.here.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 146



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   v=0
   o=IFAXTERMINAL01 2890844527 2890844527 IN IP4 ift.here.com
   s=Session SDP
   c=IN IP4 iftmg.here.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 (100 Trying) PROXY -> IFT UA

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Content-Length: 0


   F4 100 Trying IFTGW UA -> PROXY

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d007.1
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: sip:+1-650-555-2222@ss1.wcom.com;user=phone
   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Content-Length: 0


   F5 180 Ringing IFTGW UA -> PROXY

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d007.1
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Content-Length: 0


   F6 180 Ringing PROXY -> IFT UA

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617



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   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Content-Length: 0


   F7 200 OK   IFTGW UA -> PROXY

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d007.1
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Contact: <sip:+1-650-555-2222@iftgw.there.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=IFAXTERMINAL01 2890844527 2890844527 IN IP4 iftgw.there.com
   s=Session SDP
   c=IN IP4 iftmg.there.com
   t=0 0
   m=audio 12323 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 200 OK   PROXY -> IFT UA

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ift.here.com:5060
   Record-Route: <sip:+1-650-555-2222@iftgw.there.com;
                  maddr=ss1.wcom.com>
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 17 INVITE
   Contact: <sip:+1-650-555-2222@iftgw.there.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=IFAXTERMINAL01 2890844527 2890844527 IN IP4 iftgw.there.com
   s=Session SDP
   c=IN IP4 iftmg.there.com
   t=0 0
   m=audio 12323 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 ACK IFT UA -> PROXY



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   ACK sip:+1-650-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ift.here.com:5060
   Route: <sip:+1-650-555-2222@iftgw.there.com;maddr=ss1.wcom.com>
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 17 ACK
   Content-Length: 0


   F10 ACK PROXY -> IFTGW UA

   ACK sip:+1-650-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d007.1
   Via: SIP/2.0/UDP ift.here.com:5060
   Record-Route: <sip:+1-650-555-2222@iftgw.there.com;
    maddr=ss1.wcom.com>
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 17 ACK
   Content-Length: 0


   /* RTP streams are established. The CNG fax tone is sent in-band if
   it is present.  The receiving side IFT gateway DSP detects the
   Preamble. A new UDP port is open on IFTGW for T.38 IFP packets and
   the IFTGW signals the switch over to fax mode by send a re-INVITE
   with the same Call-ID and with the new UDP port in the SDP  */

   F11 INVITE  IFTGW UA -> PROXY

   INVITE sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Route: <sip:+1-650-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 1 INVITE
   Contact: <sip:+1-650-555-2222@iftgw.there.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 320

   v=0
   o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
   s=Session SDP
   c=IN IP4 iftmg.there.com
   t=0 0
   m=image 49172 udptl t38
   a=T38FaxVersion:0



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   a=T38maxBitRate:14400
   a=T38FaxFillBitRemoval:0
   a=T38FaxTranscodingMMR:0
   a=T38FaxTranscodingJBIG:0
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:72
   a=T38FaxMaxDatagram:316
   a=T38FaxUdpEC:t38UDPRedundancy



   F12 INVITE  PROXY -> IFT UA

   INVITE sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d008.1
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Record-Route: <sip:+1-650-555-2222@ss1.wcom.com;
                 maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 1 INVITE
   Contact: <sip:+1-650-555-2222@iftgw.there.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 320

   v=0
   o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
   s=Session SDP
   c=IN IP4 iftmg.there.com
   t=0 0
   m=image 49172 udptl t38
   a=T38FaxVersion:0
   a=T38maxBitRate:14400
   a=T38FaxFillBitRemoval:0
   a=T38FaxTranscodingMMR:0
   a=T38FaxTranscodingJBIG:0
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:72
   a=T38FaxMaxDatagram:316
   a=T38FaxUdpEC:t38UDPRedundancy


   F13 200 OK IFT UA -> PROXY

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d008.1
   Via: SIP/2.0/UDP iftgw.there.com:5060
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com



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   CSeq: 1 INVITE
   Contact: <sip:+1-303-555-1111@ift.here.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 320

   v=0
   o=faxgw1 2890846527 2890846527 IN IP4 ift.here.com
   s=Session SDP
   c=IN IP4 iftmg.here.com
   t=0 0
   m=image 15002 udptl t38
   a=T38FaxVersion:0
   a=T38maxBitRate:14400
   a=T38FaxFillBitRemoval:0
   a=T38FaxTranscodingMMR:0
   a=T38FaxTranscodingJBIG:0
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:72
   a=T38FaxMaxDatagram:316
   a=T38FaxUdpEC:t38UDPRedundancy


   F14 200 OK  PROXY -> IFT UA

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Record-Route: <sip:+1-650-555-2222@ss1.wcom.com;
                 maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 1 INVITE
   Contact: <sip:+1-303-555-1111@ift.here.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 320

   v=0
   o=faxgw1 2890846527 2890846527 IN IP4 ift.here.com
   s=Session SDP
   c=IN IP4 iftmg.here.com
   t=0 0
   m=image 15002 udptl t38
   a=T38FaxVersion:0
   a=T38maxBitRate:14400
   a=T38FaxFillBitRemoval:0
   a=T38FaxTranscodingMMR:0
   a=T38FaxTranscodingJBIG:0
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:72
   a=T38FaxMaxDatagram:316
   a=T38FaxUdpEC:t38UDPRedundancy



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   F15 ACK IFTGW UA -> PROXY

   ACK sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Route: <sip:+1-650-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 1 ACK
   Content-Length: 0


   F16 ACK  PROXY -> IFT UA

   ACK sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d008.1
   Via: SIP/2.0/UDP iftgw.there.com:5060
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* T.38 fax transmission established both ways which replaces the PCM
   audio session. */

   /* Then, the end of the fax transmission is detected on ingress side
   and sent to the egress side (IFTGW). IFTGW initiates the switch back
   to voice  communication */

   F17 INVITE  IFTGW UA -> PROXY

   INVITE sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Route: <sip:+1-650-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 2 INVITE
   Contact: <sip:+1-650-555-2222@iftgw.there.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 181

   v=0
   o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
   s=Session SDP
   c=IN IP4 iftmg.there.com
   t=0 0



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   m=audio 12323 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F18 INVITE  PROXY -> IFT UA

   INVITE sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d009.1
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Record-Route: <sip:+1-650-555-2222@ss1.wcom.com;
                 maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 2 INVITE
   Contact: <sip:+1-650-555-2222@iftgw.there.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 181

   v=0
   o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
   s=Session SDP
   c=IN IP4 iftmg.there.com
   t=0 0
   m=audio 12323 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F19 200 OK IFT UA -> PROXY

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d008.1
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Record-Route: <sip:+1-650-555-2222@ss1.wcom.com;
                 maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 2 INVITE
   Contact: <sip:+1-303-555-1111@ift.here.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=faxgw1 2890844527 2890844527 IN IP4 ift.here.com
   s=Session SDP
   c=IN IP4 iftmg.here.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000




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   F20 200 OK  PROXY -> IFTGW UA

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Record-Route: <sip:+1-650-555-2222@ss1.wcom.com;
                 maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 2 INVITE
   Contact: <sip:+1-303-555-1111@ift.here.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=faxgw1 2890844527 2890844527 IN IP4 ift.here.com
   s=Session SDP
   c=IN IP4 iftmg.here.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F21 ACK IFTGW UA -> PROXY

   ACK sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP iftgw.there.com:5060
   Route: <sip:+1-650-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 2 ACK
   Content-Length: 0


   F22 ACK  PROXY -> IFT UA

   ACK sip:+1-303-555-1111@ift.here.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d009.1
   Via: SIP/2.0/UDP iftgw.there.com:5060
   To: sip:+1-303-555-1111@ift.here.com;user=phone
   From: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 2 ACK
   Content-Length: 0


   F23 BYE IFT UA -> PROXY

   BYE sip:+1-650-555-2222@ss1.wcom.com SIP/2.0



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   Via: SIP/2.0/UDP ift.here.com:5060
   Route: <sip:+1-650-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 18 BYE
   Content-Length: 0


   F24 BYE PROXY -> IFTGW UA

   BYE sip:+1-650-555-2222@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d009.1
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 18 BYE
   Content-Length: 0


   F25 200 OK IFTGW UA -> PROXY

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060; branch=2d007.1
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 18 BYE
   Content-Type: application/sdp
   Content-Length: 0


   F26 200 OK  PROXY -> IFT UA

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ift.here.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: <sip:+1-650-555-2222@ss1.wcom.com;user=phone>;tag=5617
   Call-ID: 1717@ift.here.com
   CSeq: 18 BYE
   Content-Type: application/sdp
   Content-Length: 0










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   3.2 Failure Scenarios

3.2.1 Unsuccessful SIP to SIP no answer

   User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|   INVITE F2    |                |
     |    (100) F3    |--------------->|   INVITE F4    |
     |<---------------|    (100) F5    |--------------->|
     |                |<---------------|                |
     |                |                |      180 F6    |
     |                |     180 F7     |<---------------|
     |     180 F8     |<---------------|                |
     |<---------------|                |                |
     |                |                |                |
     |   CANCEL F9    |                |                |
     |--------------->|                |                |
     |     200 F10    |                |                |
     |<---------------|   CANCEL F11   |                |
     |                |--------------->|                |
     |                |     200 F12    |                |
     |                |<---------------|                |
     |                |                |   CANCEL F13   |
     |                |                |--------------->|
     |                |                |     200 F14    |
     |                |                |<---------------|
     |                |                |     487 F15    |
     |                |                |<---------------|
     |                |                |     ACK F16    |
     |                |     487 F17    |--------------->|
     |                |<---------------|                |
     |                |     ACK F18    |                |
     |     487 F19    |--------------->|                |
     |<---------------|                |                |
     |     ACK F20    |                |                |
     |--------------->|                |                |
     |                |                |                |

   In this scenario, User A gives up on the call before User B answers
   (sends a 200 OK response).  User A sends a CANCEL (F9) since no final
   response had been received from User B.  If a 200 OK to the INVITE
   had crossed with the CANCEL, User A would have sent an ACK then a BYE
   to User B in order to properly terminate the call.

