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Versions: (draft-johnston-sip-call-flows) 00 01 02 03 04 05

Internet Engineering Task Force                          Alan Johnston
Internet Draft                                           Steve Donovan
Document: draft-ietf-sip-call-flows-00.txt               Robert Sparks
Category: Informational                               Chris Cunningham
March 2000                                               Kevin Summers
Expires: September 2000                                    Dean Willis
                                                          MCI WorldCom
                                                    Jonathan Rosenberg
                                                           dynamicsoft
                                                   Henning Schulzrinne
                                                   Columbia University

                    SIP Telephony 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 document gives examples of SIP (Session Initiation Protocol)
   call flows for IP telephony and service examples.  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).  Telephony service examples
   illustrated with call flows include call hold, transfer, forwarding,
   screening, and Find-Me.  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...................................................3
   1.1    General Assumptions........................................4
   1.2    Legend for Message Flows...................................6
   1.3    SIP Protocol Assumptions...................................6
   2      SIP Registration Services..................................8
   2.1    Success Scenarios..........................................8
    2.1.1 SIP Client New Registration................................8
    2.1.2 User updates contact list.................................11
    2.1.3 User Requests Current Contact List........................13
    2.1.4 User Cancels Registration.................................15
   2.2    Failure Scenarios.........................................17
    2.2.1 Unsuccessful SIP registration.............................17
   3      SIP to SIP Dialing........................................19
   3.1    Success Scenarios.........................................19
    3.1.1 Successful Simple SIP to SIP..............................20
    3.1.2 Successful SIP to SIP through two proxies.................23
    3.1.3 Successful SIP to SIP with Proxy failure..................32
    3.1.4 Successful SIP to SIP through SIP Firewall Proxy..........40
    3.1.5 Successful SIP to SIP via Redirect and Proxy..............48
   3.2    Failure Scenarios.........................................55
    3.2.1 Unsuccessful SIP to SIP no answer.........................55
    3.2.2 Unsuccessful SIP to SIP busy..............................60
    3.2.3 Unsuccessful SIP to SIP no response.......................64
    3.2.4 Unsuccessful SIP to SIP Temporarily Unavailable...........69
   4      SIP to Gateway Dialing....................................75
   4.1    Success Scenarios.........................................75
    4.1.1 Successful SIP to ISUP PSTN call..........................76
    4.1.2 Successful SIP to ISDN PBX call...........................84
    4.1.3 Successful SIP to ISUP PSTN call with overflow............95
   4.2    Failure Scenarios........................................103
    4.2.1 Unsuccessful SIP to PSTN call: Treatment from PSTN.......104
    4.2.2 Unsuccessful SIP to PSTN: REL w/Cause from PSTN..........109
    4.2.3 Unsuccessful SIP to PSTN: ANM Timeout....................113
   5      Gateway to SIP Dialing...................................118
   5.1    Success Scenarios........................................118
    5.1.1 Successful PSTN to SIP call..............................119
    5.1.2 Successful PSTN to SIP call, Fast Answer.................125
    5.1.3 Successful PBX to SIP call...............................130
   5.2    Failure Scenarios........................................136
    5.2.1 Unsuccessful PSTN to SIP REL, SIP error mapped to REL....136
    5.2.2 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL.....138
    5.2.3 Unsuccessful PSTN->SIP, SIP error interworking to tones..142
    5.2.4 Unsuccessful PSTN->SIP, ACM timeout......................146
    5.2.5 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy.....151
    5.2.6 Unsuccessful PSTN->SIP, ANM timeout......................156
   6      Gateway to Gateway Dialing via SIP Network...............161
   6.1    Success Scenarios........................................161
    6.1.1 Successful ISUP PSTN to ISUP PSTN call...................162



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    6.1.2 Successful FGB PBX to ISDN PBX call with overflow........170
   7      IP Telephony Services Features Call Flows................179
   7.1    Call Hold................................................179
   7.2    Consultation Hold........................................190
   7.3    Unattended Transfer......................................205
   7.4    Attended Transfer........................................219
   7.5    Call Forwarding Unconditional............................220
   7.6    Call Forwarding - Busy...................................225
   7.7    Call Forwarding - No Answer..............................231
   7.8    3-way Conference.........................................238
   7.9    Single Line Extension....................................238
   7.10   Find-Me..................................................239
   7.11   Call Management (Incoming Call Screening)................248
   7.12   Call Management (Outgoing Call Screening)................251
   8      SIP Test Messages........................................254
   8.1    INVITE Parser Torture Test Message.......................254
   8.2    INVITE with Proxy-Require and Require....................255
   8.3    INVITE with Unknown Schemes in URIs and URLs.............255
   8.4    REGISTER with Y2038 Test.................................256
   8.5    INVITE with inconsistent Accept and message body.........256
   8.6    INVITE with non-SDP message body.........................257
   8.7    Unknown Method Message...................................257
   8.8    Unknown Method with CSeq Error...........................258
   8.9    REGISTER with Unknown Authorization Scheme...............258
   8.10   INVITE with Invalid SIP Version Number...................259
   8.11   INVITE missing Required Headers..........................260
   8.12   INVITE with Duplicate Required Headers...................260
   8.13   INVITE with Illegal Expires Header.......................261
   8.14   200 OK Response with Broadcast Via Header................261
   8.15   INVITE with Invalid Via and Contact Headers..............262
   8.16   INVITE with Incorrect Content-Length Header..............263
   8.17   INVITE with Invalid Value for Content-Length.............263
   8.18   INVITE with Garbage after Message Body...................264
   8.19   INVITE with Error in Display Name in To Header...........264
   9      Acknowledgements.........................................265
   10     References...............................................265

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



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

   These call flows are based on the current version 2.0 of SIP in
   RFC2543[2].  Additions and changes to SIP necessary for PSTN
   interworking are referenced as IETF Internet-Drafts as they are used
   in the call flows.

   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.

   An example set of telephony features is also illustrated with call
   flows.  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.  They are outlined in this section
   so that they may be taken into consideration.  Differences in these
   assumptions will affect the nature of the call flows.

   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.
     . The Proxy and the Gateway will authenticate each other using
       IPSec[4].



   The SIP Proxy Server has access to a Location Manager 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.



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   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.


     . 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@gw.wcom.com;phone-context=p1234) which will require
       interpretation based on From header.  The "phone-context"
       qualifier is used to interpret the private number.  It is used



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       the same as the tag of the same name from the tel URL draft[5].
       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.


     . 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[5].  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).  For example, a SIP URL can
       be used in a Contact header, but a tel URL can not.

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

   Some of the Service Examples use SIP call control extensions[10].

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. */

1.3 SIP Protocol Assumptions

   Except for the following, this call flows document uses the April
   1999 version 2.0 of SIP defined by RFC2543[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.



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     . The 183 Session Progress response message[5] is used in SIP to
       Gateway and Gateway to Gateway via SIP calling (Sections 4 and
       6). The 183 response with SDP will cause the User Agent to
       immediately play the SDP media stream to hear in-band call
       progress information.  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 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.





































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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 the SIP proxy and/or 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.  All these call
   flows assume that registration requires authentication, otherwise 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
   RFC2543[2].



   Message Details


   F1 REGISTER B -> SIP Server




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   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@here.com
   CSeq: 1 REGISTER
   Contact: LittleGuy <sip:UserB@there.com>
   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@here.com
   CSeq: 1 REGISTER
   WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="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: 123456790@here.com
   CSeq: 1 REGISTER
   Contact: LittleGuy <sip:UserB@there.com>
   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: 1234567890@here.com



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   CSeq: 1 REGISTER
   Contact: LittleGuy <sip:UserB@there.com>
   Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
   Contact: tel:+1-972-555-2222
   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: 123456791@here.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>



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   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 1234567891@here.com
   CSeq: 1 REGISTER
   Contact: LittleGuy <sip:UserB@there.com>
   Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
   Contact: tel:+1-972-555-2222
   Contact: mailto:UserB@there.com
   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.  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.

   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: 123456792@here.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: 1234567892@here.com
   CSeq: 1 REGISTER
   Contact: LittleGuy <sip:UserB@there.com>
   Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone



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   Contact: tel:+1-972-555-2222
   Contact: mailto:UserB@there.com
   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 her/his registration with the SIP
   registrar/redirect 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: 123456791@here.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: 1234567891@here.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@here.com
   CSeq: 1 REGISTER
   Contact: LittleGuy <sip:UserB@there.com>
   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@here.com
   CSeq: 1 REGISTER
   WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com",
   nonce="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: 123456791@here.com
   CSeq: 1 REGISTER
   Contact: LittleGuy <sip:UserB@there.com>
   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 _IForgotIt_ */

   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: 1234567891@here.com
   CSeq: 1 REGISTER
   WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="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 Agent Clients (UACs)
   - 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.  Calls route using at least one SIP Proxy server.
   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 the first Proxy Server 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.





































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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 A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   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 49170 RTP/AVP 0



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   a=rtpmap:0 PCMU/8000


   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>
   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>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   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@ss1.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: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0


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

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

   F6 BYE User B -> User A

   BYE sip: UserA@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   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>
   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




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   2 also inserts itself into the Record-Route header.  The ACK (F15)
   and BYE (F18) both have a Route header.

