SIPPING Working Group A. Johnston
Internet Draft WorldCom
Document: draft-ietf-sipping-pstn-call-flows-00.txt S. Donovan
Expires: February 2003 R. Sparks
C. Cunningham
dynamicsoft
K. Summers
Sonus
August 2002
Session Initiation Protocol PSTN Call Flows
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
Abstract
This informational document gives examples of Session Initiation
Protocol (SIP) call flows showing interworking with the Public
Switched Telephone Network (PSTN). Elements in these call flows
include SIP User Agents and Clients, SIP Proxy Servers, and PSTN
Gateways. Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to
PSTN via SIP. PSTN telephony protocols are illustrated using ISDN
(Integrated Services Digital Network), ANSI ISUP (ISDN User Part),
and FGB (Feature Group B) circuit associated signaling. PSTN calls
are illustrated using global telephone numbers from the PSTN and
private extensions served on by a PBX (Private Branch Exchange).
Call flow diagrams and message details are shown.
Johnston et al Expires - February 2003 [Page 1]
SIP PSTN Call Flows August 2002
Conventions used in this document
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC-2119 [1].
Table of Contents
1. Overview.......................................................2
1.1 General Assumptions........................................3
1.2 Legend for Message Flows...................................4
1.3 SIP Protocol Assumptions...................................4
2. SIP to PSTN Dialing............................................6
2.1 Successful SIP to ISUP PSTN call...........................7
2.2 Successful SIP to ISDN PBX call...........................15
2.3 Successful SIP to ISUP PSTN call with overflow............23
2.4 Unsuccessful SIP to PSTN call: Treatment from PSTN........32
2.5 Unsuccessful SIP to PSTN: REL w/Cause from PSTN...........39
2.6 Unsuccessful SIP to PSTN: ANM Timeout.....................44
3. PSTN to SIP Dialing...........................................50
3.1 Successful PSTN to SIP call...............................52
3.2 Successful PSTN to SIP call, Fast Answer..................59
3.3 Successful PBX to SIP call................................65
3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL.....72
3.5 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL......74
3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones...78
3.7 Unsuccessful PSTN->SIP, ACM timeout.......................82
3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy......86
3.9 Unsuccessful PSTN->SIP, Caller Abandonment................90
4. PSTN to PSTN Dialing via SIP Network..........................96
4.1 Successful ISUP PSTN to ISUP PSTN call....................97
4.2 Successful FGB PBX to ISDN PBX call with overflow........105
Security Considerations.........................................113
References......................................................113
Acknowledgments.................................................114
Author's Addresses..............................................115
1. Overview
The call flows shown in this document were developed in the design of
a carrier-class SIP IP Telephony network. They represent an example
minimum set of functionality for SIP to be used in IP Telephony
applications.
It is the hope of the authors that this document will be useful for
SIP implementors, designers, and protocol researchers alike and will
help further the goal of a standard SIP implementation for IP
Telephony. It is envisioned that as changes to the standard and
additional RFCs are added that this document will reflect those
Johnston et al Expires - February 2002 [Page 2]
SIP PSTN Call Flows August 2002
changes and represent the current state of a standard interoperable
SIP IP Telephony implementation.
These call flows are based on the current version 2.0 of SIP in
RFC 3261[2] with SDP usage described in RFC 3264[3].
Note that this document is informational, and is NOT NORMATIVE on any
aspect of SIP or SIP/PSTN interworking.
Various PSTN signaling protocols are illustrated in this document:
ISDN (Integrated Services Digital Network), ANSI ISUP (ISDN User
Part) and FGB (Feature Group B) circuit associated signaling. They
were chosen to illustrate the nature of SIP/PSTN interworking - they
are not a complete or even representative set. Also, some details
and parameters of these PSTN protocols have been omitted. For full
information about SIP to ISUP mapping, refer to [4].
Basic SIP call flow examples contained in a companion document, RFC
yyyy[5].
1.1 General Assumptions
A number of architecture, network, and protocol assumptions underly
the call flows in this document. Note that these assumptions are not
requirements. They are outlined in this section so that they may be
taken into consideration and to aid in the understanding of the call
flow examples.
The authentication of SIP User Agents in these example call flows is
performed using SIP Digest as defined in [3] and [6].
Some Proxy Servers in these call flows insert Record-Route headers
into requests to ensure that they are in the signaling path for
future message exchanges.
These flows show UDP for transport. Other transport schemes could
also be used.
Throughout this document the call flows show a network where the
proxy servers authenticate users on behalf of gateways. Gateways may
also authenticate users directly. Both of these are reasonable usages
of SIP. If gateways do not authenticate directly they would be to
refuse requests from entities other than trusted proxy servers with
which they have effective channel security (for example [7] or [8])."
The SIP Proxy Server has access to a Location Service and other
databases. Information present in the Request-URI and the context
(From header) is sufficient to determine to which proxy or gateway
the message should be routed. In most cases, a primary and secondary
Johnston et al Expires - February 2002 [Page 3]
SIP PSTN Call Flows August 2002
route will be determined in case of Proxy or Gateway failure
downstream.
Gateways provide tones (ringing, busy, etc) and announcements to the
PSTN side based on SIP response messages, or pass along audio in-band
tones (ringing, busy tone, etc.) in an early media stream to the SIP
side.
The interactions between the Proxy and Gateway can be summarized as
follows:
. The SIP Proxy Server performs digit analysis and lookup and
locates the correct gateway.
. The SIP Proxy Server performs gateway location based on primary
and secondary routing.
Telephone numbers are usually represented as SIP URIs. Note that an
alternative is the use of the tel URI [9].
1.2 Legend for Message Flows
Dashed lines (---) represent signaling messages that are mandatory to
the call scenario. These messages can be SIP or PSTN
signaling. The arrow indicates the direction of message flow.
Double dashed lines (===) represent media paths between network
elements.
Messages with parentheses around their name represent optional
messages.
Messages are identified in the Figures as F1, F2, etc. This
references the message details in the list that follows the Figure.
Comments in the message details are shown in the following form:
/* Comments. */
1.3 SIP Protocol Assumptions
This document is informational only and is NOT NORMATIVE in any
sense.
For simplicity in reading and editing the document, there are a
number of differences between some of the examples and actual SIP
messages. For example, the SIP Digest responses are not actual MD5
encodings. Call-IDs are often repeated, and CSeq counts often begin
at 1. Header fields are usually shown in the same order. Usually
Johnston et al Expires - February 2002 [Page 4]
SIP PSTN Call Flows August 2002
only the minimum required header field set is shown, others that
would normally be present such as Accept, Supported, Allow, etc are
not shown.
Actors:
Element Display Name URI IP Address
------- ------------ --- ----------
User Agent BigGuy UserA@atlanta.com 192.168.100.101
User Agent LittleGuy UserB@biloxi.com 192.168.200.201
Proxy Server ss1.atlanta.com 192.168.255.111
User Agent (Gateway) gw1.atlanta.com 192.168.255.201
User Agent (Gateway) gw2.atlanta.com 192.168.255.202
User Agent (Gateway) gw3.atlanta.com 192.168.255.203
User Agent (Gateway) ngw1.atlanta.com 192.168.255.101
User Agent (Gateway) ngw2.atlanta.com 192.168.255.102
Johnston et al Expires - February 2002 [Page 5]
SIP PSTN Call Flows August 2002
2. SIP to PSTN Dialing
In the following scenarios, User A (BigGuy sip:UserA@atlanta.com) is
a SIP phone or other SIP-enabled device. User B is reachable via the
PSTN at global telephone number +19725552222. User A places a call
to User B through a Proxy Server Proxy 1 and a Network Gateway. In
other scenarios, User A places calls to User C, who is served via a
PBX (Private Branch Exchange) and is identified by a private
extension 444-3333, or global number +1-918-555-3333. Note that User
A uses his/her global telephone number +1-314-555-1111 in the From
header in the INVITE messages. This then gives the Gateway the
option of using this header to populate the calling party
identification field in subsequent signaling (CgPN in ISUP). Left
open is the issue of how the Gateway can determine the accuracy of
the telephone number, necessary before passing it as a valid CgPN in
the PSTN.
