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Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows
draft-ietf-sipping-pstn-call-flows-02

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 3666.
Authors Steve Donovan , Robert Sparks , Kevin Summers , Alan Johnston , Chris Cunningham
Last updated 2020-01-21 (Latest revision 2003-04-07)
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Best Current Practice
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IESG IESG state Became RFC 3666 (Best Current Practice)
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Responsible AD Allison J. Mankin
IESG note
Send notices to <rohan@cisco.com>, <dean.willis@softarmor.com>
draft-ietf-sipping-pstn-call-flows-02
SIPPING Working Group                                    A. Johnston 
   Internet Draft                                              WorldCom 
   Document: draft-ietf-sipping-pstn-call-flows-02.txt       S. Donovan 
   Expires: October 2003                                      R. Sparks 
                                                          C. Cunningham 
                                                            dynamicsoft 
                                                             K. Summers 
                                                                  Sonus 
                                                             April 2003 
    
    
                Session Initiation Protocol PSTN Call Flows 
    
    
Status of this Memo 
    
   This document is an Internet-Draft and is in full conformance with 
   all provisions of Section 10 of RFC2026.  
    
   Internet-Drafts are working documents of the Internet Engineering 
   Task Force (IETF), its areas, and its working groups.  Note that      
   other groups may also distribute working documents as Internet-
   Drafts. 
    
   Internet-Drafts are draft documents valid for a maximum of six months 
   and may be updated, replaced, or obsoleted by other documents at any 
   time.  It is inappropriate to use Internet-Drafts as reference 
   material or to cite them other than as "work in progress." 
    
   The list of current Internet-Drafts can be accessed at 
        http://www.ietf.org/ietf/1id-abstracts.txt 
   The list of Internet-Draft Shadow Directories can be accessed at 
        http://www.ietf.org/shadow.html. 
    
    
Abstract 
    
   This document contains best current practice examples of Session 
   Initiation Protocol (SIP) call flows showing interworking with the 
   Public Switched Telephone Network (PSTN).  Elements in these call 
   flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways.  
   Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP.  
   PSTN telephony protocols are illustrated using ISDN (Integrated 
   Services Digital Network), ISUP (ISDN User Part), and FGB (Feature 
   Group B) circuit associated signaling.  PSTN calls are illustrated 
   using global telephone numbers from the PSTN and private extensions 
   served on by a PBX (Private Branch Exchange).  Call flow diagrams and 
   message details are shown.    
    
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
Conventions used in this document 
    
   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in this 
   document are to be interpreted as described in RFC-2119 [1]. 
    
Table of Contents 
    
   1. Overview.......................................................2 
      1.1 General Assumptions........................................3 
      1.2 Legend for Message Flows...................................4 
      1.3 SIP Protocol Assumptions...................................5 
   2. SIP to PSTN Dialing............................................6 
      2.1 Successful SIP to ISUP PSTN call...........................7 
      2.2 Successful SIP to ISDN PBX call...........................15 
      2.3 Successful SIP to ISUP PSTN call with overflow............23 
      2.4 Session established using ENUM Query......................32 
      2.5 Unsuccessful SIP to PSTN call: Treatment from PSTN........38 
      2.6 Unsuccessful SIP to PSTN: REL w/Cause from PSTN...........45 
      2.7 Unsuccessful SIP to PSTN: ANM Timeout.....................50 
   3. PSTN to SIP Dialing...........................................56 
      3.1 Successful PSTN to SIP call...............................57 
      3.2 Successful PSTN to SIP call, Fast Answer..................64 
      3.3 Successful PBX to SIP call................................70 
      3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL.....77 
      3.5 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL......79 
      3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones...83 
      3.7 Unsuccessful PSTN->SIP, ACM timeout.......................87 
      3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy......91 
      3.9 Unsuccessful PSTN->SIP, Caller Abandonment................95 
   4. PSTN to PSTN Dialing via SIP Network.........................101 
      4.1 Successful ISUP PSTN to ISUP PSTN call...................102 
      4.2 Successful FGB PBX to ISDN PBX call with overflow........110 
   Security Considerations.........................................118 
   Normative References............................................120 
   Informative References..........................................120 
   Acknowledgments.................................................121 
   Author's Addresses..............................................121 
    
1.   Overview 
    
   The call flows shown in this document were developed in the design of 
   a SIP IP communications network.  They represent an example 
   minimum set of functionality. 
    
   It is the hope of the authors that this document will be useful for 
   SIP implementers, designers, and protocol researchers alike and will 
   help further the goal of a standard implementation of RFC 3261 [2].  
   These flows represent carefully checked and working group reviewed 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   scenarios of the most common SIP/PSTN interworking examples as a 
   companion to the specifications. 
    
   These call flows are based on the current version 2.0 of SIP in 
   RFC 3261 [2] with SDP usage described in RFC 3264 [3]. Other RFCs 
   also comprise the SIP standard but are not used in this set of basic 
   call flows. The SIP/ISUP mapping is based on RFC zzzz [4].  
    
   Various PSTN signaling protocols are illustrated in this document: 
   ISDN (Integrated Services Digital Network), ISUP (ISDN User 
   Part) and FGB (Feature Group B) circuit associated signaling.  This 
   document shows mainly ANSI ISUP due to its practical origins.  
   However, as used in this document, the usage is virtually identical 
   to the ITU-T International ISUP used as the reference in [4]. 
    
   Basic SIP call flow examples are contained in a companion document, 
   RFC yyyy [11]. 
    
1.1     General Assumptions 
    
   A number of architecture, network, and protocol assumptions underlie 
   the call flows in this document. Note that these assumptions are not 
   requirements.  They are outlined in this section so that they may be 
   taken into consideration and to aid in the understanding of the call 
   flow examples. 
    
   The authentication of SIP User Agents in these example call flows is 
   performed using SIP Digest as defined in [3] and [5]. 
    
   Some Proxy Servers in these call flows insert Record-Route headers 
   into requests to ensure that they are in the signaling path for 
   future message exchanges.   
    
   These flows show TLS, TCP, and UDP for transport.  SCTP [6] could 
   also be used.  See the discussion in RFC 3261 [2] for details on the 
   transport issues for SIP. 
    
   The SIP Proxy Server has access to a Location Service and other 
   databases.  Information present in the Request-URI and the context 
   (From header) is sufficient to determine to which proxy or gateway 
   the message should be routed.  In most cases, a primary and secondary 
   route will be determined in case of Proxy or Gateway failure 
   downstream. 
    
   Gateways provide tones (ringing, busy, etc) and announcements to the 
   PSTN side based on SIP response messages, or pass along audio in-band 
   tones (ringing, busy tone, etc.) in an early media stream to the SIP 
   side. 
 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   The interactions between the Proxy and Gateway can be summarized as 
   follows: 
     
     . The SIP Proxy Server performs digit analysis and lookup and 
       locates the correct gateway. 
    
     . The SIP Proxy Server performs gateway location based on primary 
       and secondary routing. 
 
   Telephone numbers are usually represented as SIP URIs.  Note that an 
   alternative is the use of the tel URI [7].  
    
   This document shows typical examples of SIP/ISUP interworking.  
   Although in the spirit of the SIP-T framework [8], these examples do 
   not represent a complete implementation of the framework.  The 
   examples here represent more of a minimal set of examples for very 
   basic SIP to ISUP interworking, rather than the more complex goal of 
   ISUP transparency.  In particular, there are NO examples of 
   encapsulated ISUP in this document.  If present, these messages would 
   show S/MIME encryption due to the sensitive nature of this 
   information, as discussed in the SIP-T Framework security 
   considerations section.  (Note - RFC 3204 [9] contains an example of 
   an INVITE with encapsulated ISUP.)  See the Security Considerations 
   section for a more detailed discussion on the security of these call 
   flows. 
    
   In ISUP, the Calling Party Number is abbreviated as CgPN and the 
   Called Party Number is abbreviated as CdPN.  Other abbreviations 
   include Numbering Plan Indicator (NPI) and Nature of Address (NOA). 
 
1.2     Legend for Message Flows 
    
   Dashed lines (---) represent signaling messages that are mandatory to 
   the call scenario. These messages can be SIP or PSTN 
   signaling.  The arrow indicates the direction of message flow. 
    
   Double dashed lines (===) represent media paths between network 
   elements. 
    
   Messages with parentheses around their name represent optional 
   messages. 
    
   Messages are identified in the Figures as F1, F2, etc.  This 
   references the message details in the list that follows the Figure. 
   Comments in the message details are shown in the following form: 
    
    /* Comments. */ 
    

 
 
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1.3     SIP Protocol Assumptions 
    
   This document does not prescribe the flows precisely as they are 
   shown, but rather the flows illustrate the principles for best 
   practice.  They are best practices usages (orderings, syntax, 
   selection of features for the purpose, handling of error) of SIP 
   methods, headers and parameters.  IMPORTANT: The exact flows here 
   must not be copied as is by an implementer due to specific incorrect 
   characteristics that were introduced into the document for 
   convenience and are listed below.  To sum up, the SIP/PSTN call flows 
   represent well-reviewed examples of SIP usage, which are best common 
   practice according to IETF consensus. 
    
   For simplicity in reading and editing the document, there are a 
   number of differences between some of the examples and actual SIP 
   messages.  For example, the SIP Digest responses are not actual MD5 
   encodings.  Call-IDs are often repeated, and CSeq counts often begin 
   at 1.  Header fields are usually shown in the same order.  Usually 
   only the minimum required header field set is shown, others that 
   would normally be present such as Accept, Supported, Allow, etc are 
   not shown. 
    
   Actors: 
    
   Element       Display Name   URI                        IP Address 
   -------       ------------   ---                        ---------- 
    
   User Agent    Alice          sip:alice@a.example.com    192.0.2.101 
   User Agent    Bob            sip:bob@b.example.com      192.0.2.200 
   Proxy Server                 sip:ss1.a.example.com      192.0.2.111 
   User Agent (Gateway)         sip:gw1.a.example.com      192.0.2.201 
   User Agent (Gateway)         sip:gw2.a.example.com      192.0.2.202 
   User Agent (Gateway)         sip:gw3.a.example.com      192.0.2.203 
   User Agent (Gateway)         sip:ngw1.a.example.com     192.0.2.103 
   User Agent (Gateway)         sip:ngw2.a.example.com     192.0.2.102 
    
   Note that NGW 1 and NGW 2 also have a device URIs (Contacts) of 
   sip:ngw1@a.example.com and sip:ngw2@a.example.com which resolves to 
   the Proxy Server sip:ss1.wcom.com using DNS SRV records. 
    

 
 
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2.   SIP to PSTN Dialing 
    
    
   In the following scenarios, Alice (Alice sip:alice@a.example.com) is 
   a SIP phone or other SIP-enabled device.  Bob is reachable via the 
   PSTN at global telephone number +19725552222. Alice places a call 
   to Bob through a Proxy Server Proxy 1 and a Network Gateway.  In 
   other scenarios, Alice places calls to Carol, who is served via a 
   PBX (Private Branch Exchange) and is identified by a private 
   extension 444-3333, or global number +1-918-555-3333.  Note that User 
   A uses his/her global telephone number +1-314-555-1111 in the From 
   header in the INVITE messages.  This then gives the Gateway the 
   option of using this header to populate the calling party 
   identification field in subsequent signaling. Left open is the issue 
   of how the Gateway can determine the accuracy of the telephone 
   number, necessary before passing it as a valid calling party number 
   in the PSTN. 
    
   In these scenarios, Alice is a SIP phone or other SIP-enabled 
   device.  Alice places a call to Bob in the PSTN or Carol on a 
   PBX through a Proxy Server and a Gateway. 
    
   In the failure scenarios, the call does not complete.  In some 
   cases, however, a media stream is still setup.  This is due to the 
   fact that some failures in dialing to the PSTN result in in-band 
   tones (busy, reorder tones or announcements - "The number you have 
   dialed has changed.  The new number is...").  The 183 Session 
   Progress response containing SDP media information is used to 
   setup this early media path so that the caller Alice knows the final 
   disposition of the call.   
    
   The media stream is either terminated by the caller after the tone or 
   announcement has been heard and understood, or by the Gateway after a 
   timer expires. 
    
   In other failure scenarios, a SS7 Release with Cause Code is mapped 
   to a SIP response.  In these scenarios, the early media path is not 
   used, but the actual failure code is conveyed to the caller by the 
   SIP User Agent Client. 
    