   Note that the CANCEL message is acknowledged with a 200 OK on a hop
   by hop basis, rather than end to end.







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   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="ze7k1ee88df84f1cec431ae6cbe5a359", opaque="",
    uri="sip:ss1.wcom.com", response="b00b416324679d7e243f55708d44be7b"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49172 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0



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


   F3 (100 Trying) Proxy 1 -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F4 INVITE Proxy 2 -> B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 (100 Trying) Proxy 2 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0




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   F6 180 Ringing B -> Proxy 2

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 180 Ringing Proxy 2 -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 180 Ringing Proxy 1 -> A

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 CANCEL A -> Proxy 1

   CANCEL sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F10 200 OK Proxy 1 -> A

   SIP/2.0 200 OK



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   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F11 CANCEL Proxy 1 -> Proxy 2

   CANCEL sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F12 200 OK Proxy 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F13 CANCEL Proxy 2 -> B

   CANCEL sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F14 200 OK B -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0




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   F15 487 Request Cancelled B -> Proxy 2

   SIP/2.0 487 Request Cancelled
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F16 ACK Proxy 2 -> B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F17 487 Request Cancelled Proxy 2 -> Proxy 1

   SIP/2.0 487 Request Cancelled
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9876
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F18 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9876
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F19 487 Request Cancelled Proxy 1 -> A

   SIP/2.0 487 Request Cancelled



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   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=4321
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE


   F20 ACK A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=4321
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0






































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3.2.2 Unsuccessful SIP to SIP busy

   User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|   INVITE F2    |                |
     |    (100) F3    |--------------->|   INVITE F4    |
     |<---------------|    (100) F5    |--------------->|
     |                |<---------------|                |
     |                |                |      486 F6    |
     |                |                |<---------------|
     |                |                |     ACK F7     |
     |                |      486 F8    |--------------->|
     |                |<---------------|                |
     |                |      ACK F9    |                |
     |     486 F10    |--------------->|                |
     |<---------------|                |                |
     |     ACK F11    |                |                |
     |--------------->|                |                |
     |                |                |                |


   In this scenario, User B is busy and sends a 486 Busy Here response
   to User A's INVITE.  Note that the 4xx response is ACKed at each
   signaling leg.

   Message Details


   F1 INVITE User A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="dc3a5ab2530aa93112cf5904ba7d88fa", opaque="",
    uri="sip:ss1.wcom.com", response="702138b27d869ac8741e10ec643d55be"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0



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   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49172 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   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 here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F4 INVITE Proxy 2 -> User B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com



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   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 (100 Trying) Proxy 2 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 486 Busy Here User B -> Proxy 2

   SIP/2.0  486 Busy Here
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 ACK Proxy 2 -> User B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F8 486 Busy Here Proxy 2 -> Proxy 1



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   SIP/2.0  486 Busy Here
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=1293
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=1293
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F10 486 Busy Here Proxy 1 -> User A

   SIP/2.0  486 Busy Here
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=a6b4
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F11 ACK User A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=a6b4
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0












Johnston, et al.            Informational                    [Page 85]


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3.2.3 Unsuccessful SIP to SIP no response

   User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|   INVITE F2    |                |
     |    (100) F3    |--------------->|   INVITE F4    |
     |<---------------|    (100) F5    |--------------->|
     |                |<---------------|   INVITE F6    |
     |                |                |--------------->|
     |                |                |   INVITE F7    |
     |                |                |--------------->|
     |                |                |   INVITE F8    |
     |                |                |--------------->|
     |                |                |   INVITE F9    |
     |                |                |--------------->|
     |                |                |   INVITE F10   |
     |                |                |--------------->|
     |                |                |   INVITE F11   |
     |                |                |--------------->|
     |                |                |   CANCEL F12   |
     |                |     480 F13    |--------------->|
     |                |<---------------|                |
     |                |     ACK F14    |                |
     |     480 F15    |--------------->|                |
     |<---------------|                |                |
     |     ACK F16    |                |                |
     |--------------->|                |                |
     |                |                |                |

   In this example, there is no response from User B to User A's INVITE
   messages being re-transmitted by Proxy 2.  After the sixth re-
   transmission, Proxy 2 gives up and sends a CANCEL to User B and a 480
   No Response to User A.  Note that the CANCEL would also be
   retransmitted six times, as governed by SIP timer T1 as in Call Flow
   5.2.6.


   Message Details


   F1 INVITE User A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",



Johnston, et al.            Informational                    [Page 86]


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    realm="MCI WorldCom SIP",
    nonce="cf5904ba7d8dc3a5ab2530aa931128fa", opaque="",
    uri="sip:ss1.wcom.com", response="7afc04be7961f053c24f80e7dbaf888f"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49172 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   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 here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0




Johnston, et al.            Informational                    [Page 87]


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   F4 INVITE Proxy 2 -> User B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 (100 Trying) Proxy 2 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 INVITE Proxy 2 -> User B

   Resend of Message F4


   F7 INVITE Proxy 2 -> User B

   Resend of Message F4


   F8 INVITE Proxy 2 -> User B

   Resend of Message F4



Johnston, et al.            Informational                    [Page 88]


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   F9 INVITE Proxy 2 -> User B

   Resend of Message F4


   F10 INVITE Proxy 2 -> User B

   Resend of Message F4


   F11 INVITE Proxy 2 -> User B

   Resend of Message F4


   F12 CANCEL Proxy 2 -> User B

   CANCEL sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F13 480 No Response Proxy 2 -> Proxy 1

   SIP/2.0  480 No Response
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F14 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0





Johnston, et al.            Informational                    [Page 89]


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   F15 480 No Response Proxy 1 -> User A

   SIP/2.0  480 No Response
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F16 ACK User A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


































Johnston, et al.            Informational                    [Page 90]


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3.2.4 Unsuccessful SIP to SIP Temporarily Unavailable

   User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|   INVITE F2    |                |
     |    (100) F4    |--------------->|   INVITE F3    |
     |<---------------|    (100) F5    |--------------->|
     |                |<---------------|    (100) F6    |
     |                |                |<---------------|
     |                |                |      180 F7    |
     |                |     180 F8     |<---------------|
     |     180 F9     |<---------------|                |
     |<---------------|                |     480 F10    |
     |                |                |<---------------|
     |                |                |     ACK F11    |
     |                |     480 F12    |--------------->|
     |                |<---------------|                |
     |                |     ACK F13    |                |
     |     480 F14    |--------------->|                |
     |<---------------|                |                |
     |     ACK F15    |                |                |
     |--------------->|                |                |
     |                |                |                |


   In this scenario, User B initially sends a 180 Ringing response to
   User A, indicating that alerting is taking place.  However, then a
   480 Unavailable is then sent to User A.  This response is
   acknowledged then proxied back to User A.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="aa9311cf5904ba7d8dc3a5ab253028fa", opaque="",
    uri="sip:ss1.wcom.com", response="59a46a91bf1646562a4d486c84b399db"
   Content-Type: application/sdp
   Content-Length: 147




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   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49172 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 INVITE Proxy 2 -> B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147




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   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) Proxy 1 -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 (100 Trying) Proxy 2 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 (100 Trying) User B -> Proxy 2

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 180 Ringing B -> Proxy 2

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060



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   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 180 Ringing Proxy 2 -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 180 Ringing Proxy 1 -> A

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F10 480 Temporarily Unavailable B -> Proxy 2

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F11 ACK Proxy 2 -> B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060;branch=721e418c4.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK



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   Content-Length: 0


   F12 480 Temporarily Unavailable Proxy 2 -> Proxy 1

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F13 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=9
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F14 480 Temporarily Unavailable Proxy 1 -> A

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=489292845645245422
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F15 ACK A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=489292845645245422
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0








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   4  SIP to Gateway Dialing


   In the following scenarios, User A (BigGuy sip:UserA@here.com) is a
   SIP phone or other SIP-enabled device.  User B is reachable via the
   PSTN at global telephone number +1-972-555-2222. 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.  Note that User A still uses his/her SIP URL in the Contact
   header, since the call could be redirected back to the SIP network.

   There is a major difference in the call flows in this section.  In-
   band alerting (ringing tone, busy tone, recorded announcements, etc.)
   is present in the PSTN speech path after the receipt of the SS7
   Address Complete Message (ACM) which maps to the SIP 180 Ringing
   response.  In a SIP to SIP call, the media path is not established
   until the call is answered (200 OK sent).  In order for the SIP
   caller User A to hear this alerting, it is necessary that an early
   media path be established to perform this.  This is the purpose of
   the 183 Session Progress responses used throughout this document
   in place of the 180 Ringing.

4.1 Success Scenarios

   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 Proxy 1 and a Gateway.



