   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@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   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 49170 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="wcom.com", nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359",
   opaque="", stale="FALSE", algorithm="MD5"
   Content-Length: 0


   F3 ACK A -> Proxy 1

   ACK sip:UserB@ss1.wcom.com SIP/2.0



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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 INVITE
   Content-Length: 0


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

   F4 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   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 49170 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 49170 from the
   network. */

   F5 INVITE Proxy 1 -> Proxy 2

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



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   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 (100 Trying) Proxy 2 -> Proxy 1




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   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F9 180 Ringing B -> Proxy 2

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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
   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



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   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   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@ss2.wcom.com>,<sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159



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   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   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@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@there.com>
   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@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com>
   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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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



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   Content-Length: 0


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

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

   F18 BYE User B -> Proxy 2

   BYE sip: UserA@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@ss1.wcom.com>,<sip:UserA@here.com>
   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@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   Route:  <sip:UserA@here.com>
   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@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   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
   Via: SIP/2.0/UDP ss2.wcom.com:5060



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   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
   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
     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@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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



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   Same as Message F1


   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.  If
   any 200 OK responses come back to the INVITE, User A sends an ACK,
   then a BYE. */

   F8 CANCEL A -> Proxy 1

   CANCEL sip:UserB@ss1.wcom.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@ss2.wcom.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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 2 challenges User A for authentication */



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   F10 407 Proxy Authorization Required Proxy 2 -> 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: 12345601@here.com
   CSeq: 1 INVITE
   Proxy-Authenticate: Digest realm="MCI WorldCom SIP",
   domain="wcom.com", nonce="1ae6cbe5ea9c8e8df84fqnlec434a359",
   opaque="", stale="FALSE", algorithm="MD5"
   Content-Length: 0


   F11 ACK A -> Proxy 2

   ACK sip:UserB@ss2.wcom.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
   Content-Length: 0


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

   F12 INVITE A -> Proxy 2

   INVITE sip:UserB@ss2.wcom.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: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="1ae6cbe5ea9c8e8df84fqnlec434a359", opaque="",
   uri="sip:ss2.wcom.com", response="8a880c919d1a52f20a1593e228adf599"
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0



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   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 49170 from the network.
   */

   F13 INVITE Proxy 2 -> B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345602@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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
   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
   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



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   CSeq: 1 INVITE
   Content-Length: 0


   F16 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: 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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   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@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134



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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 A -> Proxy 2

   ACK sip:UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com>
   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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   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@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   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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   F22 BYE Proxy 2 -> User A

   BYE sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   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


   F23 200 OK User A -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   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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3.1.4
     Successful SIP to SIP through SIP Firewall Proxy

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

   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 (F11) messages.

   In addition to firewall traversal, this back-to-back User Agent
   Client and User Agent Server 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.








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   Message Details


   F1 INVITE A -> SIP FW

   INVITE sip:UserB@ fwp1.wcom.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: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="85b4f1cen4341ae6cbe5a3a9c8e88df9", opaque="",
   uri="sip:ss1.wcom.com", response="b3f392f9218a328b9294076d708e6815"
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Client for A prepares to receive data on port 49170 from the
   network. */

   F2 INVITE SIP FW -> Proxy 1

   INVITE sip:UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@fwp1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="85b4f1cen4341ae6cbe5a3a9c8e88df9", opaque="",
   uri="sip:ss1.wcom.com", response="b3f392f9218a328b9294076d708e6815"
   Content-Type: application/sdp
   Content-Length: 134

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com



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   s=Session SDP
   c=IN IP4 200.201.202.203
   t=0 0
   m=audio 1000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   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 1000 to
   100.101.102.103/49170.   Proxy 1 uses a location manager 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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>,<sip:UserB@fwp1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 134

   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



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   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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) B -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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 -> SIP FW

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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 SIP FW -> 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: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F10 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>,<sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 133

   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 Proxy 1 -> SIP FW

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>,<sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113



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


   F12 200 OK SIP FW -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>,<sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   v=0
   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 port 1002 to
   110.111.112.113/3456 */

   F13 ACK A -> SIP FW

   ACK sip:UserB@fwp1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:ss1.wcom.com>, <sip:UserB@there.com>
   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 SIP FW -> Proxy 1

   ACK sip:UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159



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


   F15 ACK Proxy 1 -> B

   ACK sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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
   between the Firewall Proxy and B*/

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

   F16 BYE A -> SIP FW

   BYE sip: UserB@fwp1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:ss1.wcom.com>, <sip:UserB@there.com>
   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 SIP FW -> Proxy 1

   BYE sip: UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com>
   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 BYE F18 Proxy 1 -> B




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   BYE sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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


   F19 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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 Proxy 1 -> SIP FW

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP fwp1.wcom.com:5060
   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


   F21 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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3.1.5
     Successful SIP to SIP via Redirect and Proxy

         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.

   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@redirect.wcom.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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Client for A prepares to receive data on port 49170 from the
   network. */

   F2 302 Moved Temporarily Redirect Proxy  -> A

   SIP/2.0 302 Moved Temporarily
   Contact: sip:UserB@ss2.wcom.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@redirect.wcom.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
   Content-Length: 0


   F4 INVITE A -> Proxy 2

   INVITE sip:UserB@ss2.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: 2 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 INVITE Proxy 2 -> B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   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




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   F7 (100 Trying) B -> Proxy 2

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   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
   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
   Content-Length: 0


   F10 200 OK B -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134




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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


   F11 200 OK Proxy -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   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


   F12 ACK A -> Proxy 2

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


   F13 ACK Proxy 2 -> B

   ACK sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   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 ACK
   Content-Length: 0


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

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

   F14 BYE B -> Proxy 2

   BYE sip: UserA@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   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


   F15 BYE Proxy 2 -> A

   BYE sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP there.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
   Via: SIP/2.0/UDP there.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 there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159



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   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345600@here.com
   CSeq: 1 BYE
   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    |
           |                |                |<---------------|
           |                |                |                |

   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.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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



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   Contact: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="ze7k1ee88df84f1cec431ae6cbe5a359", opaque="",
   uri="sip:ss1.wcom.com", response="b00b416324679d7e243f55708d44be7b"
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49170 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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



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   Content-Length: 0


   F4 INVITE Proxy 2 -> B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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
   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 180 Ringing B -> Proxy 2

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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



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   F7 180 Ringing Proxy 2 -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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@ss1.wcom.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 2 -> B

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


   F11 CANCEL Proxy 1 -> Proxy 2

   CANCEL sip: UserA@ss2.wcom.com SIP/2.0



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   Via: SIP/2.0/UDP ss1.wcom.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


   F12 200 OK Proxy 1 -> Proxy 2

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.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

   F13 CANCEL Proxy 2 -> B

   CANCEL sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.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


   F14 200 OK A -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss2.wcom.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















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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@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="dc3a5ab2530aa93112cf5904ba7d88fa", opaque="",
   uri="sip:ss1.wcom.com", response="702138b27d869ac8741e10ec643d55be"
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49170 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>



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   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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
   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
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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 ACK Proxy 2 -> User B

   ACK sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.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




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   F8 486 Busy Here Proxy 2 -> Proxy 1

   SIP/2.0  486 Busy Here
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F9 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.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


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


   F11 ACK User A -> Proxy 1

   ACK sip:UserB@ss1.wcom.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=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0









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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@ss1.wcom.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



Johnston, et al.            Informational                    [Page 64]


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   Contact: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="cf5904ba7d8dc3a5ab2530aa931128fa", opaque="",
   uri="sip:ss1.wcom.com", response="7afc04be7961f053c24f80e7dbaf888f"
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49170 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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



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   Content-Length: 0


   F4 INVITE Proxy 2 -> User B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 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
   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



Johnston, et al.            Informational                    [Page 66]


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   Resend of Message F4


   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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.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


   F13 480 No Response Proxy 2 -> Proxy 1

   SIP/2.0  480 No Response
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F14 ACK Proxy 1 -> Proxy 2

   ACK sip:UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


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


   F16 ACK User A -> Proxy 1

   ACK sip:UserB@ss1.wcom.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=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0






























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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@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="aa9311cf5904ba7d8dc3a5ab253028fa", opaque="",
   uri="sip:ss1.wcom.com", response="59a46a91bf1646562a4d486c84b399db"
   Content-Type: application/sdp
   Content-Length: 132



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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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /*Client for A prepares to receive data on port 49170 from the
   network.*/

   F2 INVITE Proxy 1 -> Proxy 2

   INVITE sip:UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 INVITE Proxy 2 -> B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss2.wcom.com>,<sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132



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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 49170 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
   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
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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



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   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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 2 -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss2.wcom.com:5060
   From: BigGuy <sip:UserA@here.com>



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   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   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
   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


   F13 ACK Proxy 1 -> Proxy 2

   ACK sip: UserB@ss2.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.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


   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=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F15 ACK A -> Proxy 1

   ACK sip:UserB@ss1.wcom.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=314159
   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[5] responses used throughout this document
   in place of the 180 Ringing.

   One example of the use of reliable provisional responses[6] for the
   183 Session Progress message is given in scenario 4.1.2.

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     |<---------------|                |
           |<---------------|                |                |
           |        Both Way RTP Media       |  One Way Voice |
           |<===============================>|<===============|
           |                |                |      ANM F9    |
           |                |    200 F10     |<---------------|
           |     200 F11    |<---------------|                |
           |<---------------|                |                |
           |     ACK F12    |                |                |
           |--------------->|     ACK F13    |                |
           |                |--------------->|                |
           |        Both Way RTP Media       | Both Way Voice |
           |<===============================>|<==============>|
           |     BYE F14    |                |                |
           |--------------->|     BYE F15    |                |
           |                |--------------->|                |
           |                |     200 F16    |                |
           |     200 F17    |<---------------|     REL F18    |
           |<---------------|                |--------------->|
           |                |                |     RLC F19    |
           |                |                |<---------------|
           |                |                |                |



   User A dials the globalized E.164 number +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



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   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
   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: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
   uri="sip:ss1.wcom.com", response="ccdca50cb091d587421457305d097458c"
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 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 manager function to determine where B is
   located.  Based upon location analysis the call is forwarded to NGW



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   1.  Client for A prepares to receive data on port 49170 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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 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
   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




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   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
   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-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 proxies the OK to User A.  At this point NGW 1 will start
   sending an RTP path to the receive port on A encoding anything that
   is being received from B via the PSTN network (i.e. ringing) */

   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-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


   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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: 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


   F11 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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



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   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


   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

   ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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



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   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
   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
   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


   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



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   RLC



















































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4.1.2
     Successful SIP to ISDN PBX call

         User A          Proxy 1           GW 1             PBX C
           |                |                |                |
           |   INVITE F1    |                |                |
           |--------------->|                |                |
           |     406 F2     |                |                |
           |<---------------|                |                |
           |     ACK F3     |                |                |
           |--------------->|                |                |
           |   INVITE F4    |                |                |
           |--------------->|                |                |
           |    (100) F5    |                |                |
           |<---------------|   INVITE F6    |                |
           |                |--------------->|                |
           |                |    (100) F7    |                |
           |                |<---------------|    SETUP F8    |
           |                |                |--------------->|
           |                |                |  CALL PROC F9  |
           |                |                |<---------------|
           |                |                |  PROGress F10  |
           |                |    183 F11     |<---------------|
           |    183 F12     |<---------------|                |
           |<---------------|                |                |
           |   PRACK F13    |                |                |
           |--------------->|    PRACK F14   |                |
           |                |--------------->|                |
           |                |     200 F15    |                |
           |     200 F16    |<---------------|                |
           |<---------------|                |                |
           |        Both Way RTP Media       |  One Way Voice |
           |<===============================>|<===============|
           |                |                |   CONNect F17  |
           |                |                |<---------------|
           |                |                | CONNect ACK F18|
           |                |    200 F19     |--------------->|
           |     200 F20    |<---------------|                |
           |<---------------|                |                |
           |     ACK F21    |                |                |
           |--------------->|     ACK F22    |                |
           |                |--------------->|                |
           |        Both Way RTP Media       | Both Way Voice |
           |<===============================>|<==============>|
           |     BYE F23    |                |                |
           |--------------->|     BYE F24    |                |
           |                |--------------->|                |
           |                |     200 F25    |                |
           |     200 F26    |<---------------| DISConnect F27 |
           |<---------------|                |--------------->|



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           |                |                |   RELease F28  |
           |                |                |<---------------|
           |                |                | RELease COM F29|
           |                |                |--------------->|
           |                |                |                |

   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.