 

 
 
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2.1    Successful SIP to ISUP PSTN call 
    
   Alice           Proxy 1           NGW 1          Switch B 
     |                |                |                | 
     |   INVITE F1    |                |                | 
     |--------------->|                |                | 
     |     100  F2    |                |                | 
     |<---------------|   INVITE F3    |                | 
     |                |--------------->|                | 
     |                |     100  F4    |                | 
     |                |<---------------|     IAM F5     | 
     |                |                |--------------->| 
     |                |                |     ACM F6     | 
     |                |     183 F7     |<---------------| 
     |     183 F8     |<---------------|                | 
     |<---------------|                |                | 
     |        Both Way RTP Media       |  One Way Voice | 
     |<===============================>|<===============| 
     |                |                |      ANM F9    | 
     |                |    200 F10     |<---------------| 
     |     200 F11    |<---------------|                | 
     |<---------------|                |                | 
     |     ACK F12    |                |                | 
     |--------------->|     ACK F13    |                | 
     |                |--------------->|                | 
     |        Both Way RTP Media       | Both Way Voice | 
     |<===============================>|<==============>| 
     |     BYE F14    |                |                | 
     |--------------->|     BYE F15    |                | 
     |                |--------------->|                | 
     |                |     200 F16    |                | 
     |     200 F17    |<---------------|     REL F18    | 
     |<---------------|                |--------------->| 
     |                |                |     RLC F19    | 
     |                |                |<---------------| 
     |                |                |                | 
    
    
    
   Alice dials the globalized E.164 number +19725552222 to reach 
   Bob.  Note that A might have only dialed the last 7 digits, or 
   some other dialing plan.  It is assumed that the SIP User Agent 
   Client converts the digits into a global number and puts them into a 
   SIP URI.  Note that tel URIs could be used instead of SIP URIs. 
    
   Alice could use either their SIP address (sip:alice@a.example.com) or 
   SIP telephone number (sip:+13145551111@ss1.a.example.com;user=phone) 
   in the From header.  In this example, the telephone number is 
   included, and it is shown as being passed as calling party 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   identification through the Network Gateway (NGW 1) to Bob (F5).  Note 
   that for this number to be passed into the SS7 network, it would have 
   to be somehow verified for accuracy. 
    
   In this scenario, Bob answers the call then Alice disconnects the 
   call.  Signaling between NGW 1 and Bob's telephone switch is ANSI 
   ISUP.  For the details of SIP to ISUP mapping, refer to [4].   
    
   In this flow, notice that the Contact returned by NGW 1 in messages 
   F7-11 is sip:ngw1@a.example.com.  This is because NGW 1 only accepts 
   SIP messages that come through Proxy 1 - any direct signaling will be 
   ignored.  Since this Contact URI may be used outside of this dialog 
   and must be routable (Section 8.1.1.8 in RFC 3261 [2]) the Contact 
   URI for NGW 1 must resolve to Proxy 1.  This Contact URI is an AOR 
   which resolves via DNS to Proxy 1 (sip:ss1.a.example.com) which then 
   resolves it to sip:ngw1.a.example.com which is the address of NGW 1. 
    
   This flow shows TCP transport. 
    
    
   Message Details 
    
    
   F1 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Proxy-Authorization: Digest username="alice", realm="a.example.com", 
    nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="", 
    uri="sip:+19725552222@ss1.a.example.com;user=phone", 
    response="ccdca50cb091d587421457305d097458c" 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
   F2 100 Trying Proxy 1 -> Alice 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   /* Proxy 1 uses a Location Service function to determine the gateway 
   for terminating this call.  The call is forwarded to NGW 1.  Client 
   for A prepares to receive data on port 49172 from the 
   network.*/ 
    
   F3 INVITE Proxy 1 -> NGW 1 
    
   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying NGW 1 -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    ;received=192.0.2.111 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 IAM NGW 1 -> Bob 
    
   IAM 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
    
    
   F6 ACM Bob -> NGW 1 
    
   ACM 
    
    
   F7 183 Session Progress NGW 1 -> Proxy 1 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* NGW 1 sends PSTN audio (ringing) in the RTP path to A */ 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
   F8 183 Session Progress Proxy 1 -> Alice 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F9 ANM Bob -> NGW 1 
    
   ANM 
    
    
   F10 200 OK NGW 1 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
 
 
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   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F11 200 OK Proxy 1 -> Alice 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F12 ACK Alice -> Proxy 1 
    
   ACK sip:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 12] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F13 ACK Proxy 1 -> NGW 1 
    
   ACK sip:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
 
 
   /* Alice Hangs Up with Bob. */ 
    
   F14 BYE Alice -> Proxy 1 
    
   BYE sip:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F15 BYE Proxy 1 -> NGW 1 
    
   BYE sip:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 13] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F16 200 OK NGW 1 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F17 200 OK Proxy 1 -> A 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F18 REL NGW 1 -> B 
    
   REL 
   CauseCode=16 Normal 
    
    
   F19 RLC B -> NGW 1 
    
   RLC 
    
    
    
    
 

 
 
Johnston et al          Expires - October 2002               [Page 14] 


                         SIP PSTN Call Flows               April 2003 
 
 
2.2    Successful SIP to ISDN PBX call 
    
   Alice            Proxy 1           GW 1             PBX C 
     |                |                |                | 
     |   INVITE F1    |                |                | 
     |--------------->|                |                | 
     |     100  F2    |                |                | 
     |<---------------|   INVITE F3    |                | 
     |                |--------------->|                | 
     |                |     100  F4    |                | 
     |                |<---------------|    SETUP F5    | 
     |                |                |--------------->| 
     |                |                |  CALL PROC F6  | 
     |                |                |<---------------| 
     |                |                |   PROGress F7  | 
     |                |    180 F8      |<---------------| 
     |    180 F9      |<---------------|                | 
     |<---------------|                |                | 
     |                |                |  One Way Voice | 
     |                |                |<===============| 
     |                |                |   CONNect F10  | 
     |                |                |<---------------| 
     |                |                | CONNect ACK F11| 
     |                |    200 F12     |--------------->| 
     |     200 F13    |<---------------|                | 
     |<---------------|                |                | 
     |     ACK F14    |                |                | 
     |--------------->|     ACK F15    |                | 
     |                |--------------->|                | 
     |        Both Way RTP Media       | Both Way Voice | 
     |<===============================>|<==============>| 
     |     BYE F16    |                |                | 
     |--------------->|     BYE F17    |                | 
     |                |--------------->|                | 
     |                |     200 F18    |                | 
     |     200 F19    |<---------------| DISConnect F20 | 
     |<---------------|                |--------------->| 
     |                |                |   RELease F21  | 
     |                |                |<---------------| 
     |                |                | RELease COM F22| 
     |                |                |--------------->| 
     |                |                |                | 
    
   Alice is a SIP device while Carol is connected via a  
   Gateway (GW 1) to a PBX.  The PBX connection is via a ISDN trunk 
   group.  Alice dials Carol's telephone number (918-555-3333) which 
   is globalized and put into a SIP URI. 
    
   The host portion of the Request-URI in the INVITE F3 is used to 
 
 
Johnston et al          Expires - October 2002               [Page 15] 


                         SIP PSTN Call Flows               April 2003 
 
 
   identify the context (customer, trunk group, or line) in which the 
   private number 444-3333 is valid.  Otherwise, this INVITE message 
   could get forwarded by GW 1 and the context of the digits could 
   become lost and the call unroutable.   
    
   Proxy 1 looks up the telephone number and locates the gateway that 
   serves Carol.  Carolis identified by its extension 
   (444-3333) in the Request-URI sent to GW 1. 
    
   Note that the Contact URI for GW1 as used in messages F8, F9, F12, 
   and F13 is sips:4443333@gw1.a.example.com which does resolve directly 
   to the gateway. 
    
   This flow shows the use of Secure SIP (sips) URIs. 
    
 
   Message Details 
    
    
   F1 INVITE Alice -> Proxy 1 
    
   INVITE sips:+19185553333@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Contact: <sips:alice@client.a.example.com> 
   Proxy-Authorization: Digest username="alice", 
    realm="a.example.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",   
    opaque="", uri="sips:+19185553333@ss1.a.example.com;user=phone", 
    response="6c792f5c9fa360358b93c7fb826bf550" 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F2 100 Trying Proxy 1 -> Alice 
    
   SIP/2.0 100 Trying 
 
 
Johnston et al          Expires - October 2002               [Page 16] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Content-Length: 0 
    
    
   F3 INVITE Proxy 1 -> GW 1 
    
   INVITE sips:4443333@gw1.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Contact: <sips:alice@client.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying GW -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Content-Length: 0 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 17] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F5 SETUP GW 1 -> Carol 
    
   Protocol discriminator=Q.931 
   Message type=SETUP 
   Bearer capability: Information transfer capability=0 (Speech) or 16 
   (3.1 kHz audio) 
   Channel identification=Preferred or exclusive B-channel 
   Progress indicator=1 (Call is not end-to-end ISDN;further call 
   progress information may be available inband) 
   Called party number: 
   Type of number unknown 
   Digits=444-3333 
    
    
   F6 CALL PROCeeding Carol-> GW 1 
    
   Protocol discriminator=Q.931 
   Message type=CALL PROC 
   Channel identification=Exclusive B-channel 
    
    
   F7 PROGress Carol-> GW 1 
    
   Protocol discriminator=Q.931 
   Message type=PROG 
   Progress indicator=1 (Call is not end-to-end ISDN;further call 
   progress information may be available inband) 
    
    
   F8 180 Ringing GW 1 -> Proxy 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Contact: <sips:4443333@gw1.a.example.com> 
   Content-Length: 0 
    
 
   F9 180 Ringing Proxy 1 -> Alice 
    
 
 
Johnston et al          Expires - October 2002               [Page 18] 


                         SIP PSTN Call Flows               April 2003 
 
 
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Contact: <sips:4443333@gw1.a.example.com> 
   Content-Length: 0 
    
 
   F10 CONNect Carol-> GW 1 
    
   Protocol discriminator=Q.931 
   Message type=CONN 
    
    
   F11 CONNect ACK GW 1 -> Carol 
    
   Protocol discriminator=Q.931 
   Message type=CONN ACK 
    
    
   F12 200 OK GW 1 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Contact: <sips:4443333@gw1.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 144 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
 
 
Johnston et al          Expires - October 2002               [Page 19] 


                         SIP PSTN Call Flows               April 2003 
 
 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F13 200 OK Proxy 1 -> Alice 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>  
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 INVITE 
   Contact: <sips:4443333@gw1.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 144 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F14 ACK Alice -> Proxy 1 
    
   ACK sips:4443333@gw1.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   Route: <sips:ss1.a.example.com;lr> 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 ACK 
   Content-Length: 0 
    
    
   F15 ACK Proxy 1 -> GW 1 
    
   ACK sips:4443333@gw1.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
 
 
Johnston et al          Expires - October 2002               [Page 20] 


                         SIP PSTN Call Flows               April 2003 
 
 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 ACK 
   Content-Length: 0 
    
 
   /* Alice Hangs Up with Bob. */ 
    
   F16 BYE Alice -> Proxy 1 
    
   BYE sips:4443333@gw1.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   Route: <sips:ss1.a.example.com;lr> 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 3 BYE 
   Content-Length: 0 
    
    
   F17 BYE Proxy 1 -> GW 1 
    
   BYE sips:4443333@gw1.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 3 BYE 
   Content-Length: 0 
    
    
   F18 200 OK GW 1 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
 
 
Johnston et al          Expires - October 2002               [Page 21] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 3 BYE 
   Content-Length: 0 
    
    
   F19 200 OK Proxy 1 -> A 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 3 BYE 
   Content-Length: 0 
    
    
   F20 DISConnect GW 1 -> Carol 
    
   Protocol discriminator=Q.931 
   Message type=DISC 
   Cause=16 (Normal clearing) 
    
    
   F21 RELease Carol-> GW 1 
    
   Protocol discriminator=Q.931 
   Message type=REL 
    
    
   F22 RELease COMplete GW 1 -> Carol 
    
   Protocol discriminator=Q.931 
   Message type=REL COM 

 
 
Johnston et al          Expires - October 2002               [Page 22] 


                         SIP PSTN Call Flows               April 2003 
 
 
2.3    Successful SIP to ISUP PSTN call with overflow 
    
   Alice          Proxy 1         NGW 1          NGW 2        Switch B 
    |              |              |              |              | 
    |  INVITE F1   |              |              |              | 
    |------------->|              |              |              | 
    |              |  INVITE F2   |              |              | 
    |    100  F3   |------------->|              |              | 
    |<-------------|    503 F4    |              |              | 
    |              |<-------------|              |              | 
    |              |    ACK F5    |              |              | 
    |              |------------->|              |              | 
    |              |   INVITE F6                 |              | 
    |              |---------------------------->|     IAM F7   | 
    |              |                             |------------->| 
    |              |                             |     ACM F8   | 
    |              |            183 F9           |<-------------| 
    |   183 F10    |<----------------------------|              | 
    |<-------------|                             |              | 
    |               Two Way RTP Media            | One Way Voice| 
    |<==========================================>|<=============| 
    |              |                             |    ANM F11   | 
    |              |           200 F12           |<-------------| 
    |    200 F13   |<----------------------------|              | 
    |<-------------|                             |              | 
    |    ACK F14   |                             |              | 
    |------------->|            ACK F15          |              | 
    |              |---------------------------->|              | 
    |             Both Way RTP Media             |Both Way Voice| 
    |<==========================================>|<============>| 
    |    BYE F16   |                             |              | 
    |------------->|           BYE F17           |              | 
    |              |---------------------------->|              | 
    |              |           200 F18           |              | 
    |    200 F19   |<----------------------------|    REL F20   | 
    |<-------------|                             |------------->| 
    |              |                             |    RLC F21   | 
    |              |                             |<-------------| 
    |              |                             |              | 
    
   Alice calls Bob through Proxy 1.  Proxy 1 tries to route to a 
   Network Gateway NGW 1. NGW 1 is not available and responds with a 503 
   Service Unavailable (F4).  The call is then routed to Network Gateway 
   NGW 2.  Bob answers the call.  The call is terminated when Alice 
   disconnects the call.  NGW 2 and Bob's telephone switch use ANSI 
   ISUP signaling. 
    