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4.1.1 Successful SIP to ISUP PSTN call

   User A          Proxy 1           NGW 1           User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |    (100) F2    |                |                |
     |<---------------|   INVITE F3    |                |
     |                |--------------->|                |
     |                |    (100) F4    |                |
     |                |<---------------|     IAM F5     |
     |                |                |--------------->|
     |                |                |     ACM F6     |
     |                |     183 F7     |<---------------|
     |     183 F8     |<---------------|                |
     |<---------------|                |                |
     |         One 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 +1-972-555-2222 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 URL.

   User A could use either their SIP address (sip:UserA@here.com) or SIP
   telephone number (sip:+1-314-555-1111@ss1.wcom.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 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



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   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.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
    uri="sip:ss1.wcom.com", response="ccdca50cb091d587421457305d097458c"
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   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 here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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.*/




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   F3 INVITE Proxy 1 -> NGW 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) NGW 1 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F6 ACM User B -> NGW 1

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber



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   End To End Method=none available
   Interworking=encountered
   End to End Information=none available
   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* 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 here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE



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   Contact: <sip:+1-972-555-2222@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 164

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:+1-972-555-2222@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 164

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F12 ACK A -> Proxy 1

   ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:+1-972-555-2222@ngw1.wcom.com;user=phone>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F13 ACK Proxy 1 -> NGW 1




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   ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between A and B (via NGW 1) */

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

   F14 BYE A -> Proxy 1

   BYE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:+1-972-555-2222@ngw1.wcom.com;user=phone>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F15 BYE Proxy 1 -> NGW 1

   BYE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE



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   Content-Length: 0


   F17 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F18 REL NGW 1 -> B

   REL
   CauseCode=16 Normal
   CodingStandard=CCITT


   F19 RLC B -> NGW 1

   RLC





























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4.1.2 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  |
     |                |    183 F8      |<---------------|
     |    183 F9      |<---------------|                |
     |<---------------|                |                |
     |         One Way RTP Media       |  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 an Enterprise
   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 URL.

   The host portion of the Request-URI in the INVITE F6 is used to
   identify the context (customer, trunk group, or line) in which the
   private number 444-3333 is valid.  Otherwise, this INVITE message



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   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 Enterprise
   Gateway that serves User C.  User C is identified by its extension
   (444-3333) in the Request-URI sent to GW 1.

   User A hears the ringing provided by the Gateway on the media path
   established after the 183 Session Progress response is received.
   Signaling between GW1 and PBX C is shown as ISDN.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:+1-918-555-3333@ss1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="qo0dc3a5ab22aa931904badfa1cf5j9h", opaque="",
    uri="sip:ss1.wcom.com", response="6c792f5c9fa360358b93c7fb826bf550"
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   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 here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0





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   F3 INVITE Proxy 1 -> GW 1

   INVITE sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-918-555-3333@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) GW -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   F5 SETUP GW 1 -> User C

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


   F6 CALL PROCeeding User C -> GW 1




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   Protocol discriminator=Q.931
   Call reference: Flag=1
   Message type=CALL PROC
   Channel identification=Exclusive B-channel


   F7 PROGress User C -> GW 1

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


   F8 183 Session Progress GW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   /* GW 1 will encode PSTN audio (ringing) to A in RTP path */

   F9 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Content-Length: 0


   F10 CONNect User C -> GW 1

   Protocol discriminator=Q.931
   Call reference: Flag=1
   Message type=CONN


   F11 CONNect ACK GW 1 -> User C




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   Protocol discriminator=Q.931
   Call reference: Flag=0
   Message type=CONN ACK


   F12 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-918-555-3333@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:444-3333@wcom.com>
   Content-Type: application/sdp
   Content-Length: 165

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   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 here.com:5060
   Record-Route: <sip:+1-918-555-3333@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: <sip:444-3333@wcom.com>
   Content-Type: application/sdp
   Content-Length: 165

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000





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   F14 ACK A -> Proxy 1

   ACK sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:444-3333@wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 ACK
   Content-Length: 0


   F15 ACK Proxy 1 -> GW 1

   ACK sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 ACK
   Content-Length: 0


   /* RTP streams are established between A and B (via GW 1) */

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

   F16 BYE A -> Proxy 1

   BYE sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:444-3333@wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 3 BYE
   Content-Length: 0


   F17 BYE Proxy 1 -> GW 1

   BYE sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159



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   Call-ID: 12345600@here.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 3 BYE
   Content-Length: 0


   F19 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 3 BYE
   Content-Length: 0


   F20 DISConnect GW 1 -> User C

   Protocol discriminator=Q.931
   Call reference: Flag=1
   Message type=DISC
   Cause=16 (Normal clearing)


   F21 RELease User C -> GW 1

   Protocol discriminator=Q.931
   Call reference: Flag=0
   Message type=REL


   F22 RELease COMplete GW 1 -> User C

   Protocol discriminator=Q.931
   Call reference: Flag=1
   Message type=REL COM




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4.1.3 Successful SIP to ISUP PSTN call with overflow

     User A        Proxy 1          NGW 1          NGW 2         User B
    |              |              |              |              |
    |  INVITE F1   |              |              |              |
    |------------->|              |              |              |
    |              |  INVITE F2   |              |              |
    |   (100) F3   |------------->|              |              |
    |<-------------|    503 F4    |              |              |
    |              |<-------------|              |              |
    |              |    ACK F5    |              |              |
    |              |------------->|              |              |
    |              |   INVITE F6                 |              |
    |              |---------------------------->|     IAM F7   |
    |              |                             |------------->|
    |              |                             |     ACM F8   |
    |              |            183 F9           |<-------------|
    |   183 F10    |<----------------------------|              |
    |<-------------|                             |              |
    |               One 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


   F1 INVITE A -> Proxy 1



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   INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="b59311c3ba05b401cf80b2a2c5ac51b0", opaque="",
    uri="sip:ss1.wcom.com", response="ba6ab44923fa2614b28e3e3957789ab0"
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


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

   F2 INVITE Proxy 1 -> NGW 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000





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   F3 (100 Trying) Proxy 1 -> User A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=123456789
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 ACK Proxy 1 -> NGW 1

   ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com>;user=phone>
    ;tag=123456789
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


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

   F6 INVITE Proxy 1 -> NGW 2

   INVITE sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>



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   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F8 ACM User B -> NGW 2

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber
   End To End Method=none available
   Interworking=encountered
   End to End Information=none available
   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   F9 183 Session Progress NGW 2 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


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



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   F10 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:+1-972-555-2222@ngw2.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 164

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   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 here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com



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   CSeq: 1 INVITE
   Contact: <sip:+1-972-555-2222@ngw2.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 164

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 ACK A -> Proxy 1

   ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:+1-972-555-2222@ngw2.wcom.com;user=phone>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F15 ACK Proxy 1 -> NGW 2

   ACK sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.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:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:+1-972-555-2222@ngw2.wcom.com;user=phone>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>



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    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F17 BYE Proxy 1 -> NGW 2

   BYE sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F19 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F20 REL NGW 2 -> B

   REL
   CauseCode=16 Normal
   CodingStandard=CCITT

   F21 RLC B -> NGW 2



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   RLC




















































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4.2  Failure Scenarios

   In these failure scenarios, the call does not complete.  In most
   cases, however, a media stream is still setup.  This is due to the
   fact that most 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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4.2.1 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     |<---------------|                |
     |<---------------|                |                |
     |         One 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 issues a CANCEL (F9) to terminate the call. (A BYE
   is not sent since no final response was ever received by User A.)


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone  SIP/2.0



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   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="01cf8311c3b0b2a2c5ac51bb59a05b40", opaque="",
    uri="sip:ss1.wcom.com", response="e178fbe430e6680a1690261af8831f40"
   Content-Type: application/sdp
   Content-Length: 140

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 (100 Trying) Proxy 1 -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 140



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   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) NGW 1 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F6 ACM User B -> NGW 1

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber
   End To End Method=none available
   Interworking=encountered
   End to End Information=none available
   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060



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   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 CANCEL A -> Proxy 1

   CANCEL sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F10 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F11 CANCEL Proxy 1 -> NGW 1

   CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0



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   F12 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F13 REL NGW 1 -> B

   REL
   CauseCode=16 Normal
   CodingStandard=CCITT


   F14 RLC B -> NGW 1

   RLC


   F15 487 Request Cancelled NGW 1 -> Proxy 1

   SIP/2.0 487 Request Cancelled
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F16 ACK Proxy 1 -> NGW 1

   ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F17 487 Request Cancelled Proxy 1 -> A



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   SIP/2.0 487 Request Cancelled
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=4f-3e-ff-23-09-43
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F18 ACK A -> Proxy 1

   ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=4f-3e-ff-23-09-43
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0

































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

   User A          Proxy 1           NGW 1           User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |    (100) F2    |                |                |
     |<---------------|   INVITE F3    |                |
     |                |--------------->|                |
     |                |    (100) F4    |                |
     |                |<---------------|     IAM F5     |
     |                |                |--------------->|
     |                |                |   REL(28) F6   |
     |                |                |<---------------|
     |                |                |     RLC F7     |
     |                |     484 F8     |--------------->|
     |                |<---------------|                |
     |                |     ACK F9     |                |
     |                |--------------->|                |
     |     484 F10    |                |                |
     |<---------------|                |                |
     |     ACK F11    |                |                |
     |--------------->|                |                |
     |                |                |                |

   User A calls PSTN User B through a Proxy Server Proxy 1 and a Network
   Gateway NGW 1.  However, User A does not provide enough digits for
   the call to be completed.  The call is rejected by the PSTN with a
   ANSI ISUP Release message REL containing a specific Cause value.
   This cause value (28) is mapped by the Gateway to a SIP 484 Address
   Incomplete response which is proxied back to User A.  For more
   details of ISUP cause value to SIP responses refer to [5].