In these scenarios, User A is a SIP phone or other SIP-enabled
device. User A places a call to User B in the PSTN or User C on a
PBX through a Proxy Server and a Gateway.
In the failure scenarios, the call does not complete. In some
cases, however, a media stream is still setup. This is due to the
fact that some failures in dialing to the PSTN result in in-band
tones (busy, reorder tones or announcements - "The number you have
dialed has changed. The new number is..."). The 183 Session
Progress response containing SDP media information is used to
setup this early media path so that the caller User A knows the final
disposition of the call.
The media stream is either terminated by the caller after the tone or
announcement has been heard and understood, or by the Gateway after a
timer expires.
In other failure scenarios, a SS7 Release with Cause Code is mapped
to a SIP response. In these scenarios, the early media path is not
used, but the actual failure code is conveyed to the caller by the
SIP User Agent Client.
Johnston et al Expires - February 2002 [Page 6]
SIP PSTN Call Flows August 2002
2.1 Successful SIP to ISUP PSTN call
User A Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| Both Way RTP Media | One Way Voice |
|<===============================>|<===============|
| | | ANM F9 |
| | 200 F10 |<---------------|
| 200 F11 |<---------------| |
|<---------------| | |
| ACK F12 | | |
|--------------->| ACK F13 | |
| |--------------->| |
| Both Way RTP Media | Both Way Voice |
|<===============================>|<==============>|
| BYE F14 | | |
|--------------->| BYE F15 | |
| |--------------->| |
| | 200 F16 | |
| 200 F17 |<---------------| REL F18 |
|<---------------| |--------------->|
| | | RLC F19 |
| | |<---------------|
| | | |
User A dials the globalized E.164 number +19725552222 to reach
User B. Note that A might have only dialed the last 7 digits, or
some other dialing plan. It is assumed that the SIP User Agent
Client converts the digits into a global number and puts them into a
SIP URI. Note that tel URIs could be used instead of SIP URIs.
User A could use either their SIP address (sip:UserA@atlanta.com) or
SIP telephone number (sip:+13145551111@ss1.atlanta.com;user=phone) in
the From header. In this example, the telephone number is included,
and it is shown as being passed as calling party identification
Johnston et al Expires - February 2002 [Page 7]
SIP PSTN Call Flows August 2002
through the Network Gateway (NGW 1) to User B (F5). Note that for
this number to be passed into the SS7 network, it would have to be
somehow verified for accuracy.
In this scenario, User B answers the call then User A disconnects the
call. Signaling between NGW 1 and User B's telephone switch is ANSI
ISUP. For the details of SIP to ISUP mapping, refer to [4].
Message Details
F1 INVITE A -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Proxy-Authorization: Digest username="UserA", realm="atlanta.com",
nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
uri="sip:+19725552222@ss1.atlanta.com;user=phone",
response="ccdca50cb091d587421457305d097458c"
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> User A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Johnston et al Expires - February 2002 [Page 8]
SIP PSTN Call Flows August 2002
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine the gateway
for terminating this call. The call is forwarded to NGW 1. Client
for A prepares to receive data on port 49172 from the
network.*/
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> User B
Johnston et al Expires - February 2002 [Page 9]
SIP PSTN Call Flows August 2002
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM User B -> NGW 1
ACM
F7 183 Session Progress NGW 1 -> Proxy 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* NGW 1 sends PSTN audio (ringing) in the RTP path to A */
F8 183 Session Progress Proxy 1 -> User A
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Johnston et al Expires - February 2002 [Page 10]
SIP PSTN Call Flows August 2002
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F9 ANM User B -> NGW 1
ANM
F10 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F11 200 OK Proxy 1 -> User A
Johnston et al Expires - February 2002 [Page 11]
SIP PSTN Call Flows August 2002
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F12 ACK A -> Proxy 1
ACK sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
F13 ACK Proxy 1 -> NGW 1
ACK sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Johnston et al Expires - February 2002 [Page 12]
SIP PSTN Call Flows August 2002
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
/* User A Hangs Up with User B. */
F14 BYE A -> Proxy 1
BYE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F15 BYE Proxy 1 -> NGW 1
BYE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F16 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Johnston et al Expires - February 2002 [Page 13]
SIP PSTN Call Flows August 2002
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F17 200 OK Proxy 1 -> A
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F18 REL NGW 1 -> B
REL
CauseCode=16 Normal
F19 RLC B -> NGW 1
RLC
Johnston et al Expires - February 2002 [Page 14]
SIP PSTN Call Flows August 2002
2.2 Successful SIP to ISDN PBX call
User A Proxy 1 GW 1 PBX C
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| SETUP F5 |
| | |--------------->|
| | | CALL PROC F6 |
| | |<---------------|
| | | PROGress F7 |
| | 180 F8 |<---------------|
| 180 F9 |<---------------| |
|<---------------| | |
| | | One Way Voice |
| | |<===============|
| | | CONNect F10 |
| | |<---------------|
| | | CONNect ACK F11|
| | 200 F12 |--------------->|
| 200 F13 |<---------------| |
|<---------------| | |
| ACK F14 | | |
|--------------->| ACK F15 | |
| |--------------->| |
| Both Way RTP Media | Both Way Voice |
|<===============================>|<==============>|
| BYE F16 | | |
|--------------->| BYE F17 | |
| |--------------->| |
| | 200 F18 | |
| 200 F19 |<---------------| DISConnect F20 |
|<---------------| |--------------->|
| | | RELease F21 |
| | |<---------------|
| | | RELease COM F22|
| | |--------------->|
| | | |
User A is a SIP device while User C is connected via a
Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk
group. User A dials User C's telephone number (918-555-3333) which
is globalized and put into a SIP URI.
The host portion of the Request-URI in the INVITE F3 is used to
Johnston et al Expires - February 2002 [Page 15]
SIP PSTN Call Flows August 2002
identify the context (customer, trunk group, or line) in which the
private number 444-3333 is valid. Otherwise, this INVITE message
could get forwarded by GW 1 and the context of the digits could
become lost and the call unroutable.
Proxy 1 looks up the telephone number and locates the gateway that
serves User C. User C is identified by its extension
(444-3333) in the Request-URI sent to GW 1.