   In this example, the initial INVITE is processed by the Proxy which
   determines that the Enterprise Gateway in the call route requires the
   in-band alerting response 183 Session Progress to be sent
   reliably[6].  The absence of a Supported header in the INVITE
   indicating support by User A of the Reliable Provisional Response
   extension (org.ietf.sip.100rel) causes the Proxy to reject the call
   with a 406 Not Acceptable containing the Required:
   org.ietf.sip.100rel header.  User A then re-sends the INVITE
   containing the Supported header.

   The phone-context tag in 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
   could get forwarded by GW 1 and the context of the digits could
   become lost and the call unroutable.  See section 1.1 for a
   discussion of phone-context.

   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.

   The 183 response contains the RSeq header which causes User A to
   acknowledge receipt of the provisional response message by sending a
   PRACK (Provisional Response Acknowledgement) request.  GW 1 receives
   the PRACK containing the RAck header and responds with a 200 OK.


   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



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   To: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com>;user=phone
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="qo0dc3a5ab22aa931904badfa1cf5j9h", opaque="",
   uri="sip:ss1.wcom.com", response="6c792f5c9fa360358b93c7fb826bf550"
   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a location manager function to determine where B is
   located.  GW 1 requires use of Reliable Provisional Response.  Since
   User has not indicated that it supports this extension, the INVITE is
   rejected. */

   F2 406 Not Acceptable Proxy 1 -> User A

   SIP/2.0 406 Not Acceptable
   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: 1 INVITE
   Require: org.ietf.sip.100rel
   Proxy-Authenticate: Digest realm="MCI WorldCom SIP",
   domain="wcom.com", nonce="x41ae6cbe5aea9c8e8wf84f1cecz8d359",
   opaque="", stale="FALSE", algorithm="MD5"
   Content-Length: 0

   F3 ACK A -> Proxy 1

   ACK 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: 1 INVITE
   Content-Length: 0


   /* User A re-sends the INVITE indicating support of the extension. */



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   F4 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: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="x41ae6cbe5aea9c8e8wf84f1cecz8d359", opaque="",
   uri="sip:ss1.wcom.com", response="f27e0e74cc29ee4761d342bdd6719edd"
   Supported: org.ietf.sip.100rel
   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F5 (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


   F6 INVITE Proxy 1 -> GW 1

   INVITE sip:444-3333@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: BigGuy <sip:UserA@here.com>
   Supported: org.ietf.sip.100rel
   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 (100 Trying) GW -> Proxy 1

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP ss1.wcom.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


   F8 SETUP GW 1 -> User C

   Protocol discriminator=Q.931
   Call reference: Flag=0, CR  value=any valid value not in use
   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=?? (private numbering plan)
   Digits=444-3333


   F9 CALL PROCeeding User C -> GW 1

   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F5 SETUP message
   Message type=CALL PROC
   Channel identification=Exclusive B-channel


   F10 PROGress User C -> GW 1

   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F5 SETUP message
   Message type=PROG



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   Progress indicator=1 (Call is not end-to-end ISDN;further call
   progress information may be available inband)


   F11 183 Session Progress GW 1 -> Proxy 1

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   RSeq: 42321
   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-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


   /* GW 1 will establish an RTP path to the receive port on A encoding
   anything that is being received from C via the PSTN network (i.e.
   ringing) */

   F12 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP here.com:5060
   RSeq: 42321
   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-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



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   a=rtpmap:0 PCMU/8000


   /* User A confirms reception of the 183 response */

   F13 PRACK A -> Proxy 1

   PRACK sip:444-3333@ss1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   RAck: 42321 2 INVITE
   Route: <sip:444-3333@gw1.wcom.com>;phone-context=p1234
   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 PRACK
   Content-Length: 0


   F14 PRACK Proxy 1 -> GW 1

   PRACK sip:444-3333@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   RAck: 42321 2 INVITE
   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 PRACK
   Content-Length: 0


   F15 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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 PRACK
   Content-Length: 0


   F16 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   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: OtherGuy <sip:+1-918-555-3333@ss1.wcom.com>;user=phone
   ;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 3 PRACK
   Content-Length: 0


   F17 CONNect User C -> GW 1

   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F5 SETUP message
   Message type=CONN


   F18 CONNect ACK GW 1 -> User C

   Protocol discriminator=Q.931
   Call reference: Flag=0, CR  value=value in F5 SETUP message
   Message type=CONN ACK


   F19 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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@gw1.wcom.com;phone-context=p1234
   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


   F20 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060



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   Record-Route: <sip:+1-314-555-1111@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@gw1.wcom.com;phone-context=p1234
   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


   F21 ACK A -> Proxy 1

   ACK sip:444-3333@ss1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:444-3333@gw1.wcom.com>;phone-context=p1234
   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


   F22 ACK Proxy 1 -> GW 1

   ACK sip:444-3333@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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. */

   F23 BYE A -> Proxy 1




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   BYE sip:444-3333@ss1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:444-3333@gw1.wcom.com>;phone-context=p1234
   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: 4 BYE
   Content-Length: 0


   F24 BYE Proxy 1 -> GW 1

   BYE sip:444-3333@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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: 4 BYE
   Content-Length: 0


   F25 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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: 4 BYE
   Content-Length: 0


   F26 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: 4 BYE
   Content-Length: 0


   F27 DISConnect GW 1 -> User C



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   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F4 SETUP message
   Message type=DISC
   Cause=16 (Normal clearing)


   F28 RELease User C -> GW 1

   Protocol discriminator=Q.931
   Call reference: Flag=0, CR  value=value in F4 SETUP message
   Message type=REL


   F29 RELease COMplete GW 1 -> User C

   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F4 SETUP message
   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    |<----------------------------|              |
       |<-------------|                             |              |
       |              Both 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. GW 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




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   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: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="b59311c3ba05b401cf80b2a2c5ac51b0", opaque="",
   uri="sip:ss1.wcom.com", response="ba6ab44923fa2614b28e3e3957789ab0"
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a location manager 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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 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
   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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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
   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


   /* 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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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



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   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 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
   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



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   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


   /* The GW will establish an RTP path to the receive port on A
   encoding anything that is being received from B via the PSTN network
   (i.e. ringing) */

   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-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


   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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@ss1.wcom.com>
   From: BigGuy <sip:+1-314-555-1111@ss1.wcom.com>;user=phone




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   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: 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


   F13 200 OK Proxy 1 -> User A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: 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


   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



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   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
   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
   ;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
   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



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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
   ;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

   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[5] 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     |<---------------|                |
           |<---------------|                |                |
           |        Both Way RTP Media       |  One Way Voice |
           |<===============================>|<===============|
           |                 Treatment Applied                |
           |<=================================================|
           |   CANCEL F9    |                |                |
           |--------------->|                |                |
           |     200 F10    |                |                |
           |<---------------|   CANCEL F11   |                |
           |                |--------------->|                |
           |                |     200 F12    |                |
           |                |<---------------|     REL F13    |
           |                |                |--------------->|
           |                |                |     RLC F14    |
           |                |                |<---------------|
           |                |                |                |

   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
   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



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   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="01cf8311c3b0b2a2c5ac51bb59a05b40", opaque="",
   uri="sip:ss1.wcom.com", response="e178fbe430e6680a1690261af8831f40"
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 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 manager 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 49170 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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

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



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   c=IN IP4 here.com
   t=0 0
   m=audio 49170 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
   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
   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



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   Call-ID: 12345600@here.com
   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 gatewayone.wcom.com
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 183 Session Progress Proxy 1 -> User A

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 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-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


   /* User A listens to recorded announcement from the PSTN then hangs
   up */

   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



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   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
   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


   F12 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.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


   F13 REL NGW 1 -> B

   REL
   CauseCode=16 Normal
   CodingStandard=CCITT


   F14 RLC B -> NGW 1

   RLC











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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 [9].


   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: BigGuy <sip:UserA@here.com>





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   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="j1c3b0b01cf832da2c5ac51bb59a05b40", opaque="",
   uri="sip:ss1.wcom.com", response="a451358d46b55512863efe1dccaa2f42"
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 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 manager 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 49170 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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

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



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   m=audio 49170 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
   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=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 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
   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



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   Content-Length: 0


   F9 ACK Proxy 1 -> NGW 1

   ACK sip:+44-1234@ngw1.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.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 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=314159
   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=314159
   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: BigGuy <sip:UserA@here.com>



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   Authorization:Digest username="UserA", realm="MCI WorldCom SIP",
   nonce="da2c5ac51bb59a05j1c3b0b01cf832b40", opaque="",
   uri="sip:ss1.wcom.com", response="579cb9db184cdc25bf816f37cbc03c7d"
   Content-Type: application/sdp
   Content-Length: 132

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 here.com
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 uses a location manager 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 49170 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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:+1-314-555-1111@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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: 132

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



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   m=audio 49170 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
   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
   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



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   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


   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-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


   /* 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




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   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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
   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 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=314159
   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=314159
   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



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   CPT=0 0
   C=Normal
   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 manager 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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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



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   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
   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
   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



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   F8 200 OK User B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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: LittleGuy <sip:UserB@there.com>
   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>
   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: LittleGuy <sip:UserB@there.com>
   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@ss1.wcom.com SIP/2.0



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   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@there.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 ACK
   Content-Length: 0


   F11 ACK Proxy 1 -> User B

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@there.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



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   CSeq: 2 BYE
   Content-Length: 0


   F16 BYE Proxy 1 -> User B

   BYE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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
   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 or ITU ISUP, 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





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   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 manager 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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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




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   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 200 OK User B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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: LittleGuy <sip:UserB@there.com>
   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>
   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: LittleGuy <sip:UserB@there.com>
   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



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


   F7 ACK NGW 1 -> Proxy 1

   ACK UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@there.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 ACK
   Content-Length: 0


   F8 ACK Proxy 1 -> User B

   ACK UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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



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   BYE sip:UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:UserB@there.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: 2 BYE
   Content-Length: 0


   F13 BYE Proxy 1 -> User B

   BYE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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
   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.