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:+44-1234@ss1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",
    nonce="j1c3b0b01cf832da2c5ac51bb59a05b40", opaque="",
    uri="sip:ss1.wcom.com", response="a451358d46b55512863efe1dccaa2f42"
   Content-Type: application/sdp



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   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 (100 Trying) Proxy 1 -> A

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+44-1234@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) NGW 1 -> Proxy 1



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   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F6 REL User B -> NGW 1

   REL
   CauseValue=28 Address Incomplete
   CodingStandard=CCITT


   F7 RLC NGW 1 -> User B

   RLC


   /* Network Gateway maps CauseValue=28 to the SIP message 484 Address
   Incomplete */

   F8 484 Address Incomplete NGW 1 -> Proxy 1

   SIP/2.0 484 Address Incomplete
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 ACK Proxy 1 -> NGW 1

   ACK sip:+44-1234@ngw1.wcom.com;user=phone SIP/2.0



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   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F10 484 Address Incomplete Proxy 1 -> User A

   SIP/2.0 484 Address Incomplete
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>;tag=141593
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F11 ACK User A -> Proxy 1

   ACK sip:+44-1234@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+44-1234@ss1.wcom.com;user=phone>;tag=141593
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


























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4.2.3 Unsuccessful SIP to PSTN: ANM Timeout

   User A          Proxy 1           NGW 1           User 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:+1-972-555-2222@ss1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Proxy-Authorization:Digest username="UserA",
    realm="MCI WorldCom SIP",



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    nonce="da2c5ac51bb59a05j1c3b0b01cf832b40", opaque="",
    uri="sip:ss1.wcom.com", response="579cb9db184cdc25bf816f37cbc03c7d"
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


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

   F2 (100 Trying Proxy 1 -> A

   SIP/2.0  100 Trying
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F3 INVITE Proxy 1 -> NGW 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000



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   F4 (100 Trying) NGW 1 -> Proxy 1

   SIP/2.0  100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345600@here.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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F6 ACM User B -> NGW 1

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber
   End To End Method=none available
   Interworking=encountered
   End to End Information=none available
   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   F7 183 Session Progress NGW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0





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   F8 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* 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
   CauseCode=16 Normal
   CodingStandard=CCITT


   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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F12 ACK Proxy 1 -> NGW 1

   ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0




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   F13 480 Temporarily Unavailable F13 Proxy 1 -> User A

   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=415913
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F14 ACK User A -> Proxy 1

   ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com;user=phone>
   To: LittleGuy <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
    ;tag=415913
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0































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5  Gateway to SIP Dialing

5.1 Success Scenarios

   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.  Alternatively, the Gateway could
   redirect the call to an Announcement Server which would complete the
   call and play announcements or tones as directed by the Gateway.




































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5.1.1 Successful PSTN to SIP call

   User 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
   USI=Speech
   CPT=0 0
   C=Normal



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   CCI=Not Required


   F2 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


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

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000





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   F4 (100 Trying) User B -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 180 Ringing Proxy 1 -> NGW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 ACM NGW 1 -> User A

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber
   End To End Method=none available
   Interworking=encountered
   End to End Information=none available
   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   F8 200 OK User B -> Proxy 1




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   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   Contact: <sip:UserB@110.111.112.113>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   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.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F10 ACK NGW 1 -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com



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   CSeq: 1 ACK
   Content-Length: 0


   F11 ACK Proxy 1 -> User B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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
   CodingStandard=CCITT


   F14 RLC NGW 1 -> User A

   RLC


   F15 BYE NGW 1-> Proxy 1

   BYE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 2 BYE
   Content-Length: 0


   F16 BYE Proxy 1 -> User B




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   BYE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 2 BYE
   Content-Length: 0























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5.1.2 Successful PSTN to SIP call, Fast Answer

   User 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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F2 INVITE NGW 1 -> Proxy 1




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   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


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

   F3 INVITE Proxy 1 -> User B

   INVITE UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) Proxy 1 -> NGW 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>



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   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   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.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000





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   F7 ACK NGW 1 -> Proxy 1

   ACK UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F8 ACK Proxy 1 -> User B

   ACK UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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
   CodingStandard=CCITT


   F11 RLC NGW 1 -> User A

   RLC


   F12 BYE NGW 1 -> Proxy 1

   BYE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>



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   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 2 BYE
   Content-Length: 0


   F13 BYE Proxy 1 -> User B

   BYE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 2 BYE
   Content-Length: 0


   F15 200 OK Proxy 1 -> NGW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 2 BYE
   Content-Length: 0















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5.1.3 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.

   The Enterprise Gateway can only identify the trunk group that the
   call came in on, it cannot identify the individual line on PBX A that



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   is placing the call.  The SIP URL used to identify the caller is
   shown in these flows as sip:IdentifierString@gw1.wcom.com.  A unique
   IdentifierString is provisioned on the Gateway against each incoming
   trunk group.  Note: the string could be a telephone number.

   Message Details


   F1 MF Digits PBX A -> GW 1

   KP 1 972 555 2222 ST


   F2 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine where the
   phone number +1-972-555-2222 is located.  Based upon location
   analysis the call is forwarded to SIP User B. */

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150



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   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) Proxy 1 -> GW 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 180 Ringing Proxy 1 -> GW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 INVITE
   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.wcom.com:5060;branch=2d4790.1



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   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.com
   Contact: <sip:UserB@110.111.112.113>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   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
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:UserB@110.111.112.113>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 ACK GW 1 -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0



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   F10 ACK Proxy 1 -> User B

   ACK sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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. */

   F11 BYE GW 1 -> Proxy 1

   BYE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Route: <sip:UserB@110.111.112.113>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 2 BYE
   Content-Length: 0


   F12 BYE Proxy 1 -> User B

   BYE sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 2 BYE



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   Content-Length: 0


   F14 200 OK Proxy 1 -> GW 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 2 BYE
   Content-Length: 0










































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5.2 Failure Scenarios

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

   User 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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F2 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-9999@ss1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-9999@ss1.wcom.com;user=phone>
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@gw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0



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   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


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

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

   SIP/2.0 604 Does Not Exist Anywhere
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-9999@ss1.wcom.com;user=phone>;tag=6a34d410
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F4 ACK GW 1 -> Proxy 1

   ACK sip:+1-972-555-9999@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@gw1.wcom.com;user=phone>
   To: <sip:+1-972-555-9999@ss1.wcom.com;user=phone>;tag=6a34d410
   Call-ID: 12345602@gw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F5 REL GW 1 -> User A

   REL
   CauseCode=1
   CodingStandard=CCITT


   F6 RLC User A -> GW 1

   RLC











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

   User 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 [5].


   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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F2 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone  SIP/2.0



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   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +1-972-555-2222.  The call is then forwarded to User B. */

   F3 INVITE F3 Proxy 1 -> User B

   INVITE UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 (100 Trying) Proxy 1 -> NGW 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com



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   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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 ACK Proxy 1 -> User B

   ACK UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=59
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 ACK NGW 1 -> Proxy 1

   ACK UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=59
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F9 REL NGW 1 -> User A




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   REL
   CauseCode=17 Busy
   CodingStandard=CCITT


   F10 RLC User A -> NGW 1

   RLC














































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

   User A           NGW 1           Proxy 1           User B
     |                |                |                |
     |     IAM F1     |                |                |
     |--------------->|   INVITE F2    |                |
     |                |--------------->|   INVITE F3    |
     |                |    (100) F4    |--------------->|
     |                |<---------------|                |
     |                |                |      600 F5    |
     |                |                |<---------------|
     |                |                |      ACK F6    |
     |                |     600 F7     |--------------->|
     |                |<---------------|                |
     |                |     ACK F8     |                |
     |     ANM F9     |--------------->|                |
     |<---------------|                |                |
     | Both 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, plays busy tone, and releases call after timeout.

   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
   CdPN=972-555-2222,NPI=E.164,NOA=National
   USI=Speech



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   CPT=0 0
   C=Normal
   CCI=Not Required
   Interworking=encountered


   F2 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +1-972-555-2222.  The call is then forwarded to User B. */

   F3 INVITE Proxy 1 -> User B

   INVITE UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000



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   F4 (100 Trying) User B -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 ACK Proxy 1 -> User B

   ACK UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=66536336
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 ACK NGW 1 -> Proxy 1

   ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0



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   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=66536336
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F9 ACM NGW 1 -> User A

   ANM


   /* A two 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
   CodingStandard=CCITT


   F11 RLC NGW 1 -> User A

   RLC



























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5.2.4 Unsuccessful PSTN->SIP, ACM timeout

   User 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    |                |
     |                |<---------------|   CANCEL F14   |
     |                |                |--------------->|
     |                |                |   CANCEL F15   |
     |                |                |--------------->|
     |                |                |   CANCEL F16   |
     |                |                |--------------->|
     |                |                |   CANCEL F17   |
     |                |                |--------------->|
     |                |                |   CANCEL F18   |
     |                |                |--------------->|
     |                |                |                |

   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
   which is again re-sent by Proxy 1 after SIP T1 timer expires.