Message Details
F1 INVITE A -> Proxy 1
INVITE sip:+19185553333@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 INVITE
Contact: <sip:UserA@192.168.100.101>
Proxy-Authorization: Digest username="UserA",
realm="atlanta.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",
opaque="", uri="sip:+19185553333@ss1.atlanta.com;user=phone",
response="6c792f5c9fa360358b93c7fb826bf550"
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> User A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
Johnston et al Expires - February 2002 [Page 16]
SIP PSTN Call Flows August 2002
CSeq: 2 INVITE
Content-Length: 0
F3 INVITE Proxy 1 -> GW 1
INVITE sip:4443333@gw1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 INVITE
Contact: <sip:UserA@192.168.100.101>
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying GW -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 INVITE
Content-Length: 0
F5 SETUP GW 1 -> User C
Protocol discriminator=Q.931
Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio)
Johnston et al Expires - February 2002 [Page 17]
SIP PSTN Call Flows August 2002
Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN;further call
progress information may be available inband)
Called party number:
Type of number unknown
Digits=444-3333
F6 CALL PROCeeding User C -> GW 1
Protocol discriminator=Q.931
Message type=CALL PROC
Channel identification=Exclusive B-channel
F7 PROGress User C -> GW 1
Protocol discriminator=Q.931
Message type=PROG
Progress indicator=1 (Call is not end-to-end ISDN;further call
progress information may be available inband)
F8 180 Ringing GW 1 -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 INVITE
Contact: <sip:4443333@gw1.atlanta.com>
Content-Length: 0
F9 180 Ringing Proxy 1 -> User A
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
Johnston et al Expires - February 2002 [Page 18]
SIP PSTN Call Flows August 2002
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 INVITE
Contact: <sip:4443333@gw1.atlanta.com>
Content-Length: 0
F10 CONNect User C -> GW 1
Protocol discriminator=Q.931
Message type=CONN
F11 CONNect ACK GW 1 -> User C
Protocol discriminator=Q.931
Message type=CONN ACK
F12 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 INVITE
Contact: <sip:4443333@gw1.atlanta.com>
Content-Type: application/sdp
Content-Length: 140
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F13 200 OK Proxy 1 -> User A
Johnston et al Expires - February 2002 [Page 19]
SIP PSTN Call Flows August 2002
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 INVITE
Contact: <sip:4443333@gw1.atlanta.com>
Content-Type: application/sdp
Content-Length: 140
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F14 ACK A -> Proxy 1
ACK sip:4443333@gw1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 ACK
Content-Length: 0
F15 ACK Proxy 1 -> GW 1
ACK sip:4443333@gw1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
Johnston et al Expires - February 2002 [Page 20]
SIP PSTN Call Flows August 2002
CSeq: 2 ACK
Content-Length: 0
/* User A Hangs Up with User B. */
F16 BYE A -> Proxy 1
BYE sip:4443333@gw1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 3 BYE
Content-Length: 0
F17 BYE Proxy 1 -> GW 1
BYE sip:4443333@gw1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 3 BYE
Content-Length: 0
F18 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
Johnston et al Expires - February 2002 [Page 21]
SIP PSTN Call Flows August 2002
CSeq: 3 BYE
Content-Length: 0
F19 200 OK Proxy 1 -> A
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: OtherGuy <sip:+19185553333@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 3 BYE
Content-Length: 0
F20 DISConnect GW 1 -> User C
Protocol discriminator=Q.931
Message type=DISC
Cause=16 (Normal clearing)
F21 RELease User C -> GW 1
Protocol discriminator=Q.931
Message type=REL
F22 RELease COMplete GW 1 -> User C
Protocol discriminator=Q.931
Message type=REL COM
Johnston et al Expires - February 2002 [Page 22]
SIP PSTN Call Flows August 2002
2.3 Successful SIP to ISUP PSTN call with overflow
User A Proxy 1 NGW 1 NGW 2 Switch B
| | | | |
| INVITE F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| 100 F3 |------------->| | |
|<-------------| 503 F4 | | |
| |<-------------| | |
| | ACK F5 | | |
| |------------->| | |
| | INVITE F6 | |
| |---------------------------->| IAM F7 |
| | |------------->|
| | | ACM F8 |
| | 183 F9 |<-------------|
| 183 F10 |<----------------------------| |
|<-------------| | |
| Two Way RTP Media | One Way Voice|
|<==========================================>|<=============|
| | | ANM F11 |
| | 200 F12 |<-------------|
| 200 F13 |<----------------------------| |
|<-------------| | |
| ACK F14 | | |
|------------->| ACK F15 | |
| |---------------------------->| |
| Both Way RTP Media |Both Way Voice|
|<==========================================>|<============>|
| BYE F16 | | |
|------------->| BYE F17 | |
| |---------------------------->| |
| | 200 F18 | |
| 200 F19 |<----------------------------| REL F20 |
|<-------------| |------------->|
| | | RLC F21 |
| | |<-------------|
| | | |
User A calls User B through Proxy 1. Proxy 1 tries to route to a
Network Gateway NGW 1. NGW 1 is not available and responds with a 503
Service Unavailable (F4). The call is then routed to Network Gateway
NGW 2. User B answers the call. The call is terminated when User A
disconnects the call. NGW 2 and User B's telephone switch use ANSI
ISUP signaling.
Message Details
Johnston et al Expires - February 2002 [Page 23]
SIP PSTN Call Flows August 2002
F1 INVITE A -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Proxy-Authorization: Digest username="UserA",
realm="atlanta.com", nonce="b59311c3ba05b401cf80b2a2c5ac51b0",
opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
response="ba6ab44923fa2614b28e3e3957789ab0"
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Proxy 1 receives a primary route NGW 1 and a secondary
route NGW 2. NGW 1 is tried first */
F2 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Content-Type: application/sdp
Content-Length: 147
Johnston et al Expires - February 2002 [Page 24]
SIP PSTN Call Flows August 2002
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F3 100 Trying Proxy 1 -> User A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F4 503 Service Unavailable NGW 1 -> Proxy 1
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 ACK Proxy 1 -> NGW 1
ACK sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com>;user=phone>
Johnston et al Expires - February 2002 [Page 25]
SIP PSTN Call Flows August 2002
;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
/* Proxy 1 now tries secondary route to NGW 2 */
F6 INVITE Proxy 1 -> NGW 2
INVITE sip:+19725552222@ngw2.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F7 IAM NGW 2 -> User B
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F8 ACM User B -> NGW 2
ACM
F9 183 Session Progress NGW 2 -> Proxy 1
SIP/2.0 183 Session Progress
Johnston et al Expires - February 2002 [Page 26]
SIP PSTN Call Flows August 2002
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
s=-
c=IN IP4 192.168.255.102
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* RTP packets are sent by GW to A for audio (e.g. ring tone) */
F10 183 Session Progress Proxy 1 -> User A
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
s=-
c=IN IP4 192.168.255.102
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Johnston et al Expires - February 2002 [Page 27]
SIP PSTN Call Flows August 2002
F11 ANM User B -> NGW 2
ANM
F12 200 OK NGW 2 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
s=-
c=IN IP4 192.168.255.102
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F13 200 OK Proxy 1 -> User A
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw2.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
Johnston et al Expires - February 2002 [Page 28]
SIP PSTN Call Flows August 2002
v=0
o=GW 2890844527 2890844527 IN IP4 ngw2.atlanta.com
s=-
c=IN IP4 192.168.255.102
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F14 ACK A -> Proxy 1
ACK sip:+19725552222@ngw2.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
F15 ACK Proxy 1 -> NGW 2
ACK sip:+19725552222@ngw2.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between A and B(via the GW) */
/* User A Hangs Up with User B. */
F16 BYE A -> Proxy 1
BYE sip:+19725552222@ngw2.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Johnston et al Expires - February 2002 [Page 29]
SIP PSTN Call Flows August 2002
Max-Forwards: 70
Route: <ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F17 BYE Proxy 1 -> NGW 2
BYE sip:+19725552222@ngw2.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F18 200 OK NGW 2 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F19 200 OK Proxy 1 -> User A
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
Johnston et al Expires - February 2002 [Page 30]
SIP PSTN Call Flows August 2002
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 2 BYE
Content-Length: 0
F20 REL NGW 2 -> B
REL
CauseCode=16 Normal
F21 RLC B -> NGW 2
RLC
Johnston et al Expires - February 2002 [Page 31]
SIP PSTN Call Flows August 2002
2.4 Unsuccessful SIP to PSTN call: Treatment from PSTN
User A Proxy 1 NGW 1 User B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| Two Way RTP Media | One Way Voice |
|<===============================>|<===============|
| Treatment Applied |
|<=================================================|
| CANCEL F9 | | |
|--------------->| | |
| 200 F10 | | |
|<---------------| CANCEL F11 | |
| |--------------->| |
| | 200 F12 | |
| |<---------------| REL F13 |
| | |--------------->|
| | | RLC F14 |
| | 487 F15 |<---------------|
| |<---------------| |
| | ACK F16 | |
| 487 F17 |--------------->| |
|<---------------| | |
| ACK F18 | | |
|--------------->| | |
| | | |
User A calls User B in the PSTN through a proxy server Proxy 1 and a
Network Gateway NGW 1. The call is rejected by the PSTN with an in-
band treatment (tone or recording) played. User A hears the
treatment and then hangs up, which results in a CANCEL (F9) being
sent to terminate the call. (A BYE is not sent since no final
response was ever received by User A.)