   NGW 2 also only accepts SIP messages that come through Proxy 1, so 
   the Contact URI sip:ngw2@a.example.com is used in this flow. 
 
 
Johnston et al          Expires - October 2002               [Page 23] 


                         SIP PSTN Call Flows               April 2003 
 
 
    
   This flow shows UDP transport. 
    
    
   Message Details 
    
    
   F1 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com> 
   Proxy-Authorization: Digest username="alice", 
    realm="a.example.com", nonce="b59311c3ba05b401cf80b2a2c5ac51b0", 
    opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone", 
    response="ba6ab44923fa2614b28e3e3957789ab0" 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine where B is 
   located.  Proxy 1 receives a primary route NGW 1 and a secondary 
   route NGW 2.  NGW 1 is tried first */ 
    
   F2 INVITE Proxy 1 -> NGW 1 
    
   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
 
 
Johnston et al          Expires - October 2002               [Page 24] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F3 100 Trying Proxy 1 -> Alice 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F4 503 Service Unavailable NGW 1 -> Proxy 1 
    
   SIP/2.0 503 Service Unavailable 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 ACK Proxy 1 -> NGW 1 
    
 
 
Johnston et al          Expires - October 2002               [Page 25] 


                         SIP PSTN Call Flows               April 2003 
 
 
   ACK sip:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com>;user=phone> 
    ;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   /* Proxy 1 now tries secondary route to NGW 2 */ 
    
   F6 INVITE Proxy 1 -> NGW 2 
    
   INVITE sip:+19725552222@ngw2.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F7 IAM NGW 2 -> Bob 
    
   IAM 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
    
    
   F8 ACM Bob -> NGW 2 
    
 
 
Johnston et al          Expires - October 2002               [Page 26] 


                         SIP PSTN Call Flows               April 2003 
 
 
   ACM 
    
    
   F9 183 Session Progress NGW 2 -> Proxy 1 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw2@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com 
   s=- 
   c=IN IP4 ngw2.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* RTP packets are sent by GW to A for audio (e.g. ring tone) */ 
    
   F10 183 Session Progress Proxy 1 -> Alice 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw2@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
 
 
Johnston et al          Expires - October 2002               [Page 27] 


                         SIP PSTN Call Flows               April 2003 
 
 
   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com 
   s=- 
   c=IN IP4 ngw2.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F11 ANM Bob -> NGW 2 
    
   ANM 
    
    
   F12 200 OK NGW 2 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw2@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com 
   s=- 
   c=IN IP4 ngw2.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F13 200 OK Proxy 1 -> Alice 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
 
 
Johnston et al          Expires - October 2002               [Page 28] 


                         SIP PSTN Call Flows               April 2003 
 
 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw2@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com 
   s=- 
   c=IN IP4 ngw2.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F14 ACK Alice -> Proxy 1 
    
   ACK sip:ngw2@a.example.com SIP/2.0 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   Route: <ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F15 ACK Proxy 1 -> NGW 2 
    
   ACK sip:ngw2@a.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   /* RTP streams are established between A and B(via the GW) */ 
    
 
 
Johnston et al          Expires - October 2002               [Page 29] 


                         SIP PSTN Call Flows               April 2003 
 
 
   /* Alice Hangs Up with Bob. */ 
    
   F16 BYE Alice -> Proxy 1 
    
   BYE sip:ngw2@a.example.com SIP/2.0 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   Route: <ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F17 BYE Proxy 1 -> NGW 2 
    
   BYE sip:ngw2@a.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F18 200 OK NGW 2 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 30] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F19 200 OK Proxy 1 -> Alice 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F20 REL NGW 2 -> B 
    
   REL 
   CauseCode=16 Normal 
    
   F21 RLC B -> NGW 2 
    
   RLC 
 
    
    
  

 
 
Johnston et al          Expires - October 2002               [Page 31] 


                         SIP PSTN Call Flows               April 2003 
 
 
2.4    Successful SIP to SIP using ENUM Query 
     
   Alice         DNS Server         Proxy 3            Bob  
     |                |                |                |  
     |  ENUM Query F1 |                |                |  
     |--------------->|                |                |  
     |   Response F2  |                |                |  
     |<---------------|                |                |  
     |            INVITE F3            |                |  
     |-------------------------------->|    INVITE F4   |  
     |             100 F5              |--------------->|  
     |<--------------------------------|      180 F6    |  
     |             180 F7              |<---------------|  
     |<--------------------------------|                |  
     |                                 |     200 F8     |  
     |             200 F9              |<---------------|  
     |<--------------------------------|                |  
     |             ACK F10             |                |  
     |-------------------------------->|     ACK F11    |  
     |                                 |--------------->|  
     |                Both Way RTP Media                |  
     |<================================================>|  
     |                                 |     BYE F12    |  
     |             BYE F13             |<---------------|  
     |<--------------------------------|                |  
     |             200 F14             |                |  
     |-------------------------------->|     200 F15    |  
     |                                 |--------------->|  
     |                                 |                |  
     
   In this scenario, Alice places a call to Bob by dialing Bob's 
   telephone number (9725552222).  Alice's UA converts the phone number 
   to an E.164 number (+19725552222) performs an ENUM query [10] on the 
   E.164 number (2.2.2.2.5.5.5.2.7.9.1.e164.arpa) which returns a NAPTR 
   record containing a SIP AOR URI for Bob 
   (sip:+19725552222@b.example.com).  As a result, Alice's UA sends an 
   INVITE and the call completes over IP bypassing the PSTN.  
     
   The call is terminated when Bob sends a BYE message.  
     
     
   Message Details  
     
    
   F1 ENUM Query Alice -> DNS Server 
    
   2.2.2.2.5.5.5.2.7.9.1.e164.arpa 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 32] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F2 ENUM NAPTR Set DNS Server -> Alice 
    
   $ORIGIN 2.2.2.2.5.5.5.2.7.9.1.e164.arpa. 
         IN NAPTR 100 10 "u" "sip+E2U"  
                "!^.*$!sip:+19725552222@b.example.com!". 
    
    
   F3 INVITE Alice -> Proxy 3  
     
   INVITE sip:+19725552222@b.example.com SIP/2.0  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9  
   Max-Forwards: 70  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 INVITE  
   Contact: <sip:+13145551111@client.a.example.com>  
   Content-Type: application/sdp  
   Content-Length: 154 
     
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
     
    
   F4 INVITE Proxy 3 -> Bob  
     
   INVITE sip:+19725552222@client.b.example.com SIP/2.0  
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9  
    ;received=192.0.2.101  
   Max-Forwards: 69  
   Record-Route: <sip:ss3.b.example.com;lr>  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 INVITE  
   Contact: <sip:+13145551111@client.a.example.com>  
   Content-Type: application/sdp  
   Content-Length: 154  
     
   v=0  
   o=UserA 2890844526 2890844526 IN IP4 client.a.example.com  
   s=-  
   c=IN IP4 client.a.example.com  
 
 
Johnston et al          Expires - October 2002               [Page 33] 


                         SIP PSTN Call Flows               April 2003 
 
 
   t=0 0  
   m=audio 49172 RTP/AVP 0  
   a=rtpmap:0 PCMU/8000  
     
     
   F5 100 Trying Proxy 3 -> Alice  
     
   SIP/2.0 100 Trying  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9  
    ;received=192.0.2.101  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 INVITE  
   Content-Length: 0  
     
     
   F6 180 Ringing B -> Proxy 3  
     
   SIP/2.0 180 Ringing  
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1  
    ;received=192.0.2.233  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9  
    ;received=192.0.2.101  
   Record-Route: <sip:ss3.b.example.com;lr>  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>;tag=314159  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 INVITE  
   Contact: <sip:+19725552222@client.b.example.com>  
   Content-Length: 0  
     
     
   F7 180 Ringing Proxy 3 -> Alice  
     
   SIP/2.0 180 Ringing  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9  
    ;received=192.0.2.101  
   Record-Route: <sip:ss3.b.example.com;lr>  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>;tag=314159  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 INVITE  
   Contact: <sip:+19725552222@client.b.example.com>  
   Content-Length: 0  
     
     
   F8 200 OK Bob -> Proxy 3  
     
 
 
Johnston et al          Expires - October 2002               [Page 34] 


                         SIP PSTN Call Flows               April 2003 
 
 
   SIP/2.0 200 OK  
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1  
    ;received=192.0.2.233  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9  
    ;received=192.0.2.101  
   Record-Route: <sip:ss3.b.example.com;lr>  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>;tag=314159  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 INVITE  
   Contact: <sip:+19725552222@client.b.example.com;transport=tcp>  
   Content-Type: application/sdp  
   Content-Length: 151 
     
   v=0 
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com 
   s=- 
   c=IN IP4 client.b.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
     
     
   F9 200 OK Proxy -> Alice  
     
   SIP/2.0 200 OK  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9  
    ;received=192.0.2.101  
   Record-Route: <sip:ss3.b.example.com;lr>  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>;tag=314159  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 INVITE  
   Contact: <sip:+19725552222@client.b.example.com>  
   Content-Type: application/sdp  
   Content-Length: 151  
     
   v=0  
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com  
   s=-  
   c=IN IP4 192.0.2.100  
   t=0 0  
   m=audio 3456 RTP/AVP 0  
   a=rtpmap:0 PCMU/8000  
     
     
   F10 ACK Alice -> Proxy 3  
     
   ACK sip:+19725552222@client.b.example.com SIP/2.0  
 
 
Johnston et al          Expires - October 2002               [Page 35] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9  
   Max-Forwards: 70  
   Route: <sip:ss3.b.example.com;lr>  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>;tag=314159  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 ACK  
   Content-Length: 0  
     
     
   F11 ACK Proxy 3 -> Bob  
     
   ACK sip:+19725552222@client.b.example.com SIP/2.0  
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1  
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9  
    ;received=192.0.2.101  
   Max-Forwards: 69  
   From: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   To: <sip:+19725552222@b.example.com>;tag=314159  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 2 ACK  
   Content-Type: application/sdp  
   Content-Length: 0  
     
    
   /* RTP streams are established between A and B*/  
     
   /* User B Hangs Up with User A. */  
     
   F12 BYE Bob -> Proxy 3  
     
   BYE sip:+13145551111@client.a.example.com SIP/2.0  
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2  
   Max-Forwards: 70  
   Route: <sip:ss3.b.example.com;lr>  
   From: <sip:+19725552222@b.example.com>;tag=314159  
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 1 BYE  
   Content-Length: 0  
     
     
   F13 BYE Proxy 3 -> Alice  
     
   BYE sip:+13145551111@client.a.example.com SIP/2.0  
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1  
    ;received=192.0.2.100  
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2  
   Max-Forwards: 69  
 
 
Johnston et al          Expires - October 2002               [Page 36] 


                         SIP PSTN Call Flows               April 2003 
 
 
   From: <sip:+19725552222@b.example.com>;tag=314159  
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 1 BYE  
   Content-Length: 0  
     
     
   F14 200 OK Alice -> Proxy 3  
     
   SIP/2.0 200 OK  
   Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1  
    ;received=192.0.2.233  
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2  
    ;received=192.0.2.100  
   From: <sip:+19725552222@b.example.com>;tag=314159  
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 1 BYE  
   Content-Length: 0  
    
     
   F15 200 OK Proxy 3 -> Bob  
     
   SIP/2.0 200 OK  
   Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2  
    ;received=192.0.2.100  
   From: <sip:+19725552222@b.example.com>;tag=314159  
   To: <sip:+13145551111@a.example.com>;tag=9fxced76sl  
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com  
   CSeq: 1 BYE  
   Content-Length: 0  
     
    
    
    
    
    
    
    
    
    
    
    
    
    

 
 
Johnston et al          Expires - October 2002               [Page 37] 


                         SIP PSTN Call Flows               April 2003 
 
 
2.5    Unsuccessful SIP to PSTN call: Treatment from PSTN 
    
   Alice            Proxy 1           NGW 1            Bob  
     |                |                |                | 
     |   INVITE F1    |                |                | 
     |--------------->|                |                | 
     |     100  F2    |                |                | 
     |<---------------|   INVITE F3    |                | 
     |                |--------------->|                | 
     |                |     100  F4    |                | 
     |                |<---------------|     IAM F5     | 
     |                |                |--------------->| 
     |                |                |     ACM F6     | 
     |                |     183 F7     |<---------------| 
     |     183 F8     |<---------------|                | 
     |<---------------|                |                | 
     |         Two Way RTP Media       |  One Way Voice | 
     |<===============================>|<===============| 
     |                 Treatment Applied                | 
     |<=================================================| 
     |   CANCEL F9    |                |                | 
     |--------------->|                |                | 
     |     200 F10    |                |                | 
     |<---------------|   CANCEL F11   |                | 
     |                |--------------->|                | 
     |                |     200 F12    |                | 
     |                |<---------------|     REL F13    | 
     |                |                |--------------->| 
     |                |                |     RLC F14    | 
     |                |     487 F15    |<---------------| 
     |                |<---------------|                | 
     |                |     ACK F16    |                | 
     |     487 F17    |--------------->|                | 
     |<---------------|                |                | 
     |     ACK F18    |                |                | 
     |--------------->|                |                | 
     |                |                |                | 
    
   Alice calls Bob in the PSTN through a proxy server Proxy 1 and a 
   Network Gateway NGW 1.  The call is rejected by the PSTN with an in- 
   band treatment (tone or recording) played.  Alice hears the 
   treatment and then hangs up, which results in a CANCEL (F9) being 
   sent to terminate the call. (A BYE is not sent since no final 
   response was ever received by Alice.) 
    