   Message Details

   F1 IAM User A -> NGW 1

   IAM



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   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required

   F2 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   /* Proxy 1 uses a Location Service function to determine a route for
   +1-972-555-2222.  The call is then forwarded to User B. */

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@there.com  SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   c c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000




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   F4 100 Trying Proxy 1 -> NGW 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   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
   CodingStandard=CCITT


   F11 RLC NGW 1 -> User A

   RLC





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   F12 CANCEL NGW 1 -> Proxy 1

   CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F14 CANCEL Proxy 1 -> User B

   CANCEL sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F15 CANCEL Proxy 1 -> User B

   Same as Message F14


   F16 CANCEL Proxy 1 -> User B

   Same as Message F14


   F17 CANCEL Proxy 1 -> User B

   Same as Message F14


   F18 CANCEL Proxy 1 -> User B

   Same as Message F14



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   F19 CANCEL Proxy 1 -> User B

   Same as Message F14


















































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

   User 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    |                |                |
     |<---------------|                |                |
     |                |   CANCEL F16   |                |
     |                |--------------->|   CANCEL F17   |
     |                |   CANCEL F18   |--------------->|
     |                |--------------->|   CANCEL F19   |
     |                |   CANCEL F20   |--------------->|
     |                |--------------->|   CANCEL F21   |
     |                |   CANCEL F22   |--------------->|
     |                |--------------->|   CANCEL F23   |
     |                |   CANCEL F24   |--------------->|
     |                |--------------->|   CANCEL F25   |
     |                |   CANCEL F26   |--------------->|
     |                |--------------->|   CANCEL F27   |
     |                |                |--------------->|
     |                |                |                |

   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 and CANCEL messages 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



Johnston, et al.            Informational                   [Page 168]


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   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=972-555-2222,NPI=E.164,NOA=National
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F2 INVITE NGW 1 -> Proxy 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +1-972-555-2222.  The call is then forwarded to User B. */

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@there.com  SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0



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   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


   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




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   Same as Message F3


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

   F14 REL User A -> NGW 1

   REL
   CauseCode=102 Timeout
   CodingStandard=CCITT


   F15 RLC NGW 1 -> User A

   RLC


   F16 CANCEL NGW 1 -> Proxy 1

   CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F17 CANCEL Proxy 1 -> User B

   CANCEL sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F18 CANCEL NGW 1 -> Proxy 1

   Same as Message F16


   F19 CANCEL Proxy 1 -> User B

   Same as Message F17


   F20 CANCEL NGW 1 -> Proxy 1



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   Same as Message F16


   F21 CANCEL Proxy 1 -> User B

   Same as Message F17


   F22 CANCEL NGW 1 -> Proxy 1

   Same as Message F16


   F23 CANCEL Proxy 1 -> User B

   Same as Message F17


   F24 CANCEL NGW 1 -> Proxy 1

   Same as Message F16


   F25 CANCEL Proxy 1 -> User B

   Same as Message F17


   F26 CANCEL NGW 1 -> Proxy 1

   Same as Message F16


   F27 CANCEL Proxy 1 -> User B

   Same as Message F17

















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5.2.6 Unsuccessful PSTN->SIP, ANM timeout

   User 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





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   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
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F2 INVITE A -> Proxy 1

   INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com
   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a Location Service function to determine a route for
   +1-972-555-2222.  The call is then forwarded to User B. */

   F3 INVITE Proxy 1 -> User B

   INVITE sip:UserB@there.com  SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 150

   v=0
   o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com



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   s=Session SDP
   c=IN IP4 gatewayone.wcom.com
   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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F6 180 Ringing Proxy 1 -> NGW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 ACM NGW 1 -> User A

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber
   End To End Method=none available
   Interworking=encountered
   End to End Information=none available



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   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   /* User A hangs up */

   F8 REL User A -> NGW 1

   REL
   CauseCode=16 Normal
   CodingStandard=CCITT


   F9 RLC NGW 1 -> User A

   RLC


   F10 CANCEL NGW 1 -> Proxy 1

   CANCEL sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.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.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F12 CANCEL Proxy 1 -> User B

   CANCEL sip:UserB@110.111.112.113 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 CANCEL
   Content-Length: 0




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   F13 200 OK User B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F14 487 Request Cancelled User B -> Proxy 1

   SIP/2.0 487 Request Cancelled
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F15 ACK Proxy 1 -> User B

   ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F16 487 Request Cancelled Proxy 1 -> NGW 1

   SIP/2.0 487 Request Cancelled
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=14159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F17 ACK NGW 1 -> Proxy 1

   ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone  SIP/2.0



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   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-972-555-2222@ss1.wcom.com;user=phone>;tag=14159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0
















































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6  Gateway to Gateway 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 [5].

6.1 Success Scenarios

   In these scenarios, the call is successfully completed between the
   two Gateways allowing the PSTN or PBX users to communicate.  The 183
   Session Progress response is used to indicate in-band alerting may
   flow from the called party telephone switch to the caller.



































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6.1.1 Successful ISUP PSTN to ISUP PSTN call

     User A         NGW 1         Proxy 1         GW 2          User C
    |              |              |              |              |
    |     IAM F1   |              |              |              |
    |------------->|              |              |              |
    |              |  INVITE F2   |              |              |
    |              |------------->|  INVITE F3   |              |
    |              |              |------------->|     IAM F4   |
    |              |              |              |------------->|
    |              |              |              |     ACM F5   |
    |              |              |   183 F6     |<-------------|
    |              |    183 F7    |<-------------|              |
    |    ACM F8    |<-------------|              |              |
    |<-------------|              |              |              |
    | One Way Voice|      One 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 Enterprise
   Gateway GW 2.  The CdPN and CgPN are mapped by GW1 into SIP URLs 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
   User C




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   served by GW 2.  The Request-URI 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 User A -> NGW 1

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=918-555-3333,NPI=E.164,NOA=National
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F2 INVITE NGW 1 -> Proxy 1

   INVITE sip:+1-918-555-3333@ss1.wcom.com;user=phone  SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=GW1 2890844526 2890844526 IN IP4 gw1.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   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:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1



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   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:+1-918-555-3333@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=GW1 2890844526 2890844526 IN IP4 gw1.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F4 IAM GW 2 -> User C

   IAM
   CgPN=314-555-1111,NPI=E.164,NOA=National
   CdPN=444-3333,NPI=Private,NOA=Subscriber
   USI=Speech
   CPT=0 0
   C=Normal
   CCI=Not Required


   F5 ACM User C -> GW 2

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber
   End To End Method=none available
   Interworking=encountered
   End to End Information=none available
   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   /* Based on PROGress 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.wcom.com:5060;branch=2d4790.1



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   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 183 Session Progress Proxy 1 -> GW 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* 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 -> User A

   ACM
   Charge Indicator=No Charge
   Called Party Status=no indication
   Called Party's Category=ordinary subscriber
   End To End Method=none available
   Interworking=encountered
   End to End Information=none available
   ISUP Indicator=not used all the way
   ISDN Access Terminating access non ISDN
   Echo Control=not included


   /* User B answers */

   F9 ANM User C -> GW 2

   ANM


   F10 200 OK GW 2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-918-555-3333@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>



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   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:444-3333@wcom.com>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.104
   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.wcom.com:5060
   Record-Route: <sip:+1-918-555-3333@ss1.wcom.com;maddr=ss1.wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:444-3333@wcom.com>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.104
   t=0 0
   m=audio 14918 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F12 ANM NGW 1 -> User A

   ANM


   F13 ACK NGW 1 -> Proxy 1

   ACK sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:444-3333@wcom.com>
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@gw1.wcom.com



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   CSeq: 1 ACK
   Content-Length: 0


   F14 ACK Proxy 1 -> GW 2

   ACK sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   From: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   To: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   Call-ID: 12345600@ngw1.wcom.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 User C -> GW 2

   REL
   CauseCode=16 Normal
   CodingStandard=CCITT


   F16 BYE GW 2 -> Proxy 1

   BYE sip:+1-314-555-1111@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP gw2.wcom.com:5060
   Route: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   From: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   To: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 4 BYE
   Content-Length: 0


   F17 RLC GW 2 -> User C

   RLC


   F18 BYE Proxy 1 -> NGW 1

   BYE sip:+1-314-555-1111@gw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw2.wcom.com:5060
   From: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   To: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>



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   Call-ID: 12345600@ngw1.wcom.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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw2.wcom.com:5060
   From: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   To: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Call-ID: 12345600@ngw1.wcom.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.wcom.com:5060
   From: <sip:+1-918-555-3333@ss1.wcom.com;user=phone>;tag=314159
   To: <sip:+1-314-555-1111@ngw1.wcom.com;user=phone>
   Call-ID: 12345600@ngw1.wcom.com
   CSeq: 4 BYE
   Content-Length: 0


   F21 REL User C -> GW 2

   REL
   CauseCode=16 Normal
   CodingStandard=CCITT


   F22 RLC GW 2 -> User C

   RLC
















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6.1.2 Successful FGB PBX to ISDN PBX call with overflow