Message Details
Johnston et al Expires - February 2002 [Page 32]
SIP PSTN Call Flows August 2002
F1 INVITE A -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Proxy-Authorization: Digest username="UserA",
realm="atlanta.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40",
opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
response="e178fbe430e6680a1690261af8831f40"
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW
1. Client for A prepares to receive data on port 49172 from the
network. */
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Johnston et al Expires - February 2002 [Page 33]
SIP PSTN Call Flows August 2002
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> User B
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM User B -> NGW 1
ACM
Johnston et al Expires - February 2002 [Page 34]
SIP PSTN Call Flows August 2002
F7 183 Session Progress NGW 1 -> Proxy 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F8 183 Session Progress Proxy 1 -> User A
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
Johnston et al Expires - February 2002 [Page 35]
SIP PSTN Call Flows August 2002
a=rtpmap:0 PCMU/8000
/* Caller hears the recorded announcement, then hangs up */
F9 CANCEL A -> Proxy 1
CANCEL sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F10 200 OK Proxy 1 -> A
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F11 CANCEL Proxy 1 -> NGW 1
CANCEL sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F12 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Johnston et al Expires - February 2002 [Page 36]
SIP PSTN Call Flows August 2002
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F13 REL NGW 1 -> B
REL
CauseCode=18 No user responding
F14 RLC B -> NGW 1
RLC
F15 487 Request Terminated NGW 1 -> Proxy 1
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F16 ACK Proxy 1 -> NGW 1
ACK sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
Johnston et al Expires - February 2002 [Page 37]
SIP PSTN Call Flows August 2002
F17 487 Request Terminated Proxy 1 -> A
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F18 ACK A -> Proxy 1
ACK sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
Johnston et al Expires - February 2002 [Page 38]
SIP PSTN Call Flows August 2002
2.5 Unsuccessful SIP to PSTN: REL w/Cause from PSTN
User A Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | REL(1) F6 |
| | |<---------------|
| | | RLC F7 |
| | 404 F8 |--------------->|
| |<---------------| |
| | ACK F9 | |
| |--------------->| |
| 404 F10 | | |
|<---------------| | |
| ACK F11 | | |
|--------------->| | |
| | | |
User A calls PSTN User B through a Proxy Server Proxy 1 and a Network
Gateway NGW 1. The call is rejected by the PSTN with a
ANSI ISUP Release message REL containing a specific Cause code.
This cause value (1) is mapped by the Gateway to a SIP 404 Address
Incomplete response which is proxied back to User A. For more
details of ISUP cause value to SIP responses refer to [4].
Message Details
F1 INVITE A -> Proxy 1
INVITE sip:+44-1234@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Proxy-Authorization: Digest username="UserA",
realm="atlanta.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",
Johnston et al Expires - February 2002 [Page 39]
SIP PSTN Call Flows August 2002
opaque="", uri="sip:+44-1234@ss1.atlanta.com;user=phone",
response="a451358d46b55512863efe1dccaa2f42"
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW1.
Client for A prepares to receive data on port 49172 from the network.
*/
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+44-1234@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Content-Type: application/sdp
Content-Length: 147
Johnston et al Expires - February 2002 [Page 40]
SIP PSTN Call Flows August 2002
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> User B
IAM
CdPN=44-1234,NPI=E.164,NOA=International
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 REL User B -> NGW 1
REL
CauseValue=1 Unallocated number
F7 RLC NGW 1 -> User B
RLC
/* Network Gateway maps CauseValue=1 to the SIP message 404 Not
Found */
F8 404 Not Found NGW 1 -> Proxy 1
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Johnston et al Expires - February 2002 [Page 41]
SIP PSTN Call Flows August 2002
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Error-Info: <sip:not-found-ann@ann.atlanta.com>
Content-Length: 0
F9 ACK Proxy 1 -> NGW 1
ACK sip:+44-1234@ngw1.atlanta.com;user=phone SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
F10 404 Not Found Proxy 1 -> User A
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Error-Info: <sip:not-found-ann@ann.atlanta.com>
Content-Length: 0
F11 ACK User A -> Proxy 1
ACK sip:+44-1234@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+44-1234@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Johnston et al Expires - February 2002 [Page 42]
SIP PSTN Call Flows August 2002
Content-Length: 0
Johnston et al Expires - February 2002 [Page 43]
SIP PSTN Call Flows August 2002
2.6 Unsuccessful SIP to PSTN: ANM Timeout
User A Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| | Timer on NGW 1 Expires |
| | | |
| | | REL F9 |
| | |--------------->|
| | | RLC F10 |
| | 480 F11 |<---------------|
| |<---------------| |
| | ACK F12 | |
| |--------------->| |
| 480 F13 | | |
|<---------------| | |
| ACK F14 | | |
|--------------->| | |
User A calls User B in the PSTN through a proxy server Proxy 1 and
Network Gateway NGW 1. The call is released by the Gateway after a
timer expires due to no ANswer Message (ANM) being received. The
Gateway sends an ISUP Release REL message to the PSTN and a 480
Temporarily Unavailable response to User A in the SIP network.
Message Details
F1 INVITE A -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
Johnston et al Expires - February 2002 [Page 44]
SIP PSTN Call Flows August 2002
CSeq: 1 INVITE
Contact: <sip:UserA@192.168.100.101>
Proxy-Authorization: Digest username="UserA",
realm="atlanta.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40",
opaque="", uri="sip:+19725552222@ss1.atlanta.com;user=phone",
response="579cb9db184cdc25bf816f37cbc03c7d"
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW
1. Client for A prepares to receive data on port 49172 from the
network.*/
F2 100 Trying Proxy 1 -> A
SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Johnston et al Expires - February 2002 [Page 45]
SIP PSTN Call Flows August 2002
Contact: <sip:UserA@192.168.100.101>
Content-Type: application/sdp
Content-Length: 147
v=0
o=UserA 2890844526 2890844526 IN IP4 client.atlanta.com
s=-
c=IN IP4 192.168.100.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> User B
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM User B -> NGW 1
ACM
F7 183 Session Progress NGW 1 -> Proxy 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
Johnston et al Expires - February 2002 [Page 46]
SIP PSTN Call Flows August 2002
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F8 183 Session Progress Proxy 1 -> User A
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19725552222@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* After NGW 1's timer expires, Network Gateway sends REL to ISUP
network and 480 to SIP network */
F9 REL NGW 1 -> User B
REL
Johnston et al Expires - February 2002 [Page 47]
SIP PSTN Call Flows August 2002
CauseCode=18 No user responding
F10 RLC User B -> NGW 1
RLC
F11 480 Temporarily Unavailable NGW 1 -> Proxy 1
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 INVITE
Error-Info: <sip:temp-unavail-ann@ann.atlanta.com>
Content-Length: 0
F12 ACK Proxy 1 -> NGW 1
ACK sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
F13 480 Temporarily Unavailable F13 Proxy 1 -> User A
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
;received=192.168.100.101
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
Johnston et al Expires - February 2002 [Page 48]
SIP PSTN Call Flows August 2002
CSeq: 1 INVITE
Error-Info: <sip:temp-unavail-ann@ann.atlanta.com>
Content-Length: 0
F14 ACK User A -> Proxy 1
ACK sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP client.atlanta.com:5060;branch=z9hG4bK74bf9
From: BigGuy <sip:+13145551111@ss1.atlanta.com;user=phone>
;tag=9fxced76sl
To: LittleGuy <sip:+19725552222@ss1.atlanta.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@atlanta.com
CSeq: 1 ACK
Content-Length: 0
Johnston et al Expires - February 2002 [Page 49]
SIP PSTN Call Flows August 2002
3. PSTN to SIP Dialing
In these scenarios, User A is placing calls from the PSTN to User B
in a SIP network. User A's telephone switch signals to a Network
Gateway (NGW 1) using ANSI ISUP.