    
   Message Details 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 38] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F1 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Proxy-Authorization: Digest username="alice", 
    realm="a.example.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40", 
    opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone", 
    response="e178fbe430e6680a1690261af8831f40" 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F2 100 Trying Proxy 1 -> A 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   /* Proxy 1 uses a Location Service function to determine where B is 
   located.  Based upon location analysis the call is forwarded to NGW 
   1.  Client for A prepares to receive data on port 49172 from the 
   network. */ 
    
   F3 INVITE Proxy 1 -> NGW 1 
    
   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
 
 
Johnston et al          Expires - October 2002               [Page 39] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying NGW 1 -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 IAM NGW 1 -> Bob 
    
   IAM 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
    
    
   F6 ACM Bob -> NGW 1 
    
   ACM 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 40] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F7 183 Session Progress NGW 1 -> Proxy 1 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F8 183 Session Progress Proxy 1 -> Alice 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>   
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
 
 
Johnston et al          Expires - October 2002               [Page 41] 


                         SIP PSTN Call Flows               April 2003 
 
 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Caller hears the recorded announcement, then hangs up */ 
    
   F9 CANCEL Alice -> Proxy 1 
    
   CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F10 200 OK Proxy 1 -> A 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F11 CANCEL Proxy 1 -> NGW 1 
    
   CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F12 200 OK NGW 1 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
 
 
Johnston et al          Expires - October 2002               [Page 42] 


                         SIP PSTN Call Flows               April 2003 
 
 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F13 REL NGW 1 -> B 
    
   REL 
   CauseCode=18 No user responding 
    
    
   F14 RLC B -> NGW 1 
    
   RLC 
    
    
   F15 487 Request Terminated NGW 1 -> Proxy 1 
    
   SIP/2.0 487 Request Terminated 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F16 ACK Proxy 1 -> NGW 1 
    
   ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
 
 
Johnston et al          Expires - October 2002               [Page 43] 


                         SIP PSTN Call Flows               April 2003 
 
 
   F17 487 Request Terminated Proxy 1 -> A 
    
   SIP/2.0 487 Request Terminated 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F18 ACK Alice -> Proxy 1 
    
   ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
 
    

 
 
Johnston et al          Expires - October 2002               [Page 44] 


                         SIP PSTN Call Flows               April 2003 
 
 
2.6    Unsuccessful SIP to PSTN: REL w/Cause from PSTN 
    
   Alice            Proxy 1           NGW 1           Switch B 
     |                |                |                | 
     |   INVITE F1    |                |                | 
     |--------------->|                |                | 
     |     100  F2    |                |                | 
     |<---------------|   INVITE F3    |                | 
     |                |--------------->|                | 
     |                |     100  F4    |                | 
     |                |<---------------|     IAM F5     | 
     |                |                |--------------->| 
     |                |                |    REL(1) F6   | 
     |                |                |<---------------| 
     |                |                |     RLC F7     | 
     |                |     404 F8     |--------------->| 
     |                |<---------------|                | 
     |                |     ACK F9     |                | 
     |                |--------------->|                | 
     |     404 F10    |                |                | 
     |<---------------|                |                | 
     |     ACK F11    |                |                | 
     |--------------->|                |                | 
     |                |                |                | 
    
   Alice calls PSTN Bob through a Proxy Server Proxy 1 and a Network 
   Gateway NGW 1.  The call is rejected by the PSTN with a 
   ANSI ISUP Release message REL containing a specific Cause code. 
   This cause value (1) is mapped by the Gateway to a SIP 404 Address 
   Incomplete response which is proxied back to Alice.  For more 
   details of ISUP cause value to SIP response mapping refer to [4]. 
    
    
   Message Details 
    
    
   F1 INVITE Alice -> Proxy 1 
    
   INVITE sip:+44-1234@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Proxy-Authorization: Digest username="alice", 
    realm="a.example.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",  
 
 
Johnston et al          Expires - October 2002               [Page 45] 


                         SIP PSTN Call Flows               April 2003 
 
 
    opaque="", uri="sip:+44-1234@ss1.a.example.com;user=phone", 
    response="a451358d46b55512863efe1dccaa2f42" 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F2 100 Trying Proxy 1 -> A 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   /* Proxy 1 uses a Location Service function to determine where B is 
   located.  Based upon location analysis the call is forwarded to NGW1. 
   Client for A prepares to receive data on port 49172 from the network. 
   */ 
    
   F3 INVITE Proxy 1 -> NGW 1 
    
   INVITE sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 154 
    
 
 
Johnston et al          Expires - October 2002               [Page 46] 


                         SIP PSTN Call Flows               April 2003 
 
 
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying NGW 1 -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 IAM NGW 1 -> Bob 
    
   IAM 
   CdPN=44-1234,NPI=E.164,NOA=International 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
    
    
   F6 REL Bob -> NGW 1 
    
   REL 
   CauseValue=1 Unallocated number 
    
    
   F7 RLC NGW 1 -> Bob 
    
   RLC 
    
    
   /* Network Gateway maps CauseValue=1 to the SIP message 404 Not  
      Found */ 
    
   F8 404 Not Found NGW 1 -> Proxy 1 
    
   SIP/2.0 404 Not Found 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Error-Info: <sip:not-found-ann@ann.a.example.com> 
   Content-Length: 0 
    
    
   F9 ACK Proxy 1 -> NGW 1 
    
   ACK sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0 
   Max-Forwards: 70 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F10 404 Not Found Proxy 1 -> Alice 
    
   SIP/2.0 404 Not Found 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Error-Info: <sip:not-found-ann@ann.a.example.com> 
   Content-Length: 0 
    
    
   F11 ACK Alice -> Proxy 1 
    
   ACK sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
 
 
Johnston et al          Expires - October 2002               [Page 48] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Content-Length: 0 
    
    
 
    
    

 
 
Johnston et al          Expires - October 2002               [Page 49] 


                         SIP PSTN Call Flows               April 2003 
 
 
2.7    Unsuccessful SIP to PSTN: ANM Timeout 
    
   Alice           Proxy 1           NGW 1           Switch B 
     |                |                |                | 
     |   INVITE F1    |                |                | 
     |--------------->|                |                | 
     |     100  F2    |                |                | 
     |<---------------|   INVITE F3    |                | 
     |                |--------------->|                | 
     |                |     100  F4    |                | 
     |                |<---------------|     IAM F5     | 
     |                |                |--------------->| 
     |                |                |     ACM F6     | 
     |                |      183 F7    |<---------------| 
     |     183 F8     |<---------------|                | 
     |<---------------|                |                | 
     |                |      Timer on NGW 1 Expires     | 
     |                |                |                | 
     |                |                |     REL F9     | 
     |                |                |--------------->| 
     |                |                |    RLC F10     | 
     |                |     480 F11    |<---------------| 
     |                |<---------------|                | 
     |                |     ACK F12    |                | 
     |                |--------------->|                | 
     |     480 F13    |                |                | 
     |<---------------|                |                | 
     |     ACK F14    |                |                | 
     |--------------->|                |                | 
    
   Alice calls Bob in the PSTN through a proxy server Proxy 1 and 
   Network Gateway NGW 1.  The call is released by the Gateway after a 
   timer expires due to no ANswer Message (ANM) being received.  The 
   Gateway sends an ISUP Release REL message to the PSTN and a 480 
   Temporarily Unavailable response to Alice in the SIP network. 
    
    
   Message Details 
    
    
   F1 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   CSeq: 1 INVITE 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Proxy-Authorization: Digest username="alice", 
    realm="a.example.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40", 
    opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone", 
    response="579cb9db184cdc25bf816f37cbc03c7d" 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine where B is 
   located.  Based upon location analysis the call is forwarded to NGW 
   1.  Client for A prepares to receive data on port 49172 from the 
   network.*/ 
    
   F2 100 Trying Proxy 1 -> A 
    
   SIP/2.0  100 Trying 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F3 INVITE Proxy 1 -> NGW 1 
    
   INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Contact: <sip:alice@client.a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 154 
    
   v=0 
   o=alice 2890844526 2890844526 IN IP4 client.a.example.com 
   s=- 
   c=IN IP4 client.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying NGW 1 -> Proxy 1 
    
   SIP/2.0  100 Trying 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 IAM NGW 1 -> Bob 
    
   IAM 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
    
    
   F6 ACM Bob -> NGW 1 
    
   ACM 
    
    
   F7 183 Session Progress NGW 1 -> Proxy 1 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F8 183 Session Progress Proxy 1 -> Alice 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* After NGW 1's timer expires, Network Gateway sends REL to ISUP 
   network and 480 to SIP network */ 
    
   F9 REL NGW 1 -> Bob 
    
   REL 
 
 
Johnston et al          Expires - October 2002               [Page 53] 


                         SIP PSTN Call Flows               April 2003 
 
 
   CauseCode=18 No user responding 
    
    
   F10 RLC Bob -> NGW 1 
    
   RLC 
    
    
   F11 480 Temporarily Unavailable NGW 1 -> Proxy 1 
    
   SIP/2.0 480 Temporarily Unavailable 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 INVITE 
   Error-Info: <sip:temp-unavail-ann@ann.a.example.com> 
   Content-Length: 0 
    
    
   F12 ACK Proxy 1 -> NGW 1 
    
   ACK sip:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Max-Forwards: 70 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F13 480 Temporarily Unavailable F13 Proxy 1 -> Alice 
    
   SIP/2.0 480 Temporarily Unavailable 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
    ;received=192.0.2.101 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   CSeq: 1 INVITE 
   Error-Info: <sip:temp-unavail-ann@ann.a.example.com> 
   Content-Length: 0 
    
    
   F14 ACK Alice -> Proxy 1 
    
   ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Max-Forwards: 70 
   Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 
   From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> 
    ;tag=9fxced76sl 
   To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> 
    ;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
 

 
 
Johnston et al          Expires - October 2002               [Page 55] 


                         SIP PSTN Call Flows               April 2003 
 
 
3.   PSTN to SIP Dialing 
    
    
   In these scenarios, Alice is placing calls from the PSTN to Bob 
   in a SIP network.  Alice's telephone switch signals to a Network 
   Gateway (NGW 1) using ANSI ISUP. 
    
   Since the called SIP User Agent does not send in-band signaling 
   information, no early media path needs to be established on the IP 
   side.  As a result, the 183 Session Progress response is not used. 
   However, NGW 1 will establish a one way speech path prior to call 
   completion, and generate ringing for the PSTN caller.  Any tones or 
   recordings are generated by NGW 1 and played in this speech path. 
   When the call completes successfully, NGW 1 bridges the PSTN speech 
   path with the IP media path.   
    