   User A       GW 1        Proxy 1      GW 2         GW 3        User B
   |            |            |            |            |            |
   |  Seizure   |            |            |            |            |
   |----------->|            |            |            |            |
   |    Wink    |            |            |            |            |
   |<-----------|            |            |            |            |
   |MF Digits F1|            |            |            |            |
   |----------->|            |            |            |            |
   |            | INVITE F2  |            |            |            |
   |            |----------->| INVITE F3  |            |            |
   |            |            |----------->|            |            |
   |            |            |   503 F4   |            |            |
   |            |            |<-----------|            |            |
   |            |            |   ACK F5   |            |            |
   |            |            |----------->|            |            |
   |            |            |  INVITE F6              |            |
   |            |            |------------------------>|  SETUP F7  |
   |            |            |         (100) F8        |----------->|
   |            |            |<------------------------|CALL PROC F9|
   |            |            |                         |<-----------|
   |            |            |                         | ALERT F10  |
   |            |            |          183 F11        |<-----------|
   |            |  183 F12   |<------------------------|            |
   |            |<-----------|                         |            |
   |OneWay Voice|           One Way RTP Media          |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|
   |            |            |                         |<-----------|
   |            |            |                         |            |






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   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:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Content-Type: application/sdp
   Content-Length: 136

   v=0
   o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   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.  Response is returned listing alternative routes, GW2 and
   GW3, which are then tried sequentially. */



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   F3 INVITE Proxy 1 -> GW 2

   INVITE sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:444-3333@wcom.com;maddr=ss1.wcom.com>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Content-Type: application/sdp
   Content-Length: 136

   v=0
   o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   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.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=314159
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 ACK Proxy 1 -> GW 2

   ACK sip:444-3333@wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.1
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=314159
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F6 INVITE Proxy 1 -> GW 3

   INVITE sip:+1-918-555-3333@gw3.wcom.com;user=phone SIP/2.0



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   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.2
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:444-3333@wcom.com;maddr=ss1.wcom.com>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Content-Type: application/sdp
   Content-Length: 136

   v=0
   o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 SETUP GW 3 -> PBX C

   Protocol discriminator=Q.931
   Call reference: Flag=0
   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


   F8 (100 Trying) GW 3 -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 CALL PROCeeding PBX C -> GW 3

   Protocol discriminator=Q.931
   Call reference: Flag=1



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   Message type=CALL PROC
   Channel identification=Exclusive B-channel


   F10 ALERT PBX C -> GW 3

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


   /* Based on PROGress message, GW 3 returns a 183 response.  In-band
   call progress indications are then sent to the originator.  */

   F11 183 Session Progress GW 3 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.2
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F12 183 Session Progress Proxy 1 -> GW 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* GW 1 receives packets from GW 3 with encoded ringback, tones or
   other audio.  GW 1 decodes this and places it on the originating
   trunk. */

   F13 CONNect PBX C -> GW 3

   Protocol discriminator=Q.931
   Call reference: Flag=1
   Message type=CONN


   F14 200 OK GW 3 -> Proxy 1



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   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.2
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:444-3333@wcom.com;maddr=ss1.wcom.com>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-918-555-3333@gw3.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.104
   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.wcom.com:5060
   Record-Route: <sip:444-3333@wcom.com;maddr=ss1.wcom.com>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Contact: <sip:+1-918-555-3333@gw3.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: 149

   v=0
   o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com
   s=Session SDP
   c=IN IP4 100.101.102.104
   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




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   ACK sip:+1-918-555-3333@ss1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Route: <sip:+1-918-555-3333@gw3.wcom.com;user=phone>
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F17 ACK Proxy 1 -> GW 3

   ACK sip:+1-918-555-3333@gw3.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.2
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: <sip:444-3333@wcom.com>;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F18 CONNect ACK GW 3 -> PBX C

   Protocol discriminator=Q.931
   Call reference: Flag=0
   Message type=CONN ACK


   /* 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
   Call reference: Flag=1
   Message type=DISC
   Cause=16 (Normal clearing)


   F20 BYE GW 3 -> Proxy 1

   BYE sip:IdentifierString@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP gw3.wcom.com:5060
   Route: <sip:IdentifierString@gw1.wcom.com>
   From: <sip:444-3333@wcom.com>;tag=123456789
   To: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 BYE
   Content-Length: 0



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   F21 BYE Proxy 1 -> GW 1

   BYE sip:IdentifierString@gw1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.2
   Via: SIP/2.0/UDP gw3.wcom.com:5060
   From: <sip:444-3333@wcom.com>;tag=123456789
   To: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 BYE
   Content-Length: 0


   GW 1 -> PBX A

   Seizure removal


   F22 RELease GW 3 -> PBX C

   Protocol discriminator=Q.931
   Call reference: Flag=0
   Message type=REL


   F23 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2d4790.2
   Via: SIP/2.0/UDP gw3.wcom.com:5060
   From: <sip:444-3333@wcom.com>;tag=123456789
   To: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Call-ID: 12345600@gw1.wcom.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.wcom.com:5060
   From: <sip:444-3333@wcom.com>;tag=123456789
   To: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 BYE
   Content-Length: 0


   F25 RELease COMplete PBX C -> GW 3




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   Protocol discriminator=Q.931
   Call reference: Flag=1
   Message type=REL COM


   PBX A -> GW 1

   Seizure removal














































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7 SIP Test Messages

   The files in here are test messages for SIP servers to exercise
   various functions. They have been used in SIP "bakeoff"
   interoperablity events.  All messages shown here are valid, unless
   otherwise noted.  The correct behavior of servers and clients is also
   described.

7.1 INVITE Parser Torture Test Message

   This message is a correctly formatting SIP message. It contains:

   line folding all over
   escaped characters within quotes
   LWS between colons, semicolons, headers, and other fields
   both comma separated and separate listing of headers
   mix or short and long form for the same header
   unknown header field
   unusual header ordering
   nested comments
   unknown parameters of a known header

   Proxies should forward message and clients should respond as to a
   normal INVITE message.


   Message Details

   INVITE sip:vivekg@chair.dnrc.bell-labs.com SIP/2.0
   TO :
    sip:vivekg@chair.dnrc.bell-labs.com ;   tag    = 1918181833n
   From   : "J Rosenberg \\\"" <sip:jdrosen@lucent.com>
     ;
     tag = 98asjd8
   Call-ID
    : 0ha0isndaksdj@10.1.1.1
   cseq: 8
     INVITE
   Via  : SIP  /   2.0
    /UDP
       135.180.130.133
   Subject :
   NewFangledHeader:   newfangled value
    more newfangled value
   Content-Type: application/sdp
   v:  SIP  / 2.0  / TCP     12.3.4.5   ;
     branch  =   9ikj8  ,
    SIP  /    2.0   / UDP  1.2.3.4   ; hidden
   m:"Quoted string \"\"" <sip:jdrosen@bell-labs.com> ; newparam =
   newvalue ;
     secondparam = secondvalue  ; q = 0.33



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     (((nested comments) and (more)))   ,
    tel:4443322

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC


7.2 INVITE with Proxy-Require and Require

   This message tests support for Proxy-Require and Require. It is a
   request that contains both headers, listing new features.

   Proxies and clients should respond with a 420 Bad Extension, and an
   Unsupported header listing these features.

   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:j_user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.1.1.1
   Require: newfeature1, newfeature2
   Proxy-Require: newfeature3, newfeature4
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133


7.3 INVITE with Unknown Schemes in URIs and URLs

   This message contains unknown schemes in the Request URI, To, From
   and Contact headers of a request.

   A server should probably return a not found error; but other
   behaviors are acceptable.


   Message Details

   INVITE name:John_Smith SIP/2.0
   To: isbn:2983792873
   From: http://www.cs.columbia.edu
   Call-ID: 0ha0isndaksdj@10.1.2.3
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp

   v=0



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   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.4 REGISTER with Y2038 Test

   This message is a registration request with an expiration year of
   2040. This makes sure that a server doesn't crash on seeing a date
   past Y2038.

   The correct behavior is probably to limit the lifetime to some
   configured maximum.


   Message Details

   REGISTER sip:company.com SIP/2.0
   To: sip:user@company.com
   From: sip:user@company.com
   Contact: sip:user@host.company.com
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 REGISTER
   Via: SIP/2.0/UDP 135.180.130.133
   Expires: Sat, 01 Dec 2040 16:00:00 GMT



7.5 INVITE with inconsistent Accept and message body

   This is a UAS test. It is a request that includes an Accept header
   without SDP. The UAS should respond with an error.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:j_user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   Accept: text/newformat
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp

   v=0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12



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   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.6 INVITE with non-SDP message body

   This is a test of a user agent server. It is a request that includes
   a body of a non-SDP type.

   The user agent server should respond with an error.

   Message Details

   INVITE sip:user@comapny.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/newformat

   <audio>
    <pcmu port="443"/>
   </audio>



7.7 Unknown Method Message

   This request message contains a new unknown method, NEWMETHOD.

   A proxy should forward this using the same retransmission rules as
   BYE. A UAS should reject it with an error, and list the available
   methods in the response.


   Message Details

   NEWMETHOD sip:user@comapny.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 NEWMETHOD
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12



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   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.8 Unknown Method with CSeq Error

   This message is nearly identical to the Unknown Method message. It is
   a request with a new unknown method, but with a CSeq method tag which
   does not match.

   A proxy should either respond with an error, or correct the method
   tag. The user agent should reject it with an error, and list the
   available methods in the response.


   Message Details

   NEWMETHOD sip:user@comapny.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.1.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.9 REGISTER with Unknown Authorization Scheme

   This message is a REGISTER request with an unknown authorization
   scheme.


   The server should do something reasonable, such as rejecting the
   request.