Since the called SIP User Agent does not send in-band signaling
information, no early media path needs to be established on the IP
side. As a result, the 183 Session Progress response is not used.
However, NGW 1 will establish a one way speech path prior to call
completion, and generate ringing for the PSTN caller. Any tones or
recordings are generated by NGW 1 and played in this speech path.
When the call completes successfully, NGW 1 bridges the PSTN speech
path with the IP media path.
To reduce the number of messages, only a single proxy server is shown
in these flows, which means that the atlanta.com proxy server has
access to the biloxi.com location service.
Johnston et al Expires - February 2002 [Page 50]
SIP PSTN Call Flows August 2002
Johnston et al Expires - February 2002 [Page 51]
SIP PSTN Call Flows August 2002
3.1 Successful PSTN to SIP call
Switch A NGW 1 Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| ACM F7 |<---------------| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | 200 F8 |
|<===============| 200 F9 |<---------------|
| |<---------------| |
| | ACK F10 | |
| ANM F12 |--------------->| ACK F11 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| REL F13 | | |
|--------------->| | |
| RLC F14 | | |
|<---------------| BYE F15 | |
| |--------------->| BYE F16 |
| | |--------------->|
| | | 200 F17 |
| | 200 F18 |<---------------|
| |<---------------| |
| | | |
In this scenario, User A from the PSTN calls User B through a Network
Gateway NGW1 and Proxy Server Proxy 1. When User B answers the call
the media path is setup end-to-end. The call terminates when User A
hangs up the call, with User A's telephone switch sending an ISUP
RELease message which is mapped to a BYE by NGW 1.
Message Details
F1 IAM User A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
Johnston et al Expires - February 2002 [Page 52]
SIP PSTN Call Flows August 2002
F2 INVITE A -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW
1. NGW 1 prepares to receive data on port 3456 from User A.*/
F3 INVITE Proxy 1 -> User B
INVITE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
Johnston et al Expires - February 2002 [Page 53]
SIP PSTN Call Flows August 2002
a=rtpmap:0 PCMU/8000
F4 100 Trying User B -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 180 Ringing User B -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Length: 0
F6 180 Ringing Proxy 1 -> NGW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Length: 0
F7 ACM NGW 1 -> User A
Johnston et al Expires - February 2002 [Page 54]
SIP PSTN Call Flows August 2002
ACM
F8 200 OK User B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
Contact: <sip:UserB@192.168.200.201>
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 145
v=0
o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
s=-
c=IN IP4 192.168.200.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F9 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Type: application/sdp
Content-Length: 145
v=0
o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
s=-
c=IN IP4 192.168.200.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Johnston et al Expires - February 2002 [Page 55]
SIP PSTN Call Flows August 2002
F10 ACK NGW 1 -> Proxy 1
ACK sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F11 ACK Proxy 1 -> User B
ACK sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F12 ANM User B -> NGW 1
ANM
/* RTP streams are established between A and B (via the GW) */
/* User A Hangs Up with User B. */
F13 REL User A -> NGW 1
REL
CauseCode=16 Normal
F14 RLC NGW 1 -> User A
RLC
Johnston et al Expires - February 2002 [Page 56]
SIP PSTN Call Flows August 2002
F15 BYE NGW 1-> Proxy 1
BYE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F16 BYE Proxy 1 -> User B
BYE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F17 200 OK User B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F18 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
Johnston et al Expires - February 2002 [Page 57]
SIP PSTN Call Flows August 2002
CSeq: 2 BYE
Content-Length: 0
Johnston et al Expires - February 2002 [Page 58]
SIP PSTN Call Flows August 2002
3.2 Successful PSTN to SIP call, Fast Answer
Switch A NGW 1 Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 200 F5 |
| | 200 F6 |<---------------|
| |<---------------| |
| | ACK F7 | |
| ANM F9 |--------------->| ACK F8 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| REL F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| BYE F12 | |
| |--------------->| BYE F13 |
| | |--------------->|
| | | 200 F14 |
| | 200 F15 |<---------------|
| |<---------------| |
| | | |
This "fast answer" scenario is similar to 5.1.1 except that User B
immediately accepts the call, sending a 200 OK (F5) without sending a
180 Ringing response. The Gateway then sends an Answer Message (ANM)
without sending an Address Complete Message (ACM). Note that for
ETSI and some other ISUP variants, a CONnect message (CON) would be
sent instead of the ANM.
Message Details
F1 IAM User A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1
INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Johnston et al Expires - February 2002 [Page 59]
SIP PSTN Call Flows August 2002
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to User
B. User B prepares to receive data on port 3456 from User A.*/
F3 INVITE Proxy 1 -> User B
INVITE UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1
Johnston et al Expires - February 2002 [Page 60]
SIP PSTN Call Flows August 2002
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 200 OK User B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Type: application/sdp
Content-Length: 145
v=0
o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
s=-
c=IN IP4 192.168.200.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F6 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Type: application/sdp
Content-Length: 145
Johnston et al Expires - February 2002 [Page 61]
SIP PSTN Call Flows August 2002
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.200.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F7 ACK NGW 1 -> Proxy 1
ACK UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F8 ACK Proxy 1 -> User B
ACK UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=130.131.132.14
Max-Forwards: 69
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F9 ANM User B -> NGW 1
ANM
/* RTP streams are established between A and B (via the GW) */
/* User A Hangs Up with User B. */
F10 REL ser A -> NGW 1
REL
CauseCode=16 Normal
Johnston et al Expires - February 2002 [Page 62]
SIP PSTN Call Flows August 2002
F11 RLC NGW 1 -> User A
RLC
F12 BYE NGW 1 -> Proxy 1
BYE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F13 BYE Proxy 1 -> User B
BYE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F14 200 OK User B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F15 200 OK Proxy 1 -> NGW 1
Johnston et al Expires - February 2002 [Page 63]
SIP PSTN Call Flows August 2002
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
Johnston et al Expires - February 2002 [Page 64]
SIP PSTN Call Flows August 2002
3.3 Successful PBX to SIP call
PBX A GW 1 Proxy 1 User B
| | | |
| Seizure | | |
|--------------->| | |
| Wink | | |
|<---------------| | |
| MF Digits F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| |<---------------| |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | 200 F7 |
|<===============| 200 F8 |<---------------|
| |<---------------| |
| | ACK F9 | |
| Seizure |--------------->| ACK F10 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| Seizure Removal| | |
|--------------->| | |
| Seizure Removal| | |
|<---------------| BYE F11 | |
| |--------------->| BYE F12 |
| | |--------------->|
| | | 200 F13 |
| | 200 F14 |<---------------|
| |<---------------| |
| | | |
In this scenario, User A dials from PBX A to User B through GW 1 and
Proxy 1. This is an example of a call that appears destined for the
PSTN but instead is routed to a SIP Client.