   To reduce the number of messages, only a single proxy server is shown 
   in these flows, which means that the a.example.com proxy server has 
   access to the b.example.com location service. 
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
3.1    Successful PSTN to SIP call 
    
   Switch A          NGW 1          Proxy 1           Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |     100  F4    |--------------->| 
     |                |<---------------|                | 
     |                |                |      180 F5    | 
     |                |    180 F6      |<---------------| 
     |     ACM F7     |<---------------|                | 
     |<---------------|                |                | 
     |  One Way Voice |                |                | 
     |<===============|                |                | 
     |  Ringing Tone  |                |      200 F8    | 
     |<===============|    200 F9      |<---------------| 
     |                |<---------------|                | 
     |                |     ACK F10    |                | 
     |     ANM F12    |--------------->|     ACK F11    | 
     |<---------------|                |--------------->| 
     | Both Way Voice |        Both Way RTP Media       | 
     |<==============>|<===============================>| 
     |     REL F13    |                |                | 
     |--------------->|                |                | 
     |     RLC F14    |                |                | 
     |<---------------|     BYE F15    |                | 
     |                |--------------->|     BYE F16    | 
     |                |                |--------------->| 
     |                |                |     200 F17    | 
     |                |     200 F18    |<---------------| 
     |                |<---------------|                | 
     |                |                |                | 
    
   In this scenario, Alice from the PSTN calls Bob through a Network 
   Gateway NGW1 and Proxy Server Proxy 1.  When Bob answers the call 
   the media path is setup end-to-end. The call terminates when Alice 
   hangs up the call, with Alice's telephone switch sending an ISUP 
   RELease message which is mapped to a BYE by NGW 1. 
    
   Message Details 
    
    
   F1 IAM Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
   F2 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine where B is 
   located.  Based upon location analysis the call is forwarded to NGW 
   1.  NGW 1  prepares to receive data on port 3456 from Alice.*/ 
    
   F3 INVITE Proxy 1 -> Bob 
    
   INVITE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying Bob -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 180 Ringing Bob -> Proxy 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com> 
   Content-Length: 0 
    
    
   F6 180 Ringing Proxy 1 -> NGW 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com> 
   Content-Length: 0 
    
    
   F7 ACM NGW 1 -> Alice 
    
 
 
Johnston et al          Expires - October 2002               [Page 59] 


                         SIP PSTN Call Flows               April 2003 
 
 
   ACM 
    
    
   F8 200 OK Bob -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   Contact: <sip:bob@client.b.example.com> 
   CSeq: 1 INVITE 
   Content-Type: application/sdp 
   Content-Length: 151 
    
   v=0 
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com 
   s=- 
   c=IN IP4 client.b.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F9 200 OK Proxy 1 -> NGW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com> 
   Content-Type: application/sdp 
   Content-Length: 151 
    
   v=0 
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com 
   s=- 
   c=IN IP4 client.b.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   F10 ACK NGW 1 -> Proxy 1 
    
   ACK sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F11 ACK Proxy 1 -> Bob 
    
   ACK sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F12 ANM Bob -> NGW 1 
    
   ANM 
    
    
   /* RTP streams are established between A and B (via the GW) */ 
    
   /* Alice Hangs Up with Bob. */ 
    
   F13 REL Alice -> NGW 1 
    
   REL 
   CauseCode=16 Normal 
    
    
   F14 RLC NGW 1 -> Alice 
    
   RLC 
    
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   F15 BYE NGW 1-> Proxy 1 
    
   BYE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F16 BYE Proxy 1 -> Bob 
    
   BYE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F17 200 OK Bob -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F18 200 OK Proxy 1 -> NGW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   CSeq: 2 BYE 
   Content-Length: 0 
    

 
 
Johnston et al          Expires - October 2002               [Page 63] 


                         SIP PSTN Call Flows               April 2003 
 
 
3.2    Successful PSTN to SIP call, Fast Answer 
    
   Switch A           NGW 1          Proxy 1           Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |     100  F4    |--------------->| 
     |                |<---------------|                | 
     |                |                |      200 F5    | 
     |                |     200 F6     |<---------------| 
     |                |<---------------|                | 
     |                |     ACK F7     |                | 
     |     ANM F9     |--------------->|     ACK F8     | 
     |<---------------|                |--------------->| 
     | Both Way Voice |        Both Way RTP Media       | 
     |<==============>|<===============================>| 
     |     REL F10    |                |                | 
     |--------------->|                |                | 
     |     RLC F11    |                |                | 
     |<---------------|     BYE F12    |                | 
     |                |--------------->|     BYE F13    | 
     |                |                |--------------->| 
     |                |                |     200 F14    | 
     |                |     200 F15    |<---------------| 
     |                |<---------------|                | 
     |                |                |                | 
    
   This "fast answer" scenario is similar to 3.1 except that Bob 
   immediately accepts the call, sending a 200 OK (F5) without sending a 
   180 Ringing response.  The Gateway then sends an Answer Message (ANM) 
   without sending an Address Complete Message (ACM).  Note that for 
   ETSI and some other ISUP variants, a CONnect message (CON) would be 
   sent instead of the ANM. 
    
   Message Details 
    
    
   F1 IAM Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
    
    
   F2 INVITE NGW 1 -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine where B is 
   located.  Based upon location analysis the call is forwarded to User 
   B.  Bob  prepares to receive data on port 3456 from Alice.*/ 
    
   F3 INVITE Proxy 1 -> Bob 
    
   INVITE bob@b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying Proxy 1 -> NGW 1 
    
 
 
Johnston et al          Expires - October 2002               [Page 65] 


                         SIP PSTN Call Flows               April 2003 
 
 
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 200 OK Bob -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 151 
    
   v=0 
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com 
   s=- 
   c=IN IP4 client.b.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F6 200 OK Proxy 1 -> NGW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 151 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   v=0 
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com 
   s=- 
   c=IN IP4 client.b.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F7 ACK NGW 1 -> Proxy 1 
    
   ACK bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F8 ACK Proxy 1 -> Bob 
    
   ACK bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=130.131.132.14 
   Max-Forwards: 69 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F9 ANM Bob -> NGW 1 
    
   ANM 
    
    
   /* RTP streams are established between A and B (via the GW) */ 
    
   /* Alice Hangs Up with Bob. */ 
    
   F10 REL ser Alice -> NGW 1 
    
   REL 
   CauseCode=16 Normal 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   F11 RLC NGW 1 -> Alice 
    
   RLC 
    
    
   F12 BYE NGW 1 -> Proxy 1 
    
   BYE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F13 BYE Proxy 1 -> Bob 
    
   BYE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F14 200 OK Bob -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F15 200 OK Proxy 1 -> NGW 1 
 
 
Johnston et al          Expires - October 2002               [Page 68] 


                         SIP PSTN Call Flows               April 2003 
 
 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    

 
 
Johnston et al          Expires - October 2002               [Page 69] 


                         SIP PSTN Call Flows               April 2003 
 
 
3.3    Successful PBX to SIP call 
    
   PBX A            GW 1           Proxy 1           Bob  
     |                |                |                | 
     |    Seizure     |                |                | 
     |--------------->|                |                | 
     |      Wink      |                |                | 
     |<---------------|                |                | 
     |  MF Digits F1  |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |     100  F4    |--------------->| 
     |                |<---------------|                | 
     |                |                |      180 F5    | 
     |                |    180 F6      |<---------------| 
     |                |<---------------|                | 
     |  One Way Voice |                |                | 
     |<===============|                |                | 
     |  Ringing Tone  |                |      200 F7    | 
     |<===============|     200 F8     |<---------------| 
     |                |<---------------|                | 
     |                |     ACK F9     |                | 
     |     Seizure    |--------------->|     ACK F10    | 
     |<---------------|                |--------------->| 
     | Both Way Voice |        Both Way RTP Media       | 
     |<==============>|<===============================>| 
     | Seizure Removal|                |                | 
     |--------------->|                |                | 
     | Seizure Removal|                |                | 
     |<---------------|     BYE F11    |                | 
     |                |--------------->|     BYE F12    | 
     |                |                |--------------->| 
     |                |                |     200 F13    | 
     |                |     200 F14    |<---------------| 
     |                |<---------------|                | 
     |                |                |                | 
    
   In this scenario, Alice dials from PBX A to Bob through GW 1 and 
   Proxy 1.  This is an example of a call that appears destined for the 
   PSTN but instead is routed to a SIP Client. 
    
   Signaling between PBX A and GW 1 is Feature Group B (FGB) circuit 
   associated signaling, in-band Mult-Frequency (MF) outpulsing.  After 
   the receipt of the 180 Ringing from Bob, GW 1 generates ringing 
   tone for Alice. 
    
   Bob answers the call by sending a 200 OK.  The call terminates 
   when Alice hangs up, causing GW1 to send a BYE. 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   The  Gateway can only identify the trunk group that the 
   call came in on, it cannot identify the individual line on PBX A that 
   is placing the call.  The SIP URI used to identify the caller is 
   shown in these flows as sip:551313@gw1.a.example.com.   
    
   Message Details 
    
 
   PBX Alice -> GW 1 
    
   Seizure 
    
    
   GW 1 -> PBX A 
    
   Wink 
    
    
   F1 MF Digits PBX Alice -> GW 1 
    
   KP 1 972 555 2222 ST 
    
    
   F2 INVITE GW 1 -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
   Max-Forwards: 70 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:551313@gw1.a.example.com;user=phone> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine where the 
   phone number +19725552222 is located.  Based upon location 
   analysis the call is forwarded to SIP Bob. */ 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   F3 INVITE Proxy 1 -> Bob 
    
   INVITE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:551313@gw1.a.example.com;user=phone> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying Proxy 1 -> GW 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 180 Ringing Bob -> Proxy 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Contact: <sip:bob@client.b.example.com> 
   Content-Length: 0 
    
    
   F6 180 Ringing Proxy 1 -> GW 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com> 
   Content-Length: 0 
    
    
   /* One way Voice path is established between GW and the PBX for 
   ringing. */ 
    
   F7 200 OK Bob -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   Contact: <sip:bob@client.b.example.com> 
   CSeq: 1 INVITE 
   Content-Type: application/sdp 
   Content-Length: 151 
    
   v=0 
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com 
   s=- 
   c=IN IP4 client.b.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F8 200 OK Proxy 1 -> GW 1 
    
   SIP/2.0 200 OK 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com> 
   Content-Type: application/sdp 
   Content-Length: 151 
    
   v=0 
   o=bob 2890844527 2890844527 IN IP4 client.b.example.com 
   s=- 
   c=IN IP4 client.b.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F9 ACK GW 1 -> Proxy 1 
    
   ACK sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F10 ACK Proxy 1 -> Bob 
    
   ACK sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   /* RTP streams are established between A and B (via the GW) */ 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   /* Alice Hangs Up with Bob. */ 
    
   F11 BYE GW 1 -> Proxy 1 
    
   BYE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F12 BYE Proxy 1 -> Bob 
    
   BYE sip:bob@client.b.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F13 200 OK Bob -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
    
   F14 200 OK Proxy 1 -> GW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
 
 
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   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 2 BYE 
   Content-Length: 0 
    
 
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
3.4    Unsuccessful PSTN to SIP REL, SIP error mapped to REL 
    
   Switch A            GW 1          Proxy 1           Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|                | 
     |                |     604 F3     |                | 
     |                |<---------------|                | 
     |                |     ACK F4     |                | 
     |                |--------------->|                | 
     |     REL F5     |                |                | 
     |<---------------|                |                | 
     |     RLC F6     |                |                | 
     |--------------->|                |                | 
     |                |                |                | 
    
   Alice attempts to place a call through Gateway GW 1 and Proxy 1, 
   which is unable to find any routing for the number.  The call is 
   rejected by Proxy 1 with a REL message containing a specific Cause 
   value mapped by the gateway based on the SIP error. 
    
   Message Details 
    
    
   F1 IAM Alice -> GW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=972-555-9999,NPI=E.164,NOA=National 
    
    
   F2 INVITE Alice -> Proxy 1 
    
   INVITE sip:+1972559999@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s 
   To: <sip:+1972559999@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: 
   <sip:+13145551111@gw1.a.example.com;user=phone;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 144 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com 
   s=- 
 
 
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   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service to find a route to +1-972-555- 
   9999.  A route is not found, so Proxy 1 rejects the call. */ 
    
   F3 604 Does Not Exist Anywhere Proxy 1 -> GW 1 
    
   SIP/2.0 604 Does Not Exist Anywhere 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s 
   To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 INVITE 
   Error-Info: <sip:does-not-exist@ann.a.example.com> 
   Content-Length: 0 
    
    
   F4 ACK GW 1 -> Proxy 1 
    
   ACK sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s 
   To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410 
   Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F5 REL GW 1 -> Alice 
    
   REL 
   CauseCode=1 
    
    
   F6 RLC Alice -> GW 1 
    
   RLC 
 
    
    
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
3.5    Unsuccessful PSTN to SIP REL, SIP busy mapped to REL 
    
   Switch A          NGW 1           Proxy 1          Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |     100  F4    |--------------->| 
     |                |<---------------|                | 
     |                |                |      600 F5    | 
     |                |                |<---------------| 
     |                |                |      ACK F6    | 
     |                |     600 F7     |--------------->| 
     |                |<---------------|                | 
     |                |     ACK F8     |                | 
     |                |--------------->|                | 
     |   REL(17) F9   |                |                | 
     |<---------------|                |                | 
     |     RLC F10    |                |                | 
     |<-------------->|                |                | 
     |                |                |                | 
    
   In this scenario, Alice calls Bob through Network Gateway NGW 1 
   and Proxy 1.  The call is routed to Bob by Proxy 1.  The call is 
   rejected by Bob who sends a 600 Busy Everywhere response.  The 
   Gateway sends a REL message containing a specific Cause value mapped 
   by the gateway based on the SIP error. 
    