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   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
   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.

   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 manager 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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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



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   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


   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
   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. */




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   F7 200 OK User B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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: LittleGuy <sip:UserB@there.com>
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 134

   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>
   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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   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@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Route: <sip:UserB@there.com>



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   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


   F10 ACK Proxy 1 -> User B

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Route: <sip:UserB@there.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: 2 BYE
   Content-Length: 0


   F12 BYE Proxy 1 -> User B

   BYE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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



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   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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
   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




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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


   /* Proxy 1 uses a location manager 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
   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
   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 [9].


   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





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   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 manager 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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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



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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=314159
   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
   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
   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=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 ACK NGW 1 -> Proxy 1

   ACK UserB@ss1.wcom.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=314159
   Call-ID: 12345602@ngw1.wcom.com
   CSeq: 1 ACK



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   Content-Length: 0


   F9 REL User A -> NGW 1

   REL
   CauseCode=17 Busy
   CodingStandard=CCITT


   F10 RLC NGW 1 -> User A

   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(17) 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, not that for ETSI or ITU ISUP, a CONnect message would be sent
   intead of the Answer Message.


   Message Details


   F1 IAM User A -> NGW 1

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



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   CdPN=972-555-2222,NPI=E.164,NOA=National
   USI=Speech
   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 manager 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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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



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   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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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
   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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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


   F6 ACK Proxy 1 -> User B

   ACK UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.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


   F7 600 Busy Everywhere Proxy 1 -> NGW 1

   SIP/2.0 600 Busy Everywhere
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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



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   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
   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 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


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

   /* Call Released after NGW treatment timer expires. */

   F10 REL User A -> NGW 1

   REL
   CauseCode=17
   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




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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 manager 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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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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   c 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
   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




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   F11 RLC NGW 1 -> User A

   RLC


   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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.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


   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



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   Same as Message F14


   F18 CANCEL Proxy 1 -> User B

   Same as Message F14


   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           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.


   Message Details


   F1 IAM User A -> NGW 1




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   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

   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 manager 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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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



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


   F4 INVITE NGW 1 -> Proxy 1

   Same as Message F2


   F5 INVITE Proxy 1 -> User B

   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




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   F13 INVITE Proxy 1 -> User B

   Same as Message F3


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

   F14 REL User A -> NGW 1

   REL
   CauseCode=16 Normal
   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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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




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   Same as Message F17


   F20 CANCEL NGW 1 -> Proxy 1

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


   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


   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



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   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 manager 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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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



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   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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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
   Content-Length: 0


   F5 180 Ringing User B -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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


   F6 180 Ringing Proxy 1 -> NGW 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-972-555-2222@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


   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


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

   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@ss1.wcom.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@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.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



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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
   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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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[9].

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 establish a media path between
   the two Gateways, allowing in-band alerting to pass 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|     Both 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   |<-------------|    REL 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 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 phone-context tag to
   identify what private dialing plan is being referenced.  For more
   information on phone-context, refer to [7].  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: 134

   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 INVITE Proxy 1 -> GW 2

   INVITE sip:444-3333@gw2.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw1.wcom.com:5060



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   Record-Route: <sip:+1-918-555-3333@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: 134

   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 49170 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 with SDP
   allowing in-band call progress indications to be sent to User A
   through NGW 1. */

   F6 183 Session Progress GW 2 -> Proxy 1

   SIP/2.0 183 Session Progress



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   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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-Type: application/sdp
   Content-Length: 134

   v=0
   o=PBX_B 987654321 987654321 IN IP4 gw2.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


   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-Type: application/sdp
   Content-Length: 134

   v=0
   o=PBX_B 987654321 987654321 IN IP4 gw2.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


   /* 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



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   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
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Record-Route: <sip:+1-918-555-3333@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@gw2.wcom.com;phone-context=p1234
   Content-Type: application/sdp
   Content-Length: 134

   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>
   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@gw2.wcom.com;phone-context=p1234
   Content-Type: application/sdp
   Content-Length: 134

   v=0



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   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@ss1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ngw1.wcom.com:5060
   Route: <sip:444-3333@gw2.wcom.com>;phone-context=p1234
   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
   CSeq: 1 ACK
   Content-Length: 0


   F14 ACK Proxy 1 -> GW 2

   ACK sip:444-3333@gw2.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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




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   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
   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


   F19 200 OK NGW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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





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   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|
   |            |            |                         |<-----------|
   |            |            |                         |  PROG F10  |
   |            |            |          183 F11        |<-----------|
   |            |  183 F12   |<------------------------|            |
   |            |<-----------|                         |            |
   |OneWay Voice|          Both 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@ss1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234
   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000




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   /* Proxy 1 uses a location manager function to determine where B is
   located.  Response is returned listing alternative routes. */

   F3 INVITE Proxy 1 -> GW 2

   INVITE sip:444-3333@gw2.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:444-3333@ss1.wcom.com>;phone-context=p1234
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234
   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 49170 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
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234;tag=314159
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 ACK Proxy 1 -> GW 2

   ACK sip:444-3333@gw2.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234;tag=314159
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0




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   F6 INVITE Proxy 1 -> GW 3

   INVITE sip:+1-918-555-3333@gw3.wcom.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:444-3333@ss1.wcom.com>;phone-context=p1234
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234
   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 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 SETUP GW 3 -> PBX C

   Protocol discriminator=Q.931
   Call reference: Flag=0, CR  value=any valid value not in use
   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@ss1.wcom.com;phone-context=p1234
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Length: 0




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   F9 CALL PROCeeding PBX C -> GW 3

   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F9 SETUP message
   Message type=CALL PROC
   Channel identification=Exclusive B-channel


   F10 PROGress PBX C -> GW 3

   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F9 SETUP message
   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 with SDP
   allowing in-band call progress indications to be 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
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 134

   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 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@ss1.wcom.com;phone-context=p1234;tag=123456789
   Call-ID: 12345600@gw1.wcom.com



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   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length: 134

   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 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, CR  value=value in F9 SETUP message
   Message type=CONN


   F14 200 OK GW 3 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:444-3333@ss1.wcom.com>;phone-context=p1234
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234;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: 134

   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



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   Via: SIP/2.0/UDP gw1.wcom.com:5060
   Record-Route: <sip:444-3333@ss1.wcom.com>;phone-context=p1234
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234;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: 134

   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

   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@ss1.wcom.com;phone-context=p1234;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
   Via: SIP/2.0/UDP gw1.wcom.com:5060
   From: PBX_A <sip:IdentifierString@gw1.wcom.com>
   To: sip:444-3333@ss1.wcom.com;phone-context=p1234;tag=123456789
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 ACK
   Content-Length: 0


   F18 CONNect ACK GW 3 -> PBX C




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   Protocol discriminator=Q.931
   Call reference: Flag=0, CR  value=value in F9 SETUP message
   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, CR  value=value in F9 SETUP message
   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@ss1.wcom.com;phone-context=p1234;tag=123456789
   To: PBX_A <sip:IdentifierString@gw1.wcom.com>
   Call-ID: 12345600@gw1.wcom.com
   CSeq: 1 BYE
   Content-Length: 0


   F21 BYE Proxy 1 -> GW 1

   BYE sip:IdentifierString@gw1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw3.wcom.com:5060
   From: sip:444-3333@ss1.wcom.com;phone-context=p1234;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, CR  value=value in F9 SETUP message



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   Message type=REL


   F23 200 OK GW 1 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP gw3.wcom.com:5060
   From: sip:444-3333@ss1.wcom.com;phone-context=p1234;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@ss1.wcom.com;phone-context=p1234;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

   Protocol discriminator=Q.931
   Call reference: Flag=1, CR  value=value in F9 SETUP message
   Message type=REL COM


   PBX A -> GW 1

   Seizure removal

















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7  IP Telephony Services Features Call Flows

   These call flows show how a number of standard telephony features can
   be implemented using SIP.  They are not meant to represent a complete
   set.  Some calls make use of SIP Call Control Extensions[10].

7.1 Call Hold

         User A           Proxy          User B
           |                |              |
           |    INVITE F1   |              |
           |--------------->|              |
           |     407 F2     |
           |<---------------|              |
           |     ACK F3     |              |
           |--------------->|              |
           |    INVITE F4   |              |
           |--------------->|              |
           |                |   INVITE F5  |
           |(100 Trying) F6 |------------->|
           |<---------------|              |
           |                |180 Ringing F7|
           | 180 Ringing F8 |<-------------|
           |<---------------|              |
           |                |  200 OK F9   |
           |    200 OK F10  |<-------------|
           |<---------------|              |
           |     ACK F11    |              |
           |--------------->|    ACK F12   |
           |                |------------->|
           |    Both way RTP Established   |
           |<=============================>|
           |                |INVITE (c=0) F13
           |INVITE (c=0) F14|<-------------|
           |<---------------|              |
           |    200 OK F15  |              |
           |--------------->|   200 OK F16 |
           |                |------------->|
           |                |     ACK F17  |
           |     ACK F18    |<-------------|
           |<---------------|              |
           |           No RTP Sent!        |
           |                |  INVITE F19  |
           |   INVITE F20   |<-------------|
           |<---------------|              |
           |   200 OK F21   |              |
           |--------------->|  200 OK F22  |
           |                |------------->|
           |                |    ACK F23   |
           |     ACK F24    |<-------------|



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           |<---------------|              |
           |    Both way RTP Established   |
           |<=============================>|
           |     BYE F25    |              |
           |--------------->|   BYE F26    |
           |                |------------->|
           |                |   200 OK F27 |
           |    200 OK F28  |<-------------|
           |<---------------|              |
           |                |              |

   User A calls User B, User B places call on hold.  User B then takes
   call off hold.  User A hangs up call.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 challenges User A for authentication */

   F2 407 Proxy Authorization Required Proxy 1 -> 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




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   Proxy-Authenticate: Digest realm="MCI WorldCom SIP",
   domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359",
   opaque="", stale="FALSE", algorithm="MD5"
   Content-Length: 0

   F3 ACK A -> Proxy 1

   ACK sip:UserB@ss1.wcom.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=314159
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length:0


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

   F4 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="b0b0da2c5ac51bb59a05j1c31cf832b40", opaque="",
   uri="sip:ss1.wcom.com", response="006b4853fd66774e6c5599867c72cb59"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE User B.
   Client for A prepares to receive data on port 49170 from the network.
   */

   F5 INVITE Proxy 1 -> B

   INVITE sip:UserB@there.com SIP/2.0



Johnston, et al.            Informational                   [Page 181]


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

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F6 (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: 12345601@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F7 180 Ringing B -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   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: 12345601@here.com



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   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
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip: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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F10 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip: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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 ACK A -> Proxy 1




Johnston, et al.            Informational                   [Page 183]


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   ACK sip:UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com>
   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


   F12 ACK Proxy 1 -> B

   ACK sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   /* User B places User A on hold. */

   F13 INVITE B -> Proxy 1

   INVITE sip:UserA@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 INVITE Proxy 1 -> A

   INVITE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060



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   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 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F15 200 OK A -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F16 200 OK Proxy 1 -> 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>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp



Johnston, et al.            Informational                   [Page 185]


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   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F17 ACK B -> Proxy 1

   ACK sip:UserA@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserB@there.com>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F18 ACK Proxy 1 -> A

   ACK sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* User B takes the call off hold */

   F19 INVITE B -> Proxy 1

   INVITE sip:UserA@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 2 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...