Signaling between PBX A and GW 1 is Feature Group B (FGB) circuit
associated signaling, in-band Mult-Frequency (MF) outpulsing. After
the receipt of the 180 Ringing from User B, GW 1 generates ringing
tone for User A.
User B answers the call by sending a 200 OK. The call terminates
when User A hangs up, causing GW1 to send a BYE.
Johnston et al Expires - February 2002 [Page 65]
SIP PSTN Call Flows August 2002
The Gateway can only identify the trunk group that the
call came in on, it cannot identify the individual line on PBX A that
is placing the call. The SIP URI used to identify the caller is
shown in these flows as sip:551313@gw1.atlanta.com.
Message Details
PBX A -> GW 1
Seizure
GW 1 -> PBX A
Wink
F1 MF Digits PBX A -> GW 1
KP 1 972 555 2222 ST
F2 INVITE GW 1 -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine where the
phone number +19725552222 is located. Based upon location
analysis the call is forwarded to SIP User B. */
Johnston et al Expires - February 2002 [Page 66]
SIP PSTN Call Flows August 2002
F3 INVITE Proxy 1 -> User B
INVITE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> GW 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 180 Ringing User B -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Johnston et al Expires - February 2002 [Page 67]
SIP PSTN Call Flows August 2002
Contact: <sip:UserB@192.168.200.201>
Content-Length: 0
F6 180 Ringing Proxy 1 -> GW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Length: 0
/* One way Voice path is established between GW and the PBX for
ringing. */
F7 200 OK User B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
Contact: <sip:UserB@192.168.200.201>
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 145
v=0
o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
s=-
c=IN IP4 192.168.200.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F8 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK
Johnston et al Expires - February 2002 [Page 68]
SIP PSTN Call Flows August 2002
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Type: application/sdp
Content-Length: 145
v=0
o=UserB 2890844527 2890844527 IN IP4 client.biloxi.com
s=-
c=IN IP4 192.168.200.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F9 ACK GW 1 -> Proxy 1
ACK sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F10 ACK Proxy 1 -> User B
ACK sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Max-Forwards: 69
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between A and B (via the GW) */
Johnston et al Expires - February 2002 [Page 69]
SIP PSTN Call Flows August 2002
/* User A Hangs Up with User B. */
F11 BYE GW 1 -> Proxy 1
BYE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F12 BYE Proxy 1 -> User B
BYE sip:UserB@192.168.200.201 SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Max-Forwards: 69
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F13 200 OK User B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
F14 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
From: <sip:551313@gw1.atlanta.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Johnston et al Expires - February 2002 [Page 70]
SIP PSTN Call Flows August 2002
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 2 BYE
Content-Length: 0
Johnston et al Expires - February 2002 [Page 71]
SIP PSTN Call Flows August 2002
3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL
Switch A GW 1 Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| |
| | 604 F3 | |
| |<---------------| |
| | ACK F4 | |
| |--------------->| |
| REL F5 | | |
|<---------------| | |
| RLC F6 | | |
|--------------->| | |
| | | |
User A attempts to place a call through Gateway GW 1 and Proxy 1,
which is unable to find any routing for the number. The call is
rejected by Proxy 1 with a REL message containing a specific Cause
value mapped by the gateway based on the SIP error.
Message Details
F1 IAM User A -> GW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-9999,NPI=E.164,NOA=National
F2 INVITE A -> Proxy 1
INVITE sip:+1972559999@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@gw1.atlanta.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@gw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 140
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
Johnston et al Expires - February 2002 [Page 72]
SIP PSTN Call Flows August 2002
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service to find a route to +1-972-555-
9999. A route is not found, so Proxy 1 rejects the call. */
F3 604 Does Not Exist Anywhere Proxy 1 -> GW 1
SIP/2.0 604 Does Not Exist Anywhere
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
From: <sip:+13145551111@gw1.atlanta.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.atlanta.com;user=phone>;tag=6a34d410
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 INVITE
Error-Info: <sip:does-not-exist@ann.atlanta.com>
Content-Length: 0
F4 ACK GW 1 -> Proxy 1
ACK sip:+1972559999@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@gw1.atlanta.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.atlanta.com;user=phone>;tag=6a34d410
Call-ID: 4Fde34wkd11wsGFDs3@gw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F5 REL GW 1 -> User A
REL
CauseCode=1
F6 RLC User A -> GW 1
RLC
Johnston et al Expires - February 2002 [Page 73]
SIP PSTN Call Flows August 2002
3.5 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL
Switch A NGW 1 Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 600 F5 |
| | |<---------------|
| | | ACK F6 |
| | 600 F7 |--------------->|
| |<---------------| |
| | ACK F8 | |
| |--------------->| |
| REL(17) F9 | | |
|<---------------| | |
| RLC F10 | | |
|<-------------->| | |
| | | |
In this scenario, User A calls User B through Network Gateway NGW 1
and Proxy 1. The call is routed to User B by Proxy 1. The call is
rejected by User B who sends a 600 Busy Everywhere response. The
Gateway sends a REL message containing a specific Cause value mapped
by the gateway based on the SIP error.
Since no interworking is indicated in the IAM (F1), the busy tone is
generated locally by User A's telephone switch. In scenario 5.2.3,
the busy signal is generated by the Gateway since interworking is
indicated. For more discussion on interworking, refer to [4].
Message Details
F1 IAM User A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE A -> Proxy 1
INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Johnston et al Expires - February 2002 [Page 74]
SIP PSTN Call Flows August 2002
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 140
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to User B. */
F3 INVITE F3 Proxy 1 -> User B
INVITE UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 140
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Johnston et al Expires - February 2002 [Page 75]
SIP PSTN Call Flows August 2002
;received=192.168.255.201
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 600 Busy Everywhere User B -> Proxy 1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F6 ACK Proxy 1 -> User B
ACK UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F7 600 Busy Everywhere Proxy 1 -> NGW 1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F8 ACK NGW 1 -> Proxy 1
ACK UserB@biloxi.com SIP/2.0
Johnston et al Expires - February 2002 [Page 76]
SIP PSTN Call Flows August 2002
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F9 REL NGW 1 -> User A
REL
CauseCode=17 Busy
F10 RLC User A -> NGW 1
RLC
Johnston et al Expires - February 2002 [Page 77]
SIP PSTN Call Flows August 2002
3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones
Switch A NGW 1 Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 600 F5 |
| | |<---------------|
| | | ACK F6 |
| | 600 F7 |--------------->|
| |<---------------| |
| | ACK F8 | |
| ACM F9 |--------------->| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Busy Tone | | |
|<===============| | |
| REL(16) F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| | |
| | | |
In this scenario, User A calls User B through Network Gateway NGW1
and Proxy 1. The call is routed to User B by Proxy 1. The call is
rejected by the User B client. NGW 1 sets up a two way voice path to
User A and plays busy tone. The caller then disconnects
NGW 1 plays the busy tone since the IAM (F1) indicates the
interworking is present. In scenario 5.2.2, with no interworking,
the busy indication is carried in the REL Cause value and is
generated locally instead.
Again, note that for ETSI or ITU ISUP, a CONnect message would be
sent instead of the Answer Message.