   Since no interworking is indicated in the IAM (F1), the busy tone is 
   generated locally by Alice's telephone switch.  In some scenarios, 
   the busy signal is generated by the Gateway since interworking is 
   indicated.  For more discussion on interworking, refer to [4]. 
    
    
   Message Details 
    
    
   F1 IAM Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
    
    
   F2 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 144 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine a route for 
   +19725552222.  The call is then forwarded to Bob. */ 
    
   F3 INVITE F3 Proxy 1 -> Bob 
    
   INVITE bob@b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 144 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying Proxy 1 -> NGW 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    ;received=192.0.2.201 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 600 Busy Everywhere Bob -> Proxy 1 
    
   SIP/2.0 600 Busy Everywhere 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F6 ACK Proxy 1 -> Bob 
    
   ACK bob@b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F7 600 Busy Everywhere Proxy 1 -> NGW 1 
    
   SIP/2.0 600 Busy Everywhere 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F8 ACK NGW 1 -> Proxy 1 
    
   ACK bob@b.example.com SIP/2.0 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F9 REL NGW 1 -> Alice 
    
   REL 
   CauseCode=17 Busy 
    
    
   F10 RLC Alice -> NGW 1 
    
   RLC 
    
    
    
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
3.6    Unsuccessful PSTN->SIP, SIP error interworking to tones 
    
   Switch A          NGW 1           Proxy 1          Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |     100  F4    |--------------->| 
     |                |<---------------|                | 
     |                |                |      600 F5    | 
     |                |                |<---------------| 
     |                |                |      ACK F6    | 
     |                |     600 F7     |--------------->| 
     |                |<---------------|                | 
     |                |     ACK F8     |                | 
     |     ACM F9     |--------------->|                | 
     |<---------------|                |                | 
     | One Way Voice  |                |                | 
     |<===============|                |                | 
     |    Busy Tone   |                |                | 
     |<===============|                |                | 
     |   REL(16) F10  |                |                | 
     |--------------->|                |                | 
     |     RLC F11    |                |                | 
     |<---------------|                |                | 
     |                |                |                | 
    
    
   In this scenario, Alice calls Bob through Network Gateway NGW1 
   and Proxy 1.  The call is routed to Bob by Proxy 1.  The call is 
   rejected by the Bob client.  NGW 1 sets up a two way voice path to 
   Alice and plays busy tone.  The caller then disconnects 
    
   NGW 1 plays the busy tone since the IAM (F1) indicates the 
   interworking is present.  In scenario 5.2.2, with no interworking, 
   the busy indication is carried in the REL Cause value and is 
   generated locally instead. 
    
   Again, note that for ETSI or ITU ISUP, a CONnect message would be 
   sent instead of the Answer Message. 
    
    
   Message Details 
    
    
   F1 IAM Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
   Interworking=encountered 
    
    
   F2 INVITE NGW1 -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine a route for 
   +19725552222.  The call is then forwarded to Bob. */ 
    
   F3 INVITE Proxy 1 -> Bob 
    
   INVITE bob@b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying Bob -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 600 Busy Everywhere Bob -> Proxy 1 
    
   SIP/2.0 600 Busy Everywhere 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F6 ACK Proxy 1 -> Bob 
    
   ACK bob@b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F7 600 Busy Everywhere Proxy 1 -> NGW 1 
    
   SIP/2.0 600 Busy Everywhere 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F8 ACK NGW 1 -> Proxy 1 
    
   ACK sip:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F9 ACM NGW 1 -> Alice 
    
   ACM 
    
    
   /* A one way speech path is established between NGW 1 and Alice. */ 
    
   /* Call Released after Alice hangs up. */ 
    
   F10 REL Alice -> NGW 1 
    
   REL 
   CauseCode=16 
    
    
   F11 RLC NGW 1 -> Alice 
    
   RLC 
    
 
    

 
 
Johnston et al          Expires - October 2002               [Page 86] 


                         SIP PSTN Call Flows               April 2003 
 
 
3.7    Unsuccessful PSTN->SIP, ACM timeout 
    
   Switch A          NGW 1           Proxy 1          Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |     100  F4    |--------------->| 
     |                |<---------------|                | 
     |                |                |   INVITE F5    | 
     |                |                |--------------->| 
     |                |                |   INVITE F6    | 
     |                |                |--------------->| 
     |                |                |   INVITE F7    | 
     |                |                |--------------->| 
     |                |                |   INVITE F8    | 
     |                |                |--------------->| 
     |                |                |   INVITE F9    | 
     |                |                |--------------->| 
     |     REL F10    |                |                | 
     |--------------->|                |                | 
     |     RLC F11    |                |                | 
     |<---------------|                |                | 
     |                |   CANCEL F12   |                | 
     |                |--------------->|                | 
     |                |     200 F13    |                | 
     |                |<---------------|                | 
    
   Alice calls Bob through NGW 1 and Proxy 1.  Proxy 1 re-sends the 
   INVITE after the expiration of SIP timer T1 without receiving any 
   response from Bob.  Bob never responds with 180 Ringing or any 
   other response (it is reachable but unresponsive).  After the 
   expiration of a timer, Alice's network disconnects the call by 
   sending a Release message REL.  The Gateway maps this to a CANCEL. 
   Message Details 
 
   F1 IAM Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
    
   F2 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
 
 
Johnston et al          Expires - October 2002               [Page 87] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine a route for 
   +19725552222.  The call is then forwarded to Bob. */ 
    
   F3 INVITE Proxy 1 -> Bob 
    
   INVITE sip:bob@b.example.com  SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   c c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying Proxy 1 -> NGW 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
 
 
Johnston et al          Expires - October 2002               [Page 88] 


                         SIP PSTN Call Flows               April 2003 
 
 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   F6 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   F7 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   F8 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   F9 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   /* Timer expires in Alice's access network. */ 
    
   F10 REL Alice -> NGW 1 
    
   REL 
   CauseCode=16 Normal 
    
    
   F11 RLC NGW 1 -> Alice 
    
   RLC 
    
    
   F12 CANCEL NGW 1 -> Proxy 1 
    
   CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
 
 
Johnston et al          Expires - October 2002               [Page 89] 


                         SIP PSTN Call Flows               April 2003 
 
 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F13 200 OK Proxy 1 -> NGW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    

 
 
Johnston et al          Expires - October 2002               [Page 90] 


                         SIP PSTN Call Flows               April 2003 
 
 
3.8    Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy 
    
   Switch A          NGW 1      Stateless Proxy 1     Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |   INVITE F4    |--------------->| 
     |                |--------------->|   INVITE F5    | 
     |                |   INVITE F6    |--------------->| 
     |                |--------------->|   INVITE F7    | 
     |                |   INVITE F8    |--------------->| 
     |                |--------------->|   INVITE F9    | 
     |                |   INVITE F10   |--------------->| 
     |                |--------------->|   INVITE F11   | 
     |                |   INVITE F12   |--------------->| 
     |                |--------------->|   INVITE F13   | 
     |                |                |--------------->| 
     |     REL F14    |                |                | 
     |--------------->|                |                | 
     |     RLC F15    |                |                | 
     |<---------------|                |                | 
    
   In this scenario, Alice calls Bob through NGW 1 and Proxy 1. 
   Since Proxy 1 is stateless (it does not send a 100 Trying response), 
   NGW 1 re-sends the INVITE message after the expiration of 
   SIP timer T1.  Bob does not respond with 180 Ringing.  Alice's 
   network disconnects the call with a release REL (CauseCode=102 
   Timeout). 
    
    
   Message Details 
    
    
   F1 IAM Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
    
    
   F2 INVITE NGW 1 -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
 
 
Johnston et al          Expires - October 2002               [Page 91] 


                         SIP PSTN Call Flows               April 2003 
 
 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine a route for 
   +19725552222.  The call is then forwarded to Bob. */ 
    
   F3 INVITE Proxy 1 -> Bob 
    
   INVITE sip:bob@b.example.com  SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 INVITE NGW 1 -> Proxy 1 
    
   Same as Message F2 
    
    
   F5 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
 
 
Johnston et al          Expires - October 2002               [Page 92] 


                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   F6 INVITE NGW 1 -> Proxy 1 
    
   Same as Message F2 
    
    
   F7 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   F8 INVITE NGW 1 -> Proxy 1 
    
   Same as Message F2 
    
    
   F9 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   F10 INVITE NGW 1 -> Proxy 1 
    
   Same as Message F2 
    
    
   F11 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   F12 INVITE NGW 1 -> Proxy 1 
    
   Same as Message F2 
    
    
   F13 INVITE Proxy 1 -> Bob 
    
   Same as Message F3 
    
    
   /* A timer expires in Alice's access network. */ 
    
   F14 REL Alice -> NGW 1 
    
   REL 
   CauseCode=102 Timeout 
    
 
 
Johnston et al          Expires - October 2002               [Page 93] 


                         SIP PSTN Call Flows               April 2003 
 
 
    
   F15 RLC NGW 1 -> Alice 
    
   RLC 
    
    
 
    

 
 
Johnston et al          Expires - October 2002               [Page 94] 


                         SIP PSTN Call Flows               April 2003 
 
 
3.9    Unsuccessful PSTN->SIP, Caller Abandonment 
    
   Switch A          NGW 1          Proxy 1           Bob  
     |                |                |                | 
     |     IAM F1     |                |                | 
     |--------------->|   INVITE F2    |                | 
     |                |--------------->|   INVITE F3    | 
     |                |     100  F4    |--------------->| 
     |                |<---------------|                | 
     |                |                |      180 F5    | 
     |                |    180 F6      |<---------------| 
     |     ACM F7     |<---------------|                | 
     |<---------------|                |                | 
     |  One Way Voice |                |                | 
     |<===============|                |                | 
     |  Ringing Tone  |                |                | 
     |<===============|                |                | 
     |                |                |                | 
     |     REL F8     |                |                | 
     |--------------->|                |                | 
     |     RLC F9     |                |                | 
     |<---------------|   CANCEL F10   |                | 
     |                |--------------->|                | 
     |                |     200 F11    |                | 
     |                |<---------------|                | 
     |                |                |   CANCEL F12   | 
     |                |                |--------------->| 
     |                |                |     200 F13    | 
     |                |                |<---------------| 
     |                |                |     487 F14    | 
     |                |                |<---------------| 
     |                |                |     ACK F15    | 
     |                |     487 F16    |--------------->| 
     |                |<---------------|                | 
     |                |     ACK F17    |                | 
     |                |--------------->|                | 
     |                |                |                | 
    
    
   In this scenario, Alice calls Bob through NGW 1 and Proxy 1. 
   Bob does not respond with 200 OK.  NGW 1 plays ringing tone since 
   the ACM indicates that interworking has been encountered.  Alice 
   disconnects the call with a Release message REL which is mapped by 
   NGW 1 to a CANCEL.  Note that if Bob had sent a 200 OK response 
   after the REL, NGW 1 would have sent an ACK then a BYE to properly 
   terminate the call. 
    
    
   Message Details 
 
 
Johnston et al          Expires - October 2002               [Page 95] 


                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   F1 IAM Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=972-555-2222,NPI=E.164,NOA=National 
    
    
   F2 INVITE Alice -> Proxy 1 
    
   INVITE sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 uses a Location Service function to determine a route for 
   +19725552222.  The call is then forwarded to Bob. */ 
    
   F3 INVITE Proxy 1 -> Bob 
    
   INVITE sip:bob@b.example.com  SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:ngw1@a.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 146 
 
 
Johnston et al          Expires - October 2002               [Page 96] 


                         SIP PSTN Call Flows               April 2003 
 
 
    
   v=0 
   o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 100 Trying Bob -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.201 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 180 Ringing Bob -> Proxy 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com;transport=tcp> 
   Content-Length: 0 
    
    
   F6 180 Ringing Proxy 1 -> NGW 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
 
 
Johnston et al          Expires - October 2002               [Page 97] 


                         SIP PSTN Call Flows               April 2003 
 
 
   CSeq: 1 INVITE 
   Contact: <sip:bob@client.b.example.com> 
   Content-Length: 0 
    
    
   F7 ACM NGW 1 -> Alice 
    
   ACM 
    
    
   /* Alice hangs up */ 
    
   F8 REL Alice -> NGW 1 
    
   REL 
   CauseCode=16 Normal 
    
    
   F9 RLC NGW 1 -> Alice 
    
   RLC 
    
    
   F10 CANCEL NGW 1 -> Proxy 1 
    
   CANCEL sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F11 200 OK Proxy 1 -> NGW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F12 CANCEL Proxy 1 -> Bob 
    
 
 
Johnston et al          Expires - October 2002               [Page 98] 


                         SIP PSTN Call Flows               April 2003 
 
 
   CANCEL sip:bob@b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F13 200 OK Bob -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone> 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 CANCEL 
   Content-Length: 0 
    
    
   F14 487 Request Terminated Bob -> Proxy 1 
    
   SIP/2.0 487 Request Terminated 
   Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F15 ACK Proxy 1 -> Bob 
    
   ACK sip:bob@b.example.com SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F16 487 Request Terminated Proxy 1 -> NGW 1 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
   SIP/2.0 487 Request Terminated 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F17 ACK NGW 1 -> Proxy 1 
    
   ACK sip:+19725552222@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
 
    
    
    
    
    
    
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
4.   PSTN to PSTN Dialing via SIP Network 
    
   In these scenarios, both the caller and the called party are in the 
   telephone network, either normal PSTN subscribers or PBX extensions. 
   The calls route through two Gateways and at least one SIP Proxy 
   Server.  The Proxy Server performs the authentication and location of 
   the Gateways. 
    