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   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F20 INVITE Proxy 1 -> A

   INVITE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 2 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F21 200 OK A -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 2 INVITE
   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0



Johnston, et al.            Informational                   [Page 187]


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   a=rtpmap:0 PCMU/8000


   F22 200 OK Proxy 1 -> 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>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 2 INVITE
   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F23 ACK B -> Proxy 1

   ACK sip:UserA@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 2 ACK
   Content-Length: 0


   F24 ACK Proxy 1 -> A

   ACK sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345601@here.com
   CSeq: 2 ACK
   Content-Length: 0


   F25 BYE A -> Proxy 1




Johnston, et al.            Informational                   [Page 188]


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   BYE sip: UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:userB@there.com>
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F26 BYE Proxy 1 -> B

   BYE sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F27 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F28 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 2 BYE
   Content-Length: 0










Johnston, et al.            Informational                   [Page 189]


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7.2 Consultation Hold


         User A           Proxy          User B              User C
           |                |              |                   |
           |    INVITE F1   |              |                   |
           |--------------->|   INVITE F2  |                   |
           |                |------------->|                   |
           |(100 Trying) F3 |              |                   |
           |<---------------|180 Ringing F4|                   |
           |                |<-------------|                   |
           | 180 Ringing F5 |              |                   |
           |<---------------|  200 OK F6   |                   |
           |                |<-------------|                   |
           |    200 OK F7   |              |                   |
           |<---------------|              |                   |
           |     ACK F8     |              |                   |
           |--------------->|     ACK F9   |                   |
           |                |------------->|                   |
           |     Both way RTP Established  |                   |
           |<=============================>|                   |
           |                |INVITE (c=0) F10                  |
           |INVITE (c=0) F11|<-------------|                   |
           |<---------------|              |                   |
           |    200 OK F12  |              |                   |
           |--------------->|   200 OK F13 |                   |
           |                |------------->|                   |
           |                |     ACK F14  |                   |
           |                |<-------------|                   |
           |    ACK F15     |              |                   |
           |<---------------|              |                   |
           |         No RTP Sent!          |                   |
           |                |   INVITE F16 |                   |
           |                |<-------------|                   |
           |                |              |    INVITE F17     |
           |                |--------------------------------->|
           |                |(100 Trying) F18                  |
           |                |------------->|                   |
           |                |              | 180 Ringing F19   |
           |                |<---------------------------------|
           |                | 180 Ringing F20                  |
           |                |------------->|                   |
           |                |              |    200 OK F21     |
           |                |<---------------------------------|
           |                |   200 OK F22 |                   |
           |                |------------->|                   |
           |                |    ACK F23   |                   |
           |                |<-------------|                   |
           |                |              |      ACK F24      |
           |                |--------------------------------->|



Johnston, et al.            Informational                   [Page 190]


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           |                |         Both way RTP Established |
           |                |              |<=================>|
           |                |   BYE F25    |                   |
           |                |<-------------|                   |
           |                |              |       BYE F26     |
           |                |--------------------------------->|
           |                |              |    200 OK F27     |
           |                |<---------------------------------|
           |                |   200 OK F28 |                   |
           |                |------------->|                   |
           |                |   INVITE F29 |                   |
           |    INVITE F30  |<-------------|                   |
           |<---------------|              |                   |
           |    200 OK F31  |              |                   |
           |--------------->|   200 OK F32 |                   |
           |                |------------->|                   |
           |                |     ACK F33  |                   |
           |                |<-------------|                   |
           |    ACK F34     |              |                   |
           |<---------------|              |                   |
           |     Both way RTP Established  |                   |
           |<=============================>|                   |
           |    BYE F35     |              |                   |
           |--------------->|    BYE F36   |                   |
           |                |------------->|                   |
           |                |   200 OK F37 |                   |
           |                |<-------------|                   |
           |  200 OK F38    |              |                   |
           |<---------------|              |                   |
           |                |              |                   |

   User A calls user B.  User B places call on hold.  User B calls User
   C, after that call is finished User B take the call with User A off
   hold.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>





Johnston, et al.            Informational                   [Page 191]


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   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="c5ac31cf51bb5b0b0da29a05j1c832b40", opaque="",
   uri="sip:ss1.wcom.com", response="dc832edc8930a761ce764bcde5ba8075"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE User B.
   Client for A prepares to receive data on port 49170 from the network.
   */

   F2 INVITE Proxy 1 -> B

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

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 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



Johnston, et al.            Informational                   [Page 192]


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   Content-Length: 0


   F4 180 Ringing B -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F5 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


   F6 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 200 OK Proxy 1 -> A



Johnston, et al.            Informational                   [Page 193]


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   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 ACK A -> Proxy 1

   ACK sip:UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com>
   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


   F9 ACK Proxy 1 -> B

   ACK sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   /* User B places User A on hold. */

   F10 INVITE B -> Proxy 1

   INVITE sip:UserA@ss1.wcom.com SIP/2.0



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

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 INVITE Proxy 1 -> A

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

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F12 200 OK A -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE



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   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F13 200 OK Proxy 1 -> 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>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 0.0.0.0
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F14 ACK B -> Proxy 1

   ACK sip:UserA@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F15 ACK Proxy 1 -> A

   ACK sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060



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   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F16 INVITE B -> Proxy 1

   INVITE sip:UserC@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844526 2890844526 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 50170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F17 INVITE Proxy 1 -> C

   INVITE sip:UserC@anywhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844526 2890844526 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 50170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000



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   F18 (100 Trying) Proxy 1 -> B

   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Content-Length: 0


   F19 180 Ringing C -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@here.com
   CSeq: 1 INVITE
   Content Length:0


   F20 180 Ringing Proxy 1 -> B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Content Length: 0


   F21 200 OK C -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserC@anywhere.com>
   Content-Type: application/sdp
   Content-Length: ...




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   v=0
   o=UserC 2890844527 2890844527 IN IP4 client.anywhere.com
   s=Session SDP
   c=IN IP4 120.121.122.123
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F22 200 OK Proxy 1 -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: OtherGuy <sip:UserC@anywhere.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserC 2890844527 2890844527 IN IP4 client.anywhere.com
   s=Session SDP
   c=IN IP4 120.121.122.123
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F23 ACK B -> Proxy 1

   ACK sip:UserC@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserC@anywhere.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 ACK
   Content-Length:0


   F24 ACK Proxy 1 -> C

   ACK sip: UserC@anywhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654



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   Call-ID: 9876543210@there.com
   CSeq: 1 ACK
   Content-Length:0


   F25 BYE B -> Proxy 1

   BYE sip: UserC@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:userC@anywhere.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 2 BYE
   Content-Length: 0


   F26 BYE Proxy 1 -> C

   BYE sip: UserC@anywhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 2 BYE
   Content-Length: 0


   F27 200 OK C -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 2 BYE
   Content-Length: 0


   F28 200 OK Proxy 1 -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 2 BYE
   Content-Length: 0



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   /* User B takes the call off hold */

   F29 INVITE B -> Proxy 1

   INVITE sip:UserA@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F30 INVITE Proxy 1 -> A

   INVITE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F31 200 OK A -> Proxy 1




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   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F32 200 OK Proxy 1 -> 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>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 INVITE
   Contact: BigGuy <sip:UserA@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F33 ACK B -> Proxy 1

   ACK sip:UserA@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 ACK
   Content-Length: 0



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   F34 ACK Proxy 1 -> A

   ACK sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 ACK
   Content-Length: 0


   F35 BYE A -> Proxy 1

   BYE sip: UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:userB@there.com>
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F36 BYE Proxy 1 -> B

   BYE sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F37 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0





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   F38 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>;tag=1234567
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0












































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7.3 Unattended Transfer


         User A           Proxy          User B     User C     User D
           |                |              |          |          |
           |    INVITE F1   |              |          |          |
           |--------------->|   INVITE F2  |          |          |
           |                |------------->|          |          |
           |(100 Trying) F3 |              |          |          |
           |<---------------|180 Ringing F4|          |          |
           |                |<-------------|          |          |
           | 180 Ringing F5 |              |          |          |
           |<---------------|  200 OK F6   |          |          |
           |                |<-------------|          |          |
           |    200 OK F7   |              |          |          |
           |<---------------|              |          |          |
           |     ACK F8     |              |          |          |
           |--------------->|     ACK F9   |          |          |
           |                |------------->|          |          |
           |     Both way RTP Established  |          |          |
           |<=============================>|          |          |
           | BYE (Also C) F10              |          |          |
           |--------------->| BYE (Also C) F11        |          |
           |                |------------->|          |          |
           |                |   200 OK F12 |          |          |
           |    200 OK F13  |<-------------|          |          |
           |<---------------|              |          |          |
           |                | INVITE (ReqBy A) F14    |          |
           |                |<-------------|          |          |
           |                |    INVITE (ReqBy A) F15 |          |
           |                |------------------------>|          |
           |                |(100 Trying) F16         |          |
           |                |------------->|          |          |
           |                |         180 Ringing F17 |          |
           |                |<------------------------|          |
           |                | 180 Ringing F18         |          |
           |                |------------->|          |          |
           |                |               200 OK F19|          |
           |                |<------------------------|          |
           |                | 200 OK F20   |          |          |
           |                |------------->|          |          |
           |                |    ACK F21   |          |          |
           |                |<-------------|          |          |
           |                |              |  ACK F22 |          |
           |                |------------------------>|          |
           |                |           Both way RTP Established |
           |                |              |<========>|          |
           |                |         BYE (Also D) F23|          |
           |                |<------------------------|          |
           |                | BYE (Also D) F24        |          |