   Message Details

   REGISTER sip:company.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:j.user@company.com
   Call-ID: 0ha0isndaksdj@10.0.1.1



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   CSeq: 8 REGISTER
   Via: SIP/2.0/UDP 135.180.130.133
   Authorization: Super-PGP ajsohdaosdh0asyhdaind08yasdknasd09asidhas0d8



7.10 Multiple SIP Request in a Single Message

   This message contains two requests, separated by a bunch of
   whitespace. Since the
   message exceeds the length indicated in the Content-Length header,
   the message should
   be rejected. (Multiple SIP requests per UDP packet are no longer
   allowed.)


   Message Details

   REGISTER sip:company.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:j.user@company.com
   Call-ID: 0ha0isndaksdj@10.0.2.2
   Contact: sip:j.user@host.company.com
   CSeq: 8 REGISTER
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Length: 0



   INVITE sip:joe@company.com SIP/2.0
   To: sip:joe@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isnda977644900765@10.0.0.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m =video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.11 INVITE missing Required Headers

   This message contains no Call-ID, From, or To header.

   The server should not crash, and ideally should respond with an



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   error.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   CSeq: 0 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.12 INVITE with Duplicate Required Headers

   The message contains a request with an extra Call-ID and To field.

   The server should not crash, and should ideally respond with an
   error.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   Via: SIP/2.0/UDP 135.180.130.133
   CSeq: 0 INVITE
   Call-ID: 98asdh@10.1.1.1
   Call-ID: 98asdh@10.1.1.2
   From: sip:caller@university.edu
   From: sip:caller@organization.org
   To: sip:user@company.com
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.13 INVITE with Illegal Expires Header

   This message contains an Expires header which has illegal values for



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   a number of components, but otherwise is syntactically correct.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   Via: SIP/2.0/UDP 135.180.130.133
   CSeq: 0 INVITE
   Call-ID: 98asdh@10.1.1.2
   Expires: Thu, 44 Dec 19999 16:00:00 EDT
   From: sip:caller@university.edu
   To: sip:user@company.com
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.14 200 OK Response with Broadcast Via Header

   This message is a response with a 2nd Via header of 255.255.255.255.

   On receiving this response, the top Via header is stripped and the
   packet forwarded.  Since the next address is the broadcast address,
   it causes the packet to be broadcast onto the network. A smart server
   should ignore packets with 2nd Via headers that are 255.255.255.255
   or 127.0.0.1. At the very least it should not crash.


   Message Details

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 135.180.130.57;branch=0
   Via: SIP/2.0/UDP 255.255.255.255;branch=0
   Call-ID: 0384840201@10.1.1.1
   CSeq: 0 INVITE
   From: sip:user@company.com
   To: sip:user@university.edu;tag=2229
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 224.2.17.12/127
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



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7.15 INVITE with Invalid Via and Contact Headers

   This is a request with the Via and Contact headers incorrect. They
   contain additional semicolons and commas without parameters or
   values.

   The server should respond with a Bad Request error.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133;;,;
   Contact: "" <> ;,"Joe" <sip:joe@org.org>;;,,;;
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.16 INVITE with Incorrect Content-Length Header

   This is a request message with a Content Length that is much larger
   than the length of the body.

   When sent UDP, the server should respond with an error. With TCP,
   there's not much you can do but wait...


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 9999



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   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.17 INVITE with Invalid Value for Content-Length

   This is a request message with a negative value for Content-Length.

   The server should respond with an error.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:j.user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: -999

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.18 INVITE with Garbage after Message Body


   This is a request message with garbage after the end of the SDP
   included in the body.

   The servers should reject the request as the body is longer than the
   Content-Length.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:j.user@company.com



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   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 138

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC
   asdpasd08asdsdk:;;asd
    a0sdjhg8a0''...'';;;;



7.19 INVITE with Error in Display Name in To Header

   This is a request with an unterminated quote in the display name of
   the To field.

   The server can either return an error, or proxy it if it is
   successful parsing without the terminating quote.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: "Mr. J. User <sip:j.user@company.com>
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.0.0.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 138

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.20 INVITE with a Semicolon-Separated Parameter in the "user" Part

   This is an INVITE request with a semicolon-separated parameter in
   the "user" part.



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   Outbound proxies should direct it appropriately.


   Message Details

   INVITE sip:user;par=u%40h.com@company.com SIP/2.0
   To: sip:j_user@company.com
   From: sip:caller@university.edu
   Call-ID: 0ha0isndaksdj@10.1.1.1
   CSeq: 8 INVITE
   Via: SIP/2.0/UDP 135.180.130.133



7.21 INVITE with Illegal Enclosing of Request-URI  in "<>"

   This INVITE is illegal because the Request-URI has been enclosed
   within in "<>".

   An intelligent server may be able to deal with this and fix up
   the Request-URI if acting as a Proxy. If not it should respond 400
   with an appropriate reason phrase.


   Message Details

   INVITE <sip:user@company.com> SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 1@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.22 INVITE with Illegal LWS within Elements of Request-URI

   This INVITE has illegal LWS within the SIP URL.




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   An intelligent server may be able to deal with this and fix up
   the Request-URI if acting as a Proxy. If not it should respond 400
   with an appropriate reason phrase.


   Message Details

   INVITE sip:user@company.com; transport=udp SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 2@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.23 INVITE with illegal >1 SP between elements of Request URI

   This INVITE has illegal >1 SP between elements of the Request-URI.

   An intelligent server may be able to deal with this and fix up
   the Request-URI if acting as a Proxy. If not it should respond 400
   with an appropriate reason phrase.


   Message Details

   INVITE sip:user@company.com  SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 3@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0



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   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.24 INVITE with a legal SIP URL containing escaped characters

   This INVITE is legal and has a Request-URI with a SIP URL containing
   escaped characters.


   Message Details

   INVITE sip:sip%3Auser%40example.com@company.com;other-param=summit
   SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 4@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.25 INVITE with the illegal use of escaped headers in Request-URI

   This INVITE is illegal as it the Request-URI contains a SIP URL
   containing
   escaped headers.

   An intelligent server may be liberal enough to accept this. A server
   acting as a proxy should remove the escaped header before processing.


   Message Details

   INVITE sip:user@company.com?Route=%3Csip:sip.example.com%3E SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu



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   Call-ID: 5@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.26 INVITE containing an unknown in the Request URI

   This INVITE contains an unknown URI scheme in the Request-URI.

   A server should reject this message with a 400 response plus an
   appropriate reason phrase despite being able to understand the
   To header as a SIP URL.


   Message Details

   INVITE name:user SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 6@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.27 OPTIONS with no LWS between display name and <

   This OPTIONS request is legal despite there being no LWS between



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   the display name and < in the From header.


   Message Details

   OPTIONS sip:user@company.com SIP/2.0
   To: sip:user@company.com
   From: "caller"<sip:caller@example.com>
   Call-ID: 1234abcd@10.0.0.1
   CSeq: 1 OPTIONS
   Via: SIP/2.0/UDP 135.180.130.133



7.28 OPTIONS with extran LWS between display name and <

   This OPTIONS request is legal despite there being extra LWS between
   the display name and < in the From header.


   Message Details

   OPTIONS sip:user@company.com SIP/2.0
   To: sip:user@company.com
   From: "caller"    <sip:caller@example.com>
   Call-ID: 1234abcd@10.0.0.1
   CSeq: 2 OPTIONS
   Via: SIP/2.0/UDP 135.180.130.133



7.29 INVITE with an illegal SIP Date format.

   This INVITE is illegal as it contains a non GMT time zone in the SIP
   Date
   of the Expires header.

   An intelligent server may be able to fix this up and correct the time
   to
   GMT. Alternatively this message may illicit a 400 response with an
   appropriate
   reason phrase.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 7@10.0.0.1
   CSeq: 1 INVITE



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   Via: SIP/2.0/UDP 135.180.130.133
   Expires: Fri, 01 Jan 2010 16:00:00 EST
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.30 INVITE with Passed Expries Time

   This is a legal INVITE but the message content has long since
   expired.

   A server should respond 408 (Timeout).


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 8@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Expires: Thu, 01 Dec 1994 16:00:00 GMT
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.31 INVITE with Max-Forwards Set to Zero

   This is a legal SIP request with the Max-Forwards header set to zero.




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   A proxy or gateway should not forward the request and respond 483
   (Too Many
   Hops).


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: sip:user@company.com
   From: sip:caller@university.edu
   Call-ID: 9@10.0.0.1
   CSeq: 1 INVITE
   Via: SIP/2.0/UDP 135.180.130.133
   Max-Forwards: 0
   Content-Type: application/sdp
   Content-Length: 174

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.32 REGISTER with a Escaped Header in a Legal SIP URL of a Contact


   This is a legal REGISTER message where the Contact header contains a
   SIP URL with an escaped header within it.


   Message Details

   REGISTER sip:company.com SIP/2.0
   To: sip:user@company.com
   From: sip:user@company.com
   Contact: sip:user@host.company.com
   Call-ID: k345asrl3fdbv@10.0.0.1
   CSeq: 1 REGISTER
   Via: SIP/2.0/UDP 135.180.130.133
   Contact: <sip:user@example.com?Route=%3Csip:sip.example.com%3E>



7.33 REGISTER with a Escaped Header in a Illegal SIP URL of a Contact

   This is an illegal message as the REGISTER request contains a SIP



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   URL with an escaped header but it is not enclosed in  <>

   A server should respond 400 with an appropriate reason phrase.