   Again it is noted that the intent of this call flows document is not 
   to provide a detailed parameter level mapping of SIP to PSTN 
   protocols.  For information on SIP to ISUP mapping, the reader is 
   referred to other references [4]. 
    
   In these scenarios, the call is successfully completed between the 
   two Gateways allowing the PSTN or PBX users to communicate.  The 183 
   Session Progress response is used to indicate in-band alerting may 
   flow from the called party telephone switch to the caller. 
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
4.1    Successful ISUP PSTN to ISUP PSTN call 
    
   Switch A       NGW 1         Proxy 1         GW 2         Switch C 
    |              |              |              |              | 
    |     IAM F1   |              |              |              | 
    |------------->|              |              |              | 
    |              |  INVITE F2   |              |              | 
    |              |------------->|  INVITE F3   |              | 
    |              |              |------------->|     IAM F4   | 
    |              |              |              |------------->| 
    |              |              |              |     ACM F5   | 
    |              |              |   183 F6     |<-------------| 
    |              |    183 F7    |<-------------|              | 
    |    ACM F8    |<-------------|              |              | 
    |<-------------|              |              |              | 
    | One Way Voice|      Two Way RTP Media      | One Way Voice| 
    |<=============|<===========================>|<=============| 
    |              |              |              |    ANM F9    | 
    |              |              |   200 F10    |<-------------| 
    |              |    200 F11   |<-------------|              | 
    |    ANM F12   |<-------------|              |              | 
    |<-------------|              |              |              | 
    |              |    ACK F13   |              |              | 
    |              |------------->|    ACK F14   |              | 
    |              |              |------------->|              | 
    |Both Way Voice|     Both Way RTP Media      |Both Way Voice| 
    |<=============|<===========================>|<=============| 
    |              |              |              |    REL F15   | 
    |              |              |              |<-------------| 
    |              |              |   BYE F16    |              | 
    |              |    BYE F18   |<-------------|    RLC F17   | 
    |              |<-------------|              |------------->| 
    |              |              |              |              | 
    |              |    200 F19   |              |              | 
    |              |------------->|    200 F20   |              | 
    |              |              |------------->|              | 
    |    REL F21   |              |              |              | 
    |<-------------|              |              |              | 
    |    RLC F22   |              |              |              | 
    |------------->|              |              |              | 
    |              |              |              |              | 
    
    
   In this scenario, Alice in the PSTN calls Carol who is an extension 
   on a PBX.  Alice's telephone switch signals via SS7 to the Network 
   Gateway NGW 1, while Carol's PBX signals via SS7 with the  
   Gateway GW 2.  The CdPN and CgPN are mapped by GW1 into SIP URIs and 
   placed in the To and From headers.  Proxy 1 looks up the dialed 
   digits in the Request-URI and maps the digits to the PBX extension of 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Carol which is served by GW 2.  The Proxy in F3 uses the host portion 
   of the Request-URI to identify what private dialing plan is being 
   referenced. The INVITE is then forwarded to GW 2 for call completion.  
   An early media path is established end-to-end so that Alice can hear 
   the ringing tone generated by PBX C. 
    
   Carol answers the call and the media path is cut through in both 
   directions.  Bob hangs up terminating the call. 
    
   Message Details 
    
    
   F1 IAM Switch Alice -> NGW 1 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=918-555-3333,NPI=E.164,NOA=National 
    
    
   F2 INVITE NGW 1 -> Proxy 1 
    
   INVITE sips:+19185553333@ss1.a.example.com;user=phone  SIP/2.0 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sips:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* Proxy 1 consults Location Service and translates the dialed number 
   to a private number in the Request-URI*/ 
    
   F3 INVITE Proxy 1 -> GW 2 
    
   INVITE sips:4443333@gw2.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKwqwee65 
 
 
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    ;received=192.0.2.103 
   Max-Forwards: 69 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone> 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sips:ngw1@a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 146 
    
   v=0 
   o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com 
   s=- 
   c=IN IP4 ngw1.a.example.com 
   t=0 0 
   m=audio 3456 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 IAM GW 2 -> Switch C 
    
   IAM 
   CgPN=314-555-1111,NPI=E.164,NOA=National 
   CdPN=444-3333,NPI=Private,NOA=Subscriber 
    
    
   F5 ACM Switch C -> GW 2 
    
   ACM 
    
    
   /* Based on the ACM message, GW 2 returns a 183 response.  In-band 
   call progress indications are sent to Alice through NGW 1. */ 
    
   F6 183 Session Progress GW 2 -> Proxy 1 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sips:4443333@gw2.a.example.com> 
   Content-Type: application/sdp 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Content-Length: 143 
    
   v=0 
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com 
   s=- 
   c=IN IP4 gw2.a.example.com 
   t=0 0 
   m=audio 14918 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F7 183 Session Progress Proxy 1 -> GW 1 
    
   SIP/2.0 183 Session Progress 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sips:4443333@gw2.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 143 
    
   v=0 
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com 
   s=- 
   c=IN IP4 gw2.a.example.com 
   t=0 0 
   m=audio 14918 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   /* NGW 1 receives packets from GW 2 with encoded ringback, tones or 
   other audio.  NGW 1 decodes this and places it on the originating 
   trunk. */ 
    
   F8 ACM NGW 1 -> Switch A 
    
   ACM 
    
    
   /* Bob answers */ 
    
   F9 ANM Switch C -> GW 2 
    
   ANM 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
   F10 200 OK GW 2 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sips:4443333@gw2.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 143 
    
   v=0 
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com 
   s=- 
   c=IN IP4 gw2.a.example.com 
   t=0 0 
   m=audio 14918 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F11 200 OK Proxy 1 -> NGW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Record-Route: <sips:ss1.a.example.com;lr> 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sips:4443333@gw2.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 143 
    
   v=0 
   o=GW 987654321 987654321 IN IP4 gw2.a.example.com 
   s=- 
   c=IN IP4 gw2.a.example.com 
   t=0 0 
   m=audio 14918 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
   F12 ANM NGW 1 -> Switch A 
    
   ANM 
    
    
   F13 ACK NGW 1 -> Proxy 1 
    
   ACK sips:4443333@gw2.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 
   Max-Forwards: 70 
   Route: <sips:ss1.a.example.com;lr> 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F14 ACK Proxy 1 -> GW 2 
    
   ACK sips:4443333@gw2.a.example.com SIP/2.0 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 
    ;received=192.0.2.103 
   Max-Forwards: 69 
   From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   /* RTP streams are established between NGW 1 and GW 2. */ 
    
   /* Bob Hangs Up with Alice. */ 
    
   F15 REL Switch C -> GW 2 
    
   REL 
   CauseCode=16 Normal 
    
    
   F16 BYE GW 2 -> Proxy 1 
    
   BYE sips:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6 
   Max-Forwards: 70 
   Route: <sips:ss1.a.example.com;lr> 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 4 BYE 
   Content-Length: 0 
    
    
   F17 RLC GW 2 -> Switch C 
    
   RLC 
    
    
   F18 BYE Proxy 1 -> NGW 1 
    
   BYE sips:ngw1@a.example.com SIP/2.0 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6 
    ;received=192.0.2.202 
   Max-Forwards: 69 
   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 4 BYE 
   Content-Length: 0 
    
    
   F19 200 OK NGW 1 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6 
    ;received=192.0.2.202 
   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 4 BYE 
   Content-Length: 0 
    
    
   F20 200 OK Proxy 1 -> GW 2 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6 
    ;received=192.0.2.202 
   From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 
   To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals 
   Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com 
   CSeq: 4 BYE 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Content-Length: 0 
    
    
   F21 REL Switch C -> GW 2 
    
   REL 
   CauseCode=16 Normal 
    
    
   F22 RLC GW 2 -> Switch C 
    
   RLC 
    
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
4.2    Successful FGB PBX to ISDN PBX call with overflow 
    
 PBX A       GW 1        Proxy 1        GW 2         GW 3        PBX C 
   |            |            |            |            |            | 
   |  Seizure   |            |            |            |            | 
   |----------->|            |            |            |            | 
   |    Wink    |            |            |            |            | 
   |<-----------|            |            |            |            | 
   |MF Digits F1|            |            |            |            | 
   |----------->|            |            |            |            | 
   |            | INVITE F2  |            |            |            | 
   |            |----------->| INVITE F3  |            |            | 
   |            |            |----------->|            |            | 
   |            |            |   503 F4   |            |            | 
   |            |            |<-----------|            |            | 
   |            |            |   ACK F5   |            |            | 
   |            |            |----------->|            |            | 
   |            |            |  INVITE F6              |            | 
   |            |            |------------------------>|  SETUP F7  | 
   |            |            |          100  F8        |----------->| 
   |            |            |<------------------------|CALL PROC F9| 
   |            |            |                         |<-----------| 
   |            |            |                         | ALERT F10  | 
   |            |            |          180 F11        |<-----------| 
   |            |  180 F12   |<------------------------|            | 
   |            |<-----------|                         |            | 
   | Ringtone   |            |                         |OneWay Voice| 
   |<===========|            |                         |<===========| 
   |            |            |                         | CONNect F13| 
   |            |            |         200 F14         |<-----------| 
   |            |  200 F15   |<------------------------|            | 
   |  Seizure   |<-----------|                         |            | 
   |<-----------|  ACK F16   |                         |            | 
   |            |----------->|         ACK F17         |            | 
   |            |            |------------------------>|CONN ACK F18| 
   |            |            |                         |----------->| 
   |BothWayVoice|          Both Way RTP Media          |BothWayVoice| 
   |<==========>|<====================================>|<==========>| 
   |            |            |                         |  DISC F19  | 
   |            |            |                         |<-----------| 
   |            |            |         BYE F20         |            | 
   |            |  BYE F21   |<------------------------|  REL F22   | 
   |Seiz Removal|<-----------|                         |----------->| 
   |<-----------|  200 F23   |                         |            | 
   |Seiz Removal|----------->|         200 F24         |            | 
   |----------->|            |------------------------>| REL COM F25| 
   |            |            |                         |<-----------| 
   |            |            |                         |            | 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   PBX Alice calls PBX Carol via Gateway GW 1 and Proxy 1.  During the 
   attempt to reach Carol via GW 2, an error is encountered - Proxy 1 
   receives a 503 Service Unavailable (F4) response to the forwarded 
   INVITE.  This could be due to all circuits being busy, or some other 
   outage at GW 2.  Proxy 1 recognizes the error and uses an alternative 
   route via GW 3 to terminate the call.  From there, the call proceeds 
   normally with Carol answering the call.  The call is terminated when 
   Carol hangs up. 
    