Message Details
F1 IAM User A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
Johnston et al Expires - February 2002 [Page 78]
SIP PSTN Call Flows August 2002
CdPN=972-555-2222,NPI=E.164,NOA=National
Interworking=encountered
F2 INVITE NGW1 -> Proxy 1
INVITE sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to User B. */
F3 INVITE Proxy 1 -> User B
INVITE UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
Johnston et al Expires - February 2002 [Page 79]
SIP PSTN Call Flows August 2002
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying User B -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 600 Busy Everywhere User B -> Proxy 1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F6 ACK Proxy 1 -> User B
ACK UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F7 600 Busy Everywhere Proxy 1 -> NGW 1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Johnston et al Expires - February 2002 [Page 80]
SIP PSTN Call Flows August 2002
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F8 ACK NGW 1 -> Proxy 1
ACK sip:+19725552222@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F9 ACM NGW 1 -> User A
ACM
/* A one way speech path is established between NGW 1 and User A. */
/* Call Released after User A hangs up. */
F10 REL User A -> NGW 1
REL
CauseCode=16
F11 RLC NGW 1 -> User A
RLC
Johnston et al Expires - February 2002 [Page 81]
SIP PSTN Call Flows August 2002
3.7 Unsuccessful PSTN->SIP, ACM timeout
Switch A NGW 1 Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | INVITE F5 |
| | |--------------->|
| | | INVITE F6 |
| | |--------------->|
| | | INVITE F7 |
| | |--------------->|
| | | INVITE F8 |
| | |--------------->|
| | | INVITE F9 |
| | |--------------->|
| REL F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| | |
| | CANCEL F12 | |
| |--------------->| |
| | 200 F13 | |
| |<---------------| |
User A calls User B through NGW 1 and Proxy 1. Proxy 1 re-sends the
INVITE after the expiration of SIP timer T1 without receiving any
response from User B. User B never responds with 180 Ringing or any
other response (it is reachable but unresponsive). After the
expiration of a timer, User A's network disconnects the call by
sending a Release message REL. The Gateway maps this to a CANCEL.
Message Details
F1 IAM User A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE A -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Johnston et al Expires - February 2002 [Page 82]
SIP PSTN Call Flows August 2002
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to User B. */
F3 INVITE Proxy 1 -> User B
INVITE sip:UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
c c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
Johnston et al Expires - February 2002 [Page 83]
SIP PSTN Call Flows August 2002
CSeq: 1 INVITE
Content-Length: 0
F5 INVITE Proxy 1 -> User B
Same as Message F3
F6 INVITE Proxy 1 -> User B
Same as Message F3
F7 INVITE Proxy 1 -> User B
Same as Message F3
F8 INVITE Proxy 1 -> User B
Same as Message F3
F9 INVITE Proxy 1 -> User B
Same as Message F3
/* Timer expires in User A's access network. */
F10 REL User A -> NGW 1
REL
CauseCode=16 Normal
F11 RLC NGW 1 -> User A
RLC
F12 CANCEL NGW 1 -> Proxy 1
CANCEL sip:+19725552222@ss11.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Johnston et al Expires - February 2002 [Page 84]
SIP PSTN Call Flows August 2002
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F13 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
Johnston et al Expires - February 2002 [Page 85]
SIP PSTN Call Flows August 2002
3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy
Switch A NGW 1 Stateless Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | INVITE F4 |--------------->|
| |--------------->| INVITE F5 |
| | INVITE F6 |--------------->|
| |--------------->| INVITE F7 |
| | INVITE F8 |--------------->|
| |--------------->| INVITE F9 |
| | INVITE F10 |--------------->|
| |--------------->| INVITE F11 |
| | INVITE F12 |--------------->|
| |--------------->| INVITE F13 |
| | |--------------->|
| REL F14 | | |
|--------------->| | |
| RLC F15 | | |
|<---------------| | |
In this scenario, User A calls User B through NGW 1 and Proxy 1.
Since Proxy 1 is stateless (it does not send a 100 Trying response),
NGW 1 re-sends the INVITE message after the expiration of
SIP timer T1. User B does not respond with 180 Ringing. User A's
network disconnects the call with a release REL (CauseCode=102
Timeout).
Message Details
F1 IAM User A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
Johnston et al Expires - February 2002 [Page 86]
SIP PSTN Call Flows August 2002
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to User B. */
F3 INVITE Proxy 1 -> User B
INVITE sip:UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
Max-Forwards: 69
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 INVITE NGW 1 -> Proxy 1
Same as Message F2
F5 INVITE Proxy 1 -> User B
Same as Message F3
Johnston et al Expires - February 2002 [Page 87]
SIP PSTN Call Flows August 2002
F6 INVITE NGW 1 -> Proxy 1
Same as Message F2
F7 INVITE Proxy 1 -> User B
Same as Message F3
F8 INVITE NGW 1 -> Proxy 1
Same as Message F2
F9 INVITE Proxy 1 -> User B
Same as Message F3
F10 INVITE NGW 1 -> Proxy 1
Same as Message F2
F11 INVITE Proxy 1 -> User B
Same as Message F3
F12 INVITE NGW 1 -> Proxy 1
Same as Message F2
F13 INVITE Proxy 1 -> User B
Same as Message F3
/* A timer expires in User A's access network. */
F14 REL User A -> NGW 1
REL
CauseCode=102 Timeout
Johnston et al Expires - February 2002 [Page 88]
SIP PSTN Call Flows August 2002
F15 RLC NGW 1 -> User A
RLC
Johnston et al Expires - February 2002 [Page 89]
SIP PSTN Call Flows August 2002
3.9 Unsuccessful PSTN->SIP, Caller Abandonment
Switch A NGW 1 Proxy 1 User B
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| ACM F7 |<---------------| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | |
|<===============| | |
| | | |
| REL F8 | | |
|--------------->| | |
| RLC F9 | | |
|<---------------| CANCEL F10 | |
| |--------------->| |
| | 200 F11 | |
| |<---------------| |
| | | CANCEL F12 |
| | |--------------->|
| | | 200 F13 |
| | |<---------------|
| | | 487 F14 |
| | |<---------------|
| | | ACK F15 |
| | 487 F16 |--------------->|
| |<---------------| |
| | ACK F17 | |
| |--------------->| |
| | | |
In this scenario, User A calls User B through NGW 1 and Proxy 1.
User B does not respond with 200 OK. NGW 1 plays ringing tone since
the ACM indicates that interworking has been encountered. User A
disconnects the call with a Release message REL which is mapped by
NGW 1 to a CANCEL. Note that if User B had sent a 200 OK response
after the REL, NGW 1 would have sent an ACK then a BYE to properly
terminate the call.
Message Details
Johnston et al Expires - February 2002 [Page 90]
SIP PSTN Call Flows August 2002
F1 IAM User A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE A -> Proxy 1
INVITE sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to User B. */
F3 INVITE Proxy 1 -> User B
INVITE sip:UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
Johnston et al Expires - February 2002 [Page 91]
SIP PSTN Call Flows August 2002
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying User B -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.201
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 180 Ringing User B -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Length: 0
F6 180 Ringing Proxy 1 -> NGW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
Johnston et al Expires - February 2002 [Page 92]
SIP PSTN Call Flows August 2002
CSeq: 1 INVITE
Contact: <sip:UserB@192.168.200.201>
Content-Length: 0
F7 ACM NGW 1 -> User A
ACM
/* User A hangs up */
F8 REL User A -> NGW 1
REL
CauseCode=16 Normal
F9 RLC NGW 1 -> User A
RLC
F10 CANCEL NGW 1 -> Proxy 1
CANCEL sip:+19725552222@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F11 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F12 CANCEL Proxy 1 -> User B
Johnston et al Expires - February 2002 [Page 93]
SIP PSTN Call Flows August 2002
CANCEL sip:UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F13 200 OK User B -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 CANCEL
Content-Length: 0
F14 487 Request Terminated User B -> Proxy 1
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F15 ACK Proxy 1 -> User B
ACK sip:UserB@biloxi.com SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F16 487 Request Terminated Proxy 1 -> NGW 1
Johnston et al Expires - February 2002 [Page 94]
SIP PSTN Call Flows August 2002
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F17 ACK NGW 1 -> Proxy 1
ACK sip:+19725552222@ss11.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
Johnston et al Expires - February 2002 [Page 95]
SIP PSTN Call Flows August 2002
4. PSTN to PSTN Dialing via SIP Network
In these scenarios, both the caller and the called party are in the
telephone network, either normal PSTN subscribers or PBX extensions.