   Message Details

   REGISTER sip:company.com SIP/2.0
   To: sip:user@company.com
   From: sip:user@company.com
   Contact: sip:user@host.company.com
   Call-ID: k345asrl3fdbv@10.0.0.1
   CSeq: 1 REGISTER
   Via: SIP/2.0/UDP 135.180.130.133
   Contact: sip:user@example.com?Route=%3Csip:sip.example.com%3E



7.34 INVITE with Long Values in Headers

   This is a legal message that contains long values in many headers.


   Message Details

   INVITE sip:user@company.com SIP/2.0
   To: "I have a user name of extreme proportion"
   <sip:user@company.com:6000;other-
   param=1234567890somethingelselong1234567890>
   From: sip:caller@university.edu
   Call-ID:
   kl24ahsd546folnyt2vbak9sad98u23naodiunzds09a3bqw0sdfbsk34poouymnae004
   3nsed09mfkvc74bd0cuwnms05dknw87hjpobd76f
   CSeq: 1 INVITE
   My-State: sldkjflzdsfaret0803adgaasd0afds0asdaasd
   Via: SIP/2.0/UDP sip33.example.com
   Via: SIP/2.0/UDP sip32.example.com
   Via: SIP/2.0/UDP sip31.example.com
   Via: SIP/2.0/UDP sip30.example.com
   Via: SIP/2.0/UDP sip29.example.com
   Via: SIP/2.0/UDP sip28.example.com
   Via: SIP/2.0/UDP sip27.example.com
   Via: SIP/2.0/UDP sip26.example.com
   Via: SIP/2.0/UDP sip25.example.com
   Via: SIP/2.0/UDP sip24.example.com
   Via: SIP/2.0/UDP sip23.example.com
   Via: SIP/2.0/UDP sip22.example.com
   Via: SIP/2.0/UDP sip21.example.com
   Via: SIP/2.0/UDP sip20.example.com
   Via: SIP/2.0/UDP sip19.example.com
   Via: SIP/2.0/UDP sip18.example.com



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   Via: SIP/2.0/UDP sip17.example.com
   Via: SIP/2.0/UDP sip16.example.com
   Via: SIP/2.0/UDP sip15.example.com
   Via: SIP/2.0/UDP sip14.example.com
   Via: SIP/2.0/UDP sip13.example.com
   Via: SIP/2.0/UDP sip12.example.com
   Via: SIP/2.0/UDP sip11.example.com
   Via: SIP/2.0/UDP sip10.example.com
   Via: SIP/2.0/UDP sip9.example.com
   Via: SIP/2.0/UDP sip8.example.com
   Via: SIP/2.0/UDP sip7.example.com
   Via: SIP/2.0/UDP sip6.example.com
   Via: SIP/2.0/UDP sip5.example.com
   Via: SIP/2.0/UDP sip4.example.com
   Via: SIP/2.0/UDP sip3.example.com
   Via: SIP/2.0/UDP sip2.example.com
   Via: SIP/2.0/UDP sip1.example.com
   Via: SIP/2.0/UDP
   host.example.com;received=135.180.130.133;branch=C1C3344E2710000000E2
   99E568E7potato10potato0potato0
   Content-Type: application/sdp

   v=0
   o=mhandley 29739 7272939 IN IP4 126.5.4.3
   s=SIP Call
   t=3149328700 0
   c=IN IP4 135.180.130.88
   m=audio 492170 RTP/AVP 0 12
   m=video 3227 RTP/AVP 31
   a=rtpmap:31 LPC



7.35 OPTIONS with multiple headers.

   This is an illegal and badly mangled message.

   A server should respond 400 with an appropriate reason phrase if it
   can. It may just drop this message.


   Message Details

   OPTIONS sip:135.180.130.133 SIP/2.0
   Via: SIP/2.0/UDP company.com:5604
   From: sip:iuser@company.com
   To: sip:user@135.180.130.133
   Call-ID: 1804928587@company.com
   CSeq: 1 OPTIONS
   Expires: 0 0l@company.com
   To: sip:user@135.180.130.133



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   Call-ID: 1804928587@company.com
   CSeq: 1 OPTIONS
   Contact: sip:host.company.com
   Expires: 0xpires: 0sip:host.company.com
   Expires: 0
   Contact: sip:host.company.com



7.36 INVITE with large number of SDP attributes and telephone subscriber
Request-URI

   This is a legal message with a large number of SDP attributes and a
   long telephone subscriber Request-URI


   Message Details

   INVITE sip:+1-972-555-2222;phone-
   context=name%40domain;new=user?%22Route%3a%20X%40Y%3bZ=W%22@gw1.wcom.
   com;user=phone SIP/2.0
   Via: SIP/2.0/UDP iftgw.there.com:5060
   From: sip:+1-303-555-1111@ift.here.com;user=phone
   To: sip:+1-650-555-2222@ss1.wcom.com;user=phone
   Call-ID: 1717@ift.here.com
   CSeq: 56 INVITE
   Content-Type: application/sdp
   Content-Length: 320

   v=0
   o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
   s=Session SDP
   c=IN IP4 iftmg.there.com
   t=0 0
   m=image 49172 udptl t38
   a=T38FaxVersion:0
   a=T38maxBitRate:14400
   a=T38FaxFillBitRemoval:0
   a=T38FaxTranscodingMMR:0
   a=T38FaxTranscodingJBIG:0
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:260
   a=T38FaxUdpEC:t38UDPR



7.37 REGISTER with a contact parameter.

   This REGISTER contains a contact where the 'user' parameter should be
   interpreted
   as being a contact-param and not a url-param.



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   The register should succeed but a subsequent retrieval of the
   registration must not
   include "user=phone" as a url-parameter.

   Message Details

   REGISTER sip:bell-tel.com SIP/2.0
   Via: SIP/2.0/UDP saturn.bell-tel.com
   From: sip:watson@bell-tel.com
   To: sip:watson@bell-tel.com
   Call-ID: 70710@saturn.bell-tel.com
   CSeq: 2 REGISTER
   Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone



7.38 REGISTER with a url parameter.

   This register contains a contact where the 'user'parameter is a url-
   param.

   The register should succeed and a subsequent retrieval of the
   registration must
   include "user=phone" as a url-parameter.

   Message Details

   REGISTER sip:bell-tel.com SIP/2.0
   Via: SIP/2.0/UDP saturn.bell-tel.com
   From: sip:watson@bell-tel.com
   To: sip:watson@bell-tel.com
   Call-ID: 70710@saturn.bell-tel.com
   CSeq: 3 REGISTER
   Contact: <sip:+1-972-555-2222@gw1.wcom.com;user=phone>



7.39 INVITE with an Unquoted Display Name Containing Multiple Tokens

   This is a legal INVITE where the To and From header contain display
   names
   that contain multiple tokens but are unquoted,


   Message Details

   INVITE sip:t.watson@ieee.org SIP/2.0
   Via:     SIP/2.0/UDP c.bell-tel.com
   From:    A. Bell <sip:a.g.bell@bell-tel.com>
   To:      T. Watson <sip:t.watson@ieee.org>



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   Call-ID: 31414@c.bell-tel.com
   CSeq:    1 INVITE



7.40 INVITE with an Unquoted Display Name Containg Non-Token Characters

   This is an illegal invite at the display names in the To and From
   headers
   contain non-token characters but are unquoted.

   A server may be intelligent enough to cope with this but may also
   return a
   400 response with an appropriate reason phrase.


   Message Details

   INVITE sip:t.watson@ieee.org SIP/2.0
   Via:     SIP/2.0/UDP c.bell-tel.com
   From:    Bell, Alexander <sip:a.g.bell@bell-tel.com>
   To:      Watson, Thomas <sip:t.watson@ieee.org>
   Call-ID: 31415@c.bell-tel.com
   CSeq:    1 INVITE



7.41 INVITE with Unknown (Higher) Protocol Version in Start Line

   This is an illegal INVITE as the SIP Protocol version is unknown.

   The server should respond to the request with a bad version error.


   Message Details

   INVITE sip:t.watson@ieee.org SIP/7.0
   Via:     SIP/2.0/UDP c.bell-tel.com
   From:    A. Bell <sip:a.g.bell@bell-tel.com>
   To:      T. Watson <sip:t.watson@ieee.org>
   Call-ID: 31417@c.bell-tel.com
   CSeq:    1 INVITE





8  Acknowledgements






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

   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.


9 References


   [1] S. Bradner, "The Internet Standards Process -- Revision 3", BCP
    9, RFC 2026, October 1996.

   [2] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
    Session Initiation Protocol", RFC 2543, March 1999.

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

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

   [5] G. Camarillo, "Best Current Practice for ISUP to SIP Mapping",
    Internet Draft, Internet Engineering Task Force,
    Work in progress.


Author's Addresses

   Alan Johnston               Email: alan.johnston@wcom.com
                               sip:alan.johnston@siptest.wcom.com

   Steve Donovan               Email: sdonovan@dynamicsoft.com

   Robert Sparks               Email: rsparks@dynamicsoft.com

   Chris Cunningham            Email: ccunningham@dynamicsoft.com

   Dean Willis                 Email: dwillis@dynamicsoft.com



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   Jonathan Rosenberg          Email: jdrosen@dynamicsoft.com

   Kevin Summers               Email: kevin.summers@sonusnet.com

   Henning Schulzrinne         Email: schulzrinne@cs.columbia.edu



   Copyright Notice

   "Copyright (C) The Internet Society 2001. 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.


















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