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           |                |------------->|          |          |
           |                | 200 OK F25   |          |          |
           |                |<-------------|          |          |
           |                |              |200 OK F26|          |
           |                |------------------------>|          |
           |                | INVITE (ReqBy C) F27    |          |
           |                |<-------------|          |          |
           |                |        INVITE (ReqBy C) F28 |
           |                |----------------------------------->|
           |                |(100 Trying) F29         |          |
           |                |------------->|          |          |
           |                |              |     180 Ringing F30 |
           |                |<-----------------------------------|
           |                | 180 Ringing F31         |          |
           |                |------------->|          |          |
           |                |              |          |200 OK F32|
           |                |<-----------------------------------|
           |                | 200 OK F33   |          |          |
           |                |------------->|          |          |
           |                |    ACK F34   |          |          |
           |                |<-------------|          |          |
           |                |              |          |  ACK F35 |
           |                |----------------------------------->|
           |                |           Both way RTP Established |
           |                |              |<===================>|
           |                |   BYE F36    |          |          |
           |                |<-------------|          |  BYE F37 |
           |                |----------------------------------->|
           |                |              |          |200 OK F38|
           |                |<-----------------------------------|
           |                | 200 OK F39   |          |          |
           |                |------------->|          |          |
           |                |              |          |          |


   User A call user B.  User A then transfers User B to user C.  User C
   then transfers User B to User D.  User B hangs up the call.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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



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   Contact: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="761ce7cde5ba8d64bc832edc8930a075", opaque="",
   uri="sip:ss1.wcom.com", response="61e1fc70ec739cc79fda2cfaf2791d96"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE User B.
   Client for A prepares to receive data on port 49170 from the network.
   */

   F2 INVITE Proxy 1 -> B

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

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 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



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   CSeq: 1 INVITE
   Content-Length: 0


   F4 180 Ringing B -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F5 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


   F6 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000





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   F7 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.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: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F8 ACK A -> Proxy 1

   ACK sip:UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com>
   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


   F9 ACK Proxy 1 -> B

   ACK sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   /* User A transfers User B to User C */

   F10 BYE A -> Proxy 1




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


   F11 BYE Proxy 1 -> B

   BYE sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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
   Also: OtherGuy <sip:UserC@anywhere.com>
   Content-Length: 0


   F12 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F13 200 OK Proxy 1 -> 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


   F14 INVITE B -> Proxy 1

   INVITE sip:UserC@ss1.wcom.com SIP/2.0



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   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Requested-By: BigGuy <sip:UserA@here.com>
   Content-Length: ...

   v=0
   o=UserB 2890844526 2890844526 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 50170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE User
   B.  Client for A prepares to receive data on port 49170 from the
   network. */

   F15 INVITE Proxy 1 -> C

   INVITE sip:UserC@anywhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Requested-By: BigGuy<sip:UserA@here.com>
   Content-Length: ...

   v=0
   o=UserB 2890844526 2890844526 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 50170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F16 (100 Trying) Proxy 1 -> B

   SIP/2.0 100 Trying



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   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Content-Length: 0


   F17 180 Ringing C -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@here.com
   CSeq: 1 INVITE
   Content Length:0


   F18 180 Ringing Proxy 1 -> B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Content Length: 0


   F19 200 OK C -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserC@anywhere.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserC 2890844527 2890844527 IN IP4 client.anywhere.com
   s=Session SDP
   c=IN IP4 120.121.122.123
   t=3034423619 0



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


   F20 200 OK Proxy 1 -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 INVITE
   Contact: OtherGuy <sip:UserC@anywhere.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserC 2890844527 2890844527 IN IP4 client.anywhere.com
   s=Session SDP
   c=IN IP4 120.121.122.123
   t=3034423619 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F21 ACK B -> Proxy 1

   ACK sip:UserC@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserC@anywhere.com>
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 ACK
   Content-Length:0


   F22 ACK Proxy 1 -> C

   ACK sip: UserC@anywhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   Call-ID: 9876543210@there.com
   CSeq: 1 ACK
   Content-Length:0





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   F23 BYE C -> Proxy 1

   BYE sip: UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP anywhere.com:5060
   Route: <sip:userB@there.com>
   From: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   To: LittleGuy <sip:UserB@there.com>
   Also: TheNewGuy<sip:UserD@nowhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 200 BYE
   Content-Length: 0


   F24 BYE Proxy 1 -> B

   BYE sip: UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP anywhere.com:5060
   From: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   To: LittleGuy <sip:UserB@there.com>;tag=1234567
   Also: TheNewGuy<sip:UserD@nowhere.com>
   Call-ID: 9876543210@there.com
   CSeq: 200 BYE
   Content-Length: 0


   F25 200 OK B -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP anywhere.com:5060
   From: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   To: LittleGuy <sip:UserB@there.com>;tag=1234567
   Call-ID: 9876543210@there.com
   CSeq: 200 BYE
   Content-Length: 0


   F26 200 OK Proxy 1 -> C

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP anywhere.com:5060
   From: OtherGuy <sip:UserC@anywhere.com>;tag=456654
   To: LittleGuy <sip:UserB@there.com>;tag=1234567
   Call-ID: 9876543210@there.com
   CSeq: 200 BYE
   Content-Length: 0


   F27 INVITE B -> Proxy 1



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   INVITE sip:UserD@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>
   Call-ID: 123123123@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Requested-By: OtherGuy <sip:UserC@here.com>
   Content-Length: ...

   v=0
   o=UserB 2890844526 2890844526 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 50170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE to
   User D.  Client for B prepares to receive data on port 49170 from the
   network. */

   F28 INVITE Proxy 1 -> D

   INVITE sip:UserD@nowhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>
   Call-ID: 123123123@there.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Requested-By: OtherGuy<sip:UserC@here.com>
   Content-Length: ...

   v=0
   o=UserB 2890844526 2890844526 IN IP4 client.there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=3034423619 0
   m=audio 50170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F29 (100 Trying) Proxy 1 -> B



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   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>
   Call-ID: 123123123@there.com
   CSeq: 1 INVITE
   Content-Length: 0


   F30 180 Ringing D -> Proxy 1

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 1 INVITE
   Content Length:0


   F31 180 Ringing Proxy 1 -> B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 1 INVITE
   Content Length: 0


   F32 200 OK D -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 1 INVITE
   Contact: TheNewGuy <sip:UserD@nowhere.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserD 2890844527 2890844527 IN IP4 client.nowhere.com
   s=Session SDP



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   c=IN IP4 150.151.152.153
   t=3094598698 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F33 200 OK Proxy 1 -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 1 INVITE
   Contact: TheNewGuy <sip:UserD@nowhere.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserD 2890844527 2890844527 IN IP4 client.nowhere.com
   s=Session SDP
   c=IN IP4 150.151.152.153
   t=3094598698 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F34 ACK B -> Proxy 1

   ACK sip:UserD@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserD@nowhere.com>
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 1 ACK
   Content-Length:0


   F35 ACK Proxy 1 -> D

   ACK sip: UserD@nowhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 1 ACK
   Content-Length:0



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   F36 BYE B -> Proxy 1

   BYE sip: UserD@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:userD@nowhere.com>
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 2 BYE
   Content-Length: 0


   F37 BYE Proxy 1 -> D

   BYE sip: UserD@nowhere.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 2 BYE
   Content-Length: 0


   F38 200 OK D -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 2 BYE
   Content-Length: 0


   F39 200 OK Proxy 1 -> B

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>
   To: TheNewGuy <sip:UserD@nowhere.com>;tag=789789
   Call-ID: 123123123@there.com
   CSeq: 2 BYE
   Content-Length: 0






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7.4 Attended Transfer

   User A calls user B.  User A puts User B on hold, and Calls user C to
   announce transfer.  User A terminates call with User C.  User A
   transfers User B to User C.  The call flow for this session would be
   a combination of Consultation Hold (7.1) and Unattended Transfer
   (7.3).














































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7.5 Call Forwarding Unconditional


         User A           Proxy          Gateway
           |                |              |
           |    INVITE F1   |              |
           |--------------->|   INVITE F2  |
           |(100 Trying) F3 |------------->|
           |<---------------|180 Ringing F4|
           | 180 Ringing F5 |<-------------|
           |<---------------|  200 OK F6   |
           |    200 OK F7   |<-------------|
           |<---------------|              |
           |     ACK F8     |              |
           |--------------->|    ACK F9    |
           |                |------------->|
           |    Both way RTP Established   |
           |<=============================>|
           |     BYE F10    |              |
           |--------------->|   BYE F11    |
           |                |------------->|
           |                |   200 OK F12 |
           |    200 OK F13  |<-------------|
           |<---------------|              |
           |                |              |

   User B wants all calls forwarded to a Private Dialing Plan number
   777-3660. (This information is known by the proxy).  User A calls
   User B.  The Proxy server rewrites the request URI, and forwards the
   INVITE to a gateway.


   Message Details


   F1 INVITE A -> Proxy

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="4bc83v32edc893075761ce7cde5ba8d6", opaque="",
   uri="sip:ss1.wcom.com", response="0a0d6eadd7be1c8639137d175c2302ae"
   Content-Type: application/sdp
   Content-Length: ...




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   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 accepts the credentials and forwards the INVITE the
   gateway for the extension that User B wants his calls forwarded to.
   Client for A prepares to receive data on port 49170 from the network.
   */

   F2 INVITE Proxy -> B2

   INVITE sip:7773660@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 (100 Trying) Proxy -> 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 180 Ringing B2 -> Proxy




Johnston, et al.            Informational                   [Page 221]


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   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F5 180 Ringing Proxy -> 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


   F6 200 OK B2 -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.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:7773660,phone-context=p1234@gw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: ...

   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


   F7 200 OK Proxy -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>



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   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:7773660,phone-
   context=p1234@gw1.wcom.com;user=phone>
   Content-Type: application/sdp
   Content-Length: ...