    
   Message Details 
    
   PBX Alice -> GW 1 
    
   Seizure 
    
    
   GW 1 -> PBX A 
    
   Wink 
    
    
   F1 MF Digits PBX Alice -> GW 1 
    
   KP 444 3333 ST 
    
    
   F2 INVITE GW 1 -> Proxy 1 
    
   INVITE sip:4443333@ss1.a.example.com SIP/2.0 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
   Max-Forwards: 70 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com> 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:551313@gw1.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 155 
    
   v=0 
   o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   /* Proxy 1 uses a Location Service function to determine where B is 
   located.  Response is returned listing alternative routes, GW2 and 
   GW3, which are then tried sequentially. */ 
    
   F3 INVITE Proxy 1 -> GW 2 
    
   INVITE sip:4443333@gw2.a.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com> 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:551313@gw1.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 155 
    
   v=0 
   o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F4 503 Service Unavailable GW 2 -> Proxy 1 
    
   SIP/2.0 503 Service Unavailable 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F5 ACK Proxy 1 -> GW 2 
    
   ACK sip:4443333@ss1.a.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Max-Forward: 70 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=314159 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F6 INVITE Proxy 1 -> GW 3 
    
   INVITE sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com> 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:551313@gw1.a.example.com> 
   Content-Type: application/sdp 
   Content-Length: 155 
    
   v=0 
   o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com 
   s=- 
   c=IN IP4 gw1.a.example.com 
   t=0 0 
   m=audio 49172 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F7 SETUP GW 3 -> PBX C 
    
   Protocol discriminator=Q.931 
   Message type=SETUP 
   Bearer capability: Information transfer capability=0 (Speech) or 16 
   (3.1 kHz audio) 
   Channel identification=Preferred or exclusive B-channel 
   Progress indicator=1 (Call is not end-to-end ISDN; further call 
   progress information may be available inband) 
   Called party number: 
   Type of number and numbering plan ID=33 (National number in ISDN 
   numbering plan) 
   Digits=918-555-3333 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
   F8 100 Trying GW 3 -> Proxy 1 
    
   SIP/2.0 100 Trying 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com> 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Content-Length: 0 
    
    
   F9 CALL PROCeeding PBX C -> GW 3 
    
   Protocol discriminator=Q.931 
   Message type=CALL PROC 
    
    
   F10 ALERT PBX C -> GW 3 
    
   Protocol discriminator=Q.931 
   Message type=PROG 
    
    
   /* Based on ALERT message, GW 3 returns a 180 response. */ 
    
   F11 180 Ringing GW 3 -> Proxy 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:+19185553333@gw3.a.example.com;user=phone> 
   Content-Length: 0 
    
    
   F12 180 Ringing Proxy 1 -> GW 1 
    
   SIP/2.0 180 Ringing 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:+19185553333@gw3.a.example.com;user=phone> 
   Content-Length: 0 
 
    
   F13 CONNect PBX C -> GW 3 
    
   Protocol discriminator=Q.931 
   Message type=CONN 
    
    
   F14 200 OK GW 3 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:+19185553333@gw3.a.example.com;user=phone> 
   Content-Type: application/sdp 
   Content-Length: 143 
    
   v=0 
   o=GW 987654321 987654321 IN IP4 gw3.a.example.com 
   s=- 
   c=IN IP4 gw3.a.example.com 
   t=0 0 
   m=audio 14918 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   F15 200 OK Proxy 1 -> GW 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Record-Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 INVITE 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   Contact: <sip:+19185553333@gw3.a.example.com;user=phone> 
   Content-Type: application/sdp 
   Content-Length: 143 
    
   v=0 
   o=GW 987654321 987654321 IN IP4 gw3.a.example.com 
   s=- 
   c=IN IP4 gw3.a.example.com 
   t=0 0 
   m=audio 14918 RTP/AVP 0 
   a=rtpmap:0 PCMU/8000 
    
    
   GW 1 -> PBX A 
    
   Seizure 
    
    
   F16 ACK GW 1 -> Proxy 1 
    
   ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F17 ACK Proxy 1 -> GW 3 
    
   ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
   Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 
    ;received=192.0.2.201 
   Max-Forwards: 69 
   From: <sip:551313@gw1.a.example.com>;tag=63412s 
   To: <sip:4443333@ss1.a.example.com>;tag=123456789 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 ACK 
   Content-Length: 0 
    
    
   F18 CONNect ACK GW 3 -> PBX C 
    
   Protocol discriminator=Q.931 
   Message type=CONN ACK 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   /* RTP streams are established between GW 1 and GW 3. */ 
    
   /* Bob Hangs Up with Alice. */ 
    
   F19 DISConnect PBX C -> GW 3 
    
   Protocol discriminator=Q.931 
   Message type=DISC 
   Cause=16 (Normal clearing) 
    
    
   F20 BYE GW 3 -> Proxy 1 
    
   BYE sip:551313@gw1.a.example.com SIP/2.0 
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq 
   Max-Forwards: 70 
   Route: <sip:ss1.a.example.com;lr> 
   From: <sip:4443333@ss1.a.example.com>;tag=123456789 
   To: <sip:551313@gw1.a.example.com>;tag=63412s 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 BYE 
   Content-Length: 0 
    
    
   F21 BYE Proxy 1 -> GW 1 
    
   BYE sip:551313@gw1.a.example.com SIP/2.0 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq 
    ;received=192.0.2.203 
   Max-Forwards: 69 
   From: <sip:4443333@ss1.a.example.com>;tag=123456789 
   To: <sip:551313@gw1.a.example.com>;tag=63412s 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 BYE 
   Content-Length: 0 
    
    
   GW 1 -> PBX A 
    
   Seizure removal 
    
    
   F22 RELease GW 3 -> PBX C 
    
   Protocol discriminator=Q.931 
   Message type=REL 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
    
   F23 200 OK GW 1 -> Proxy 1 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 
    ;received=192.0.2.111 
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq 
    ;received=192.0.2.203 
   From: <sip:4443333@ss1.a.example.com>;tag=123456789 
   To: <sip:551313@gw1.a.example.com>;tag=63412s 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 BYE 
   Content-Length: 0 
    
    
   F24 200 OK Proxy 1 -> GW 3 
    
   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq 
    ;received=192.0.2.203 
   From: <sip:4443333@ss1.a.example.com>;tag=123456789 
   To: <sip:551313@gw1.a.example.com>;tag=63412s 
   Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com 
   CSeq: 1 BYE 
   Content-Length: 0 
    
    
   F25 RELease COMplete PBX C -> GW 3 
    
   Protocol discriminator=Q.931 
   Message type=REL COM 
    
    
   PBX Alice -> GW 1 
    
   Seizure removal 
    
    
    
    
Security Considerations 
    
   This document provides examples of mapping from SIP to ISUP and ISUP 
   to SIP.  The gateways in these examples are compliant with the 
   Security Considerations Section of RFC zzzz [4] which is summarized 
   here.  
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
   There are few security concerns relating to the mapping of ISUP to 
   SIP besides privacy considerations in the calling party number 
   passing.  Some concerns relating to the mapping from tel URI 
   parameters to ISUP including the user creation of parameters and 
   codes relating to called number and local number portability (LNP).  
   An operator of a gateway should use policies similar to those present 
   in PSTN switches to avoid security problems.   
    
   The mapping from a SIP response code to an ISUP Cause Code presents a 
   theoretical risk, so a gateway operator may implement policies 
   controlling this mapping.  Gateways should also not rely on the 
   contents of the From header field for identity information, as it may 
   be arbitrarily populated by a user.  Instead, some sort of 
   cryptographic authentication and authorization should be used for 
   identity determination.  These flows show both HTTP Digest for 
   authentication of users, although for brevity the challenge is not 
   always shown. 
    
   The early media cut-through shown in some flows is another potential 
   security risk, but it is also required for proper interaction with 
   the PSTN.  Again, a gateway operator should use proper policies 
   relating to early media to prevent fraud and misuse.  Finally, a user 
   agent (even a properly authenticated one) can launch multiple 
   simultaneous requests through a gateway, constituting a denial of 
   service attack.  The adoption of policies to limit the number of 
   simultaneous requests from a single entity may be used to prevent 
   this attack. 
    
   As discussed in the SIP-T framework [8] SIP/ISUP interworking can be 
   employed as an interdomain signaling mechanism that may be subject to 
   pre-existing trust relationships between administrative domains.  Any 
   administrative domain implementing SIP-T or SIP/ISUP interworking 
   should have an adequate security apparatus (including elements that 
   manage any appropriate policies to manage fraud and billing in an 
   interdomain environment) in place to ensure that the translation of 
   ISUP information does not result in any security violations. 
    
   Although no examples of this are shown in this document, transporting 
   ISUP in SIP bodies may provide opportunities for abuse, fraud, and 
   privacy concerns, especially when SIP-T requests can be generated, 
   inspected or modified by arbitrary SIP endpoints. ISUP MIME bodies 
   should be secured (preferably with S/MIME as detailed in RFC 3261 
   [2]) to alleviate this concern. Authentication properties provided by 
   S/MIME would allow the recipient of a SIP-T message to ensure that 
   the ISUP MIME body was generated by an authorized entity. Encryption 
   would ensure that only carriers possessing a particular decryption 
   key are capable of inspecting encapsulated ISUP MIME bodies in a SIP 
   request. 
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
Normative References 
    
                     
   1  Bradner, S., "Key words for use in RFCs to Indicate Requirement 
      Levels", BCP 14, RFC 2119, March 1997 
    
    2 Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., 
      Peterson, J., Sparks, R., Handley, M., and Schooler, E., "SIP: 
      Session Initiation Protocol", RFC 3261, June 2002. 
     
    3 Rosenberg, J. and Schulzrinne, H., "An Offer/Answer Model with 
      SDP", Internet Engineering Task Force, RFC 3264, April 2002. 
     
   4 G. Camarillo, A. Roach, J. Peterson, L. Ong, "ISUP to SIP 
      Mapping", Internet Draft, Internet Engineering Task Force, Work in 
      progress. August 2002. 
    
   5 Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach, 
      P., Luotonen, A. and L. Stewart, "HTTP authentication: Basic and 
      Digest Access Authentication", RFC 2617, June 1999. 
    
   6 J. Rosenberg, H. Schulzrinne, and G. Camarillo, "The Stream 
      Control Transmission Protocol as a Transport for the Session 
      Initiation Protocol," Internet Draft, Internet Engineering Task 
      Force, Work in progress. June 2002. 
    
   7 A. Vaha-Sipila, "URLs for Telephone Calls", Internet Draft, 
      Internet Engineering Task Force, RFC 2806, April 2000. 
    
   8 A. Vemuri and J. Peterson, "Session Initiation Protocol for 
      Telephones (SIP-T): Context and Architectures," RFC 3372, 
      September 2002. 
    
   9  E. Zimmerer, J. Peterson, A. Vemuri, L. Ong, F. Audet, M. Watson, 
      M. Zonoun, "MIME media types for ISUP and QSIG Objects," RFC 3204, 
      December 2001.  
    
   10 P. Faltstrom, "E.164 Numbers and DNS," RFC 2916, September 2000. 
    
Informative References 
     
    
   11 Johnston, A., Donovan, S., Sparks, R., Cunningham, C., Summers, 
      K., "Session Initiation Protocol Basic Call Flow Examples", RFC 
      yyyv, August 2002. 
    
    
    

 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
Acknowledgments 
    
   Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings, 
   and Tom Taylor for their detailed comments during the final review.  
   Thanks to Dean Willis for his early contributions to the development 
   of this document.  Thanks to Jon Peterson for his help on the 
   security section. 
    
   The authors wish to thank Kundan Singh for performing parser 
   validation of messages.  
    
   The authors wish to thank the following individuals for their 
   participation in a detailed review of this call flows document: Aseem 
   Agarwal, Rafi Assadi, Ben Campbell, Sunitha Kumar, Jon Peterson, Marc 
   Petit-Huguenin, Vidhi Rastogi, and Bodgey Yin Shaohua.   
    
   The authors also wish to thank the following individuals for their 
   assistance: Jean-Francois Mule, Hemant Agrawal, Henry Sinnreich, 
   David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole 
   MCI WorldCom IPOP Design team, Scott Orton, Greg Osterhout, Pat 
   Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise 
   Ingram, Denise Caballero, Tom Redman, Ilya Slain, Pat Sollee, John 
   Truetken, and others from MCI WorldCom, 3Com, Cisco, Lucent and 
   Nortel. 
    
Author's Addresses 
    
   All listed authors actively contributed large amounts of text to this 
   document. 
    
      Alan Johnston 
      WorldCom 
      100 South 4th Street 
      St. Louis, MO 63102 
      USA 
    
      EMail:  alan.johnston@wcom.com 
    
    
      Steve Donovan 
      dynamicsoft, Inc. 
      5100 Tennyson Parkway 
      Suite 1200 
      Plano, Texas 75024 
      USA 
    
      EMail:  sdonovan@dynamicsoft.com 
    
    
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
      Robert Sparks 
      dynamicsoft, Inc. 
      5100 Tennyson Parkway 
      Suite 1200 
      Plano, Texas 75024 
      USA 
    
      EMail:  rsparks@dynamicsoft.com 
    
      Chris Cunningham 
      dynamicsoft, Inc. 
      5100 Tennyson Parkway 
      Suite 1200 
      Plano, Texas 75024 
      USA 
    
      EMail: ccunningham@dynamicsoft.com 
    
       
      Kevin Summers 
      Sonus 
      1701 North Collins Blvd, Suite 3000 
      Richardson, TX 75080 
      USA 
    
      Email: kevin.summers@sonusnet.com 
    
    
Intellectual Property Statement 
    
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   pertain to the implementation or use of the technology described in 
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   copyrights, patents or patent applications, or other proprietary 
   rights which may cover technology that may be required to practice 
   this standard. Please address the information to the IETF Executive 
   Director. 
 
 
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                         SIP PSTN Call Flows               April 2003 
 
 
    
    
Full Copyright Statement 
    
   Copyright (C) The Internet Society (2003). All Rights Reserved. 
    
   This document and translations of it may be copied and furnished to 
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Acknowledgement 
 
   Funding for the RFC Editor function is currently provided by the 
   Internet Society. 
    

 
 
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