   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


   F8 ACK A -> Proxy

   ACK sip:7773660@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: < sip:7773660,phone-context=p1234@gw1.wcom.com;user=phone>
   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


   F9 ACK Proxy -> B4

   ACK sip:7773660@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F10 BYE A -> Proxy 1

   BYE sip:7773660@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: < sip:7773660,phone-context=p1234@gw1.wcom.com;user=phone>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345600@here.com
   CSeq: 2 BYE



Johnston, et al.            Informational                   [Page 223]


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   Content-Length: 0


   F11 BYE Proxy 1 -> B2

   BYE sip:7773660@gw1.wcom.com;phone-context=p1234 SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F12 200 OK B2 -> Proxy 1

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   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


   F13 200 OK Proxy 1 -> 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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7.6 Call Forwarding - Busy

         User A           Proxy          User B1             User B2
           |                |              |                   |
           |    INVITE F1   |              |                   |
           |--------------->|   INVITE F2  |                   |
           |                |------------->|                   |
           |(100 Trying) F3 |              |                   |
           |<---------------|  486 Busy F4 |                   |
           |                |<-------------|                   |
           |                |     ACK F5   |                   |
           |                |------------->|                   |
           |                |              |    INVITE F6      |
           |                |--------------------------------->|
           |                |              | 180 Ringing F7    |
           |                |<---------------------------------|
           | 180 Ringing F8 |              |                   |
           |<---------------|              |      200 OK F9    |
           |                |<---------------------------------|
           |    200 OK F10  |              |                   |
           |<---------------|              |                   |
           |     ACK F11    |              |                   |
           |--------------->|              |     ACK F12       |
           |                |--------------------------------->|
           |               Both way RTP Established            |
           |<=================================================>|
           |    BYE F13     |              |                   |
           |--------------->|              |      BYE F14      |
           |                |--------------------------------->|
           |                |              |    200 OK F15     |
           |  200 OK F16    |<---------------------------------|
           |<---------------|              |                   |
           |                |              |                   |

   User B wants calls to B1 forwarded to B2 if B1 is busy (this
   information is known to the proxy).  User A calls B1, B1 is busy, the
   proxy server places call to B2.


   Message Details


   F1 INVITE A -> Proxy

   INVITE sip:UserB@ss1.wcom.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



Johnston, et al.            Informational                   [Page 225]


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   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="9137d175c20a0d6eadd7be1c863302ae", opaque="",
   uri="sip:ss1.wcom.com", response="cf25aad811c806bde46a369220158cec"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 INVITE Proxy -> B1

   INVITE sip:UserB1@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 (100 Trying) Proxy -> 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 226]


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   F4 486 Busy Here B1 -> Proxy

   SIP/2.0 486 Busy Here
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 ACK Proxy -> B1

   ACK sip: UserB1@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F6 INVITE Proxy -> B2

   INVITE sip:UserB2@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F7 180 Ringing B4 -> Proxy

   SIP/2.0 180 Ringing



Johnston, et al.            Informational                   [Page 227]


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   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F8 180 Ringing Proxy -> 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=7654321
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F9 200 OK B2 -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB2@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client2.there.com
   s=Session SDP
   c=IN IP4 110.111.112.114
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F10 200 OK Proxy -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321



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   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB2@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client2.there.com
   s=Session SDP
   c=IN IP4 110.111.112.114
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F11 ACK A -> Proxy

   ACK sip:UserB2@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB2@there.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F12 ACK Proxy -> B2

   ACK sip: UserB2@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between A and B2 */

   /* User A eventually hangs up with User B2. */

   F13 BYE A -> Proxy

   BYE sip: UserB2@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:userB2@there.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321



Johnston, et al.            Informational                   [Page 229]


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


   F14 BYE Proxy -> B2

   BYE sip: UserB2@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F15 200 OK B2 -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F16 200 OK Proxy -> 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=7654321
   Call-ID: 12345600@here.com
   CSeq:  2 BYE
   Content-Length: 0















Johnston, et al.            Informational                   [Page 230]


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7.7 Call Forwarding - No Answer

         User A           Proxy          User B1             User B2
           |                |              |                   |
           |    INVITE F1   |              |                   |
           |--------------->|   INVITE F2  |                   |
           |(100 Trying) F3 |------------->|                   |
           |<---------------|180 Ringing F4|                   |
           | 180 Ringing F5 |<-------------|                   |
           |<---------------|              |                   |
           |                |Request Timeout                   |
           |                |   CANCEL F6  |                   |
           |                |------------->|                   |
           |                |   200 OK F7  |                   |
           |                |<-------------|                   |
           |                |              |    INVITE F8      |
           |                |--------------------------------->|
           |                |              | 180 Ringing F9    |
           |                |<---------------------------------|
           |                |              |      200 OK F10   |
           |                |<---------------------------------|
           |   200 OK F11   |              |                   |
           |<---------------|              |                   |
           |     ACK F12    |              |                   |
           |--------------->|              |     ACK F13       |
           |                |--------------------------------->|
           |               Both way RTP Established            |
           |<=================================================>|
           |    BYE F14     |              |                   |
           |--------------->|              |      BYE F15      |
           |                |--------------------------------->|
           |                |              |    200 OK F16     |
           |  200 OK F17    |<---------------------------------|
           |<---------------|              |                   |
           |                |              |                   |

   User B wants calls to B1 forwarded to B2 if B1 is not answered
   (information is known to the proxy server).  User A calls B1 and no
   one answers.  The proxy server then places the call to B2.


   Message Details


   F1 INVITE A -> Proxy

   INVITE sip:UserB@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>



Johnston, et al.            Informational                   [Page 231]


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   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="0d6eadd7be1c863302ae9137d175c20a", opaque="",
   uri="sip:ss1.wcom.com", response="6c0595216fad51f97f992a4479f4cc8e"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 INVITE Proxy -> B1

   INVITE sip:UserB1@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 (100 Trying) Proxy -> 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



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   F4 180 Ringing B1 -> Proxy

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=3145678
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 180 Ringing Proxy -> 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=3145678
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* B1 rings until a configurable timer expires in the Proxy. The
   Proxy sends Cancel and proceeds down the list of routes. */

   F6 CANCEL Proxy -> B1

   CANCEL sip:UserB1@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F7 200 OK B1 -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=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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   F8 INVITE Proxy -> B2

   INVITE sip:UserB4@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 180 Ringing B2 -> Proxy

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F10 200 OK B2 -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB2@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0



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   o=UserB 2890844527 2890844527 IN IP4 client2.there.com
   s=Session SDP
   c=IN IP4 110.111.112.114
   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:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB2@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client2.there.com
   s=Session SDP
   c=IN IP4 110.111.112.114
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F12 ACK A -> Proxy

   ACK sip:UserB2@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB2@there.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F13 ACK Proxy -> B2

   ACK sip: UserB2@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com



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   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between A and B2*/

   /* User A Hangs Up with User B2. */

   F14 BYE A -> Proxy

   BYE sip: UserB2@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:userB2@there.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F15 BYE Proxy -> B2

   BYE sip: UserB2@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F16 200 OK B2 -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F17 200 OK Proxy -> 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=123456



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


















































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7.8 3-way Conference

   User A calls User B, User B then invites user C to a 3-way call.
   User B will mix the audio streams (act as the conference bridge).  If
   user B drops out of the call then the entire call is dropped.  This
   is not a fully meshed conference, and does not make use of the
   concepts in the call control draft.

   The signaling for this scenario is as follows:  User A calls User B,
   this establishes the call between A and B.  User B calls User C, this
   establishes the call between B and C.  User B will mix the audio
   streams, sending media originating at A to C, and media originating
   at C to A.  There is no SIP signaling relationship between User A
   and User C.



7.9 Single Line Extension

   Single Line Extension (Sequential, First Wins implementation), a call
   will ring several extensions in sequence.  The extension to answer
   the call becomes the active set, no other sets may join the call.

   The signaling is described in Section 7.10 of this document.  It is
   anticipated that Single Line Extension will be associated with help
   desk/call center applications rather then individual users.  The
   signaling for this implementation of Single Line Extension and Find-
   Me is the same, the difference may be in the provisioning of the
   service.

   Note that the call flows for a Home Extension have not yet been
   designed.





















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7.10
    Find-Me

         User A           Proxy           User   User   User   User
                                           B1     B2     B3     B4
           |                |              |      |      |      |
           |    INVITE F1   |              |      |      |      |
           |--------------->|   INVITE F2  |      |      |      |
           |                |------------->|      |      |      |
           |(100 Trying) F3 |              |      |      |      |
           |<---------------|180 Ringing F4|      |      |      |
           |                |<-------------|      |      |      |
           | 180 Ringing F5 |              |      |      |      |
           |<---------------|              |      |      |      |
           |                |    Timeout   |      |      |      |
           |                |              |      |      |      |
           |                |   CANCEL F6  |      |      |      |
           |                |------------->|      |      |      |
           |                |   200 OK F7  |      |      |      |
           |                |<-------------|      |      |      |
           |                |     INVITE F8|      |      |      |
           |                |-------------------->|      |      |
           |                | 480 Not Logged In F9|      |      |
           |                |<--------------------|      |      |
           |                |              ACK F10|      |      |
           |                |-------------------->|      |      |
           |                |          INVITE F11        |      |
           |                |--------------------------->|      |
           |                |      486 Busy Here F12     |      |
           |                |<---------------------------|      |
           |                |            ACK F13         |      |
           |                |--------------------------->|      |
           |                |              INVITE F14           |
           |                |---------------------------------->|
           |                |            180 Ringing F15        |
           |                |<----------------------------------|
           |                |               200 OK F16          |
           |   200 OK F17   |<----------------------------------|
           |<---------------|                                   |
           |     ACK F18    |                                   |
           |--------------->|                 ACK F19           |
           |                |---------------------------------->|
           |               Both way RTP Established             |
           |<==================================================>|
           |                |                  BYE F20          |
           |     BYE F21    |<----------------------------------|
           |<---------------|                                   |
           |    200 OK F22  |                                   |
           |--------------->|                 200 OK F23        |
           |                |---------------------------------->|
           |                |                                   |



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   A call to a user will attempt to locate that user by calling
   locations from a list of contacts.  The  location to answer the call
   becomes the active set, no other sets may join the call.

   It is anticipated that the Find-me feature will be associated with
   individual users.  The signaling for the implementation of Single
   Line Extension and Find-Me is the same, the difference may be in the
   provisioning of the service.


   Message Details


   F1 INVITE A -> Proxy

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="dd7be1c863302a0d6eae9137d175c20a", opaque="",
   uri="sip:ss1.wcom.com", response="cc14961f4bd9d8103a69630097bdbd34"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t= 0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F2 INVITE Proxy -> B1

   INVITE sip:UserB1@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp



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   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t= 0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F3 (100 Trying) Proxy -> 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 180 Ringing B1 -> Proxy

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=123456
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F5 180 Ringing Proxy -> 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=123456
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   /* B1 rings for until a configurable timer in the Proxy expires. The
   Proxy then sends Cancel and proceeds down the list of routes. */

   F6 CANCEL Proxy -> B1




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   CANCEL sip:UserB1@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=1
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 CANCEL
   Content-Length: 0


   F7 200 OK B1 -> Proxy

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


   F8 INVITE Proxy -> B2

   INVITE sip:UserB2@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F9 480 Not Logged In B2 -> Proxy

   SIP/2.0 480 Not Logged In
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314756



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


   F10 ACK Proxy -> B2

   ACK sip: UserB2@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=2
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314756
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F11 INVITE Proxy -> B3

   INVITE sip:UserB3@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=3
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F12 486 Busy Here B3 -> Proxy

   SIP/2.0 486 Busy Here
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=3
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0





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   F13 ACK Proxy -> B3

   ACK sip: UserB3@ss1.wcom.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=3
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7654321
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   F14 INVITE Proxy -> B4

   INVITE sip:UserB4@ there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=4
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F15 180 Ringing B4 -> Proxy

   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=4
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7137136
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Content-Length: 0


   F16 200 OK B4 -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060;branch=4



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   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7137136
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB4@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client4.there.com
   s=Session SDP
   c=IN IP4 110.111.112.116
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F17 200 OK Proxy -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7137136
   Call-ID: 12345600@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB4@there.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserB 2890844527 2890844527 IN IP4 client4.there.com
   s=Session SDP
   c=IN IP4 110.111.112.116
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F18 ACK A -> Proxy

   ACK sip:UserB4@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <UserB4@there.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7137136
   Call-ID: 12345600@here.com
   CSeq: 1 ACK



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   Content-Length: 0


   F19 ACK Proxy -> B4

   ACK sip: UserB4@there.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=7137136
   Call-ID: 12345600@here.com
   CSeq: 1 ACK
   Content-Length: 0


   /* RTP streams are established between A and B4*/

   /* User B4 Hangs Up with User A. */

   F20 BYE B4 -> Proxy

   BYE sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   From: LittleGuy <sip:UserB@there.com>;tag=7137136
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F21 BYE Proxy -> A

   BYE sip: UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=7137136
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0


   F22 200 OK A -> Proxy

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP ss1.wcom.com:5060
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=7137136
   To: BigGuy <sip:UserA@here.com>;tag=1234567



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


   F23 200 OK Proxy -> B4

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP there.com:5060
   From: LittleGuy <sip:UserB@there.com>;tag=7137136
   To: BigGuy <sip:UserA@here.com>;tag=1234567
   Call-ID: 12345600@here.com
   CSeq: 2 BYE
   Content-Length: 0







































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7.11
    Call Management (Incoming Call Screening)

         User A           Proxy          User B
           |                |              |
           |    INVITE F1   |              |
           |--------------->|              |
           | 407 Proxy Authorization F2    |
           |<---------------|              |
           |     ACK F3     |              |
           |--------------->|              |
           |    INVITE F4   |              |
           |--------------->|              |
           | 403 Screening Failure (Terminating) F5
           |<---------------|              |
           |     ACK F3     |              |
           |--------------->|              |
           |                |              |

   User B has an incoming call screening list, User A is included on the
   list of addresses User B will not accept calls from.  User A attempts
   to call user B.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 challenges User A for authentication */

   F2 407 Proxy Authorization Required Proxy 1 -> A



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   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="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359",
   opaque="", stale="FALSE", algorithm="MD5"
   Content-Length:0


   F3 ACK A -> Proxy 1

   ACK sip:UserB@ss1.wcom.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


   /* User A responds by sending an INVITE with authentication
   credentials in it.  */

   F4 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="ae9137be1c87d175c2dd63302a0d6e0a", opaque="",
   uri="sip:ss1.wcom.com", response="bbaec39f943bdcb3620d90afc548a45c"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000




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   F5 403 Screening Failure (Terminating) Proxy 1 -> A

   SIP/2.0 403 Screening Failure (Terminating)
   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


   F6 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: 12345601@here.com
   CSeq: 1 ACK
   Content-Length:0
































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7.12
    Call Management (Outgoing Call Screening)

         User A           Proxy          User B
           |                |              |
           |    INVITE F1   |              |
           |--------------->|              |
           | 407 Proxy Authorization F2    |
           |<---------------|              |
           |     ACK F3     |              |
           |--------------->|              |
           |    INVITE F4   |              |
           |--------------->|              |
           | 403 Screening Failure (Originating) F5
           |<---------------|              |
           |     ACK F3     |              |
           |--------------->|              |
           |                |              |

   User A has an outgoing call screening list, User B is included on the
   list of addresses User A will not be able to place a call to.  User A
   attempts to call user B.


   Message Details


   F1 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   /* Proxy 1 challenges User A for authentication */

   F2 407 Proxy Authorization Required Proxy 1 -> A



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   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="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359",
   opaque="", stale="FALSE", algorithm="MD5"
   Content-Length:0


   F3 ACK A -> Proxy 1

   ACK sip:UserB@ss1.wcom.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


   /* User A responds be sending an INVITE with authentication
   credentials in it.  */

   F4 INVITE A -> Proxy 1

   INVITE sip:UserB@ss1.wcom.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: BigGuy <sip:UserA@here.com>
   Proxy-Authorization: DIGEST username="UserA", realm="MCI WorldCom
   SIP", nonce="cb360afc54bbaec39f943bd820d9a45c", opaque="",
   uri="sip:ss1.wcom.com", response="b9d2e5bcdec9f69ab2a9b44f270285a6"
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=UserA 2890844526 2890844526 IN IP4 client.here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=3034423619 0
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000




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   F5 403 Screening Failure (Originating) Proxy 1 -> A

   SIP/2.0 403 Screening Failure (Originating)
   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


   F6 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: 12345601@here.com
   CSeq: 1 ACK
   Content-Length:0
































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8  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.

8.1 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 ;



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     secondparam = secondvalue  ; q = 0.33
     (((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


8.2 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


8.3 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



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   Contact: Joe Bob Briggs <urn:ipaddr:122.1.2.3>

   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



8.4 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: Thu, 01 Dec 2040 16:00:00 GMT



8.5 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



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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



8.6 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>



8.7 Unknown Method Message

   This request message contains a new unknown method, NEWMETHOD.

   A proxy should forward this using the same retransmission rules as
   INVITE. 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



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



8.8 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



8.9 REGISTER with Unknown Authorization Scheme

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





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



8.10
    INVITE with Invalid SIP Version Number

   This message contains two requests, separated by a bunch of
   whitespace. The first request is a valid registration.  The second
   request is an INVITE with a wrong SIP version number (3.0).

   The server should accept the first REGISTER request.  The server
   should respond to the second request with a bad version error.


   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/3.0
   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



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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



8.11
    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
   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



8.12
    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



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   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



8.13
    INVITE with Illegal Expires Header

   This message contains an Expires header which has illegal values for
   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



8.14
    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.





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   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



8.15
    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





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8.16
    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

   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



8.17
    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




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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



8.18
    INVITE with Garbage after Message Body

   This is a request message with garbage after the end of the SDP
   included in the body. In fact, the server should treat this garbage
   as a second request. However, it is not even close to a valid
   message. The server should therefore ignore it, and forward the first
   message normally.


   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: 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''...'';;;;



8.19
    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



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   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



9  Acknowledgements


   The authors wish to thank the following individuals for their
   assistance and review of this call flows document: 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.


10 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.

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

   [5] S. Donovan, J. Hearty, M. Cannon, H. Schulzrinne, and J.
       Rosenberg, "SIP 183 Session Progress Message", Internet Draft,
       Internet Engineering Task Force, October 1999.  Work in
       progress.




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   [6] J. Rosenberg, and H. Schulzrinne, "Reliability of Provisional
       Responses in SIP", Internet Draft, Internet Engineering Task
       Force, January 2000,  Work in progress.

   [7] A. Vaha-Sipila, "URLs for Telephone Calls", Internet Draft,
       Internet Engineering Task Force, December 1999,  Work in
       progress.

   [8] S. Donovan, M. Cannon, "The SIP INFO Method", Internet Draft,
       Internet Engineering Task Force, January 2000.  Work in
       progress.

   [9] G. Camarillo, "Best Current Practice for ISUP to SIP Mapping",
       Internet Draft, Internet Engineering Task Force, October 1999,
       Work in progress.

   [10]H. Schulzrinne, and J. Rosenberg, "SIP Call Control Services",
       Internet Draft, Internet Engineering Task Force, June 1999, Work
       in progress.


































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Author's Addresses

   Alan Johnston
   MCI WorldCom
   100 S 4th Street
   St. Louis, MO  63104         Email:  alan.johnston@wcom.com
   USA

   Steve Donovan
   MCI WorldCom
   901 International Parkway
   Richardson, TX  65081        Email:  steven.r.donovan@wcom.com
   USA

   Robert Sparks
   MCI WorldCom
   2400 N Glenville Drive
   Richardson, TX  75082         Email:  robert.sparks@wcom.com
   USA

   Chris Cunningham
   MCI WorldCom
   400 International Parkway
   Richardson, TX  75081        Email:  chris.cunningham@wcom.com
   USA

   Kevin Summers
   MCI WorldCom
   2400 N Glenvile Drive
   Richardson, TX  75082        Email:  kevin.summers@wcom.com
   USA

   Dean Willis
   MCI WorldCom
   400 International Parkway
   Richardson, TX  75081        Email:  dean.willis@wcom.com
   USA

   Jonathan Rosenberg
   dynamicsoft
   200 Executive Drive
   Suite 120
   West Orange, NJ 07052        Email: jdrosen@dynamicsoft.com
   USA

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University 1214 Amsterdam Avenue, MC 0401
   New York, NY 10027           Email:  schulzrinne@cs.columbia.edu
   USA



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