The calls route through two Gateways and at least one SIP Proxy
Server. The Proxy Server performs the authentication and location of
the Gateways.
Again it is noted that the intent of this call flows document is not
to provide a detailed parameter level mapping of SIP to PSTN
protocols. For information on SIP to ISUP mapping, the reader is
referred to other references [4].
In these scenarios, the call is successfully completed between the
two Gateways allowing the PSTN or PBX users to communicate. The 183
Session Progress response is used to indicate in-band alerting may
flow from the called party telephone switch to the caller.
Johnston et al Expires - February 2002 [Page 96]
SIP PSTN Call Flows August 2002
4.1 Successful ISUP PSTN to ISUP PSTN call
Switch A NGW 1 Proxy 1 GW 2 Switch C
| | | | |
| IAM F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| |------------->| INVITE F3 | |
| | |------------->| IAM F4 |
| | | |------------->|
| | | | ACM F5 |
| | | 183 F6 |<-------------|
| | 183 F7 |<-------------| |
| ACM F8 |<-------------| | |
|<-------------| | | |
| One Way Voice| Two Way RTP Media | One Way Voice|
|<=============|<===========================>|<=============|
| | | | ANM F9 |
| | | 200 F10 |<-------------|
| | 200 F11 |<-------------| |
| ANM F12 |<-------------| | |
|<-------------| | | |
| | ACK F13 | | |
| |------------->| ACK F14 | |
| | |------------->| |
|Both Way Voice| Both Way RTP Media |Both Way Voice|
|<=============|<===========================>|<=============|
| | | | REL F15 |
| | | |<-------------|
| | | BYE F16 | |
| | BYE F18 |<-------------| RLC F17 |
| |<-------------| |------------->|
| | | | |
| | 200 F19 | | |
| |------------->| 200 F20 | |
| | |------------->| |
| REL F21 | | | |
|<-------------| | | |
| RLC F22 | | | |
|------------->| | | |
| | | | |
In this scenario, User A in the PSTN calls User C who is an extension
on a PBX. User A's telephone switch signals via SS7 to the Network
Gateway NGW 1, while User C's PBX signals via SS7 with the
Gateway GW 2. The CdPN and CgPN are mapped by GW1 into SIP URIs and
placed in the To and From headers. Proxy 1 looks up the dialed
digits in the Request-URI and maps the digits to the PBX extension of
Johnston et al Expires - February 2002 [Page 97]
SIP PSTN Call Flows August 2002
User C which is served by GW 2. The Proxy in F3 uses the host
portion of the Request-URI to identify what private dialing plan is
being referenced. The INVITE is then forwarded to GW 2 for call
completion. An early media path is established end-to-end so that
User A can hear the ringing tone generated by PBX C.
User C answers the call and the media path is cut through in both
directions. User B hangs up terminating the call.
Message Details
F1 IAM Switch A -> NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=918-555-3333,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1
INVITE sip:+19185553333@ss1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 consults Location Service and translates the dialed number
to a private number in the Request-URI*/
F3 INVITE Proxy 1 -> GW 2
INVITE sip:4443333@gw2.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKwqwee65
Johnston et al Expires - February 2002 [Page 98]
SIP PSTN Call Flows August 2002
;received=192.168.255.101
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+13145551111@ngw1.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844526 2890844526 IN IP4 ngw1.atlanta.com
s=-
c=IN IP4 192.168.255.101
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 IAM GW 2 -> Switch C
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=444-3333,NPI=Private,NOA=Subscriber
F5 ACM Switch C -> GW 2
ACM
/* Based on the ACM message, GW 2 returns a 183 response. In-band
call progress indications are sent to User A through NGW 1. */
F6 183 Session Progress GW 2 -> Proxy 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:4443333@gw2.atlanta.com>
Content-Type: application/sdp
Johnston et al Expires - February 2002 [Page 99]
SIP PSTN Call Flows August 2002
Content-Length: 149
v=0
o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
s=-
c=IN IP4 192.168.255.202
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F7 183 Session Progress Proxy 1 -> GW 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:4443333@gw2.atlanta.com>
Content-Type: application/sdp
Content-Length: 149
v=0
o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
s=-
c=IN IP4 192.168.255.202
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* NGW 1 receives packets from GW 2 with encoded ringback, tones or
other audio. NGW 1 decodes this and places it on the originating
trunk. */
F8 ACM NGW 1 -> Switch A
ACM
/* User B answers */
F9 ANM Switch C -> GW 2
ANM
Johnston et al Expires - February 2002 [Page 100]
SIP PSTN Call Flows August 2002
F10 200 OK GW 2 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:4443333@gw2.atlanta.com>
Content-Type: application/sdp
Content-Length: 149
v=0
o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
s=-
c=IN IP4 192.168.255.202
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F11 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:4443333@gw2.atlanta.com>
Content-Type: application/sdp
Content-Length: 149
v=0
o=GW 987654321 987654321 IN IP4 gw2.atlanta.com
s=-
c=IN IP4 192.168.255.202
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Johnston et al Expires - February 2002 [Page 101]
SIP PSTN Call Flows August 2002
F12 ANM NGW 1 -> Switch A
ANM
F13 ACK NGW 1 -> Proxy 1
ACK sip:4443333@gw2.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F14 ACK Proxy 1 -> GW 2
ACK sip:4443333@gw2.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.atlanta.com:5060;branch=z9hG4bKlueha2
;received=192.168.255.101
Max-Forwards: 69
From: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
To: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between NGW 1 and GW 2. */
/* User B Hangs Up with User A. */
F15 REL Switch C -> GW 2
REL
CauseCode=16 Normal
F16 BYE GW 2 -> Proxy 1
BYE sip:+13145551111@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
Johnston et al Expires - February 2002 [Page 102]
SIP PSTN Call Flows August 2002
From: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
To: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 4 BYE
Content-Length: 0
F17 RLC GW 2 -> Switch C
RLC
F18 BYE Proxy 1 -> NGW 1
BYE sip:+13145551111@ngw1.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6
;received=192.168.255.202
Max-Forwards: 69
From: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
To: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 4 BYE
Content-Length: 0
F19 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6
;received=192.168.255.202
From: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
To: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 4 BYE
Content-Length: 0
F20 200 OK Proxy 1 -> GW 2
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw2.atlanta.com:5060;branch=z9hG4bKtexx6
;received=192.168.255.202
From: <sip:+19185553333@ss1.atlanta.com;user=phone>;tag=314159
To: <sip:+13145551111@ngw1.atlanta.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.atlanta.com
CSeq: 4 BYE
Johnston et al Expires - February 2002 [Page 103]
SIP PSTN Call Flows August 2002
Content-Length: 0
F21 REL Switch C -> GW 2
REL
CauseCode=16 Normal
F22 RLC GW 2 -> Switch C
RLC
Johnston et al Expires - February 2002 [Page 104]
SIP PSTN Call Flows August 2002
4.2 Successful FGB PBX to ISDN PBX call with overflow
PBX A GW 1 Proxy 1 GW 2 GW 3 PBX C
| | | | | |
| Seizure | | | | |
|----------->| | | | |
| Wink | | | | |
|<-----------| | | | |
|MF Digits F1| | | | |
|----------->| | | | |
| | INVITE F2 | | | |
| |----------->| INVITE F3 | | |
| | |----------->| | |
| | | 503 F4 | | |
| | |<-----------| | |
| | | ACK F5 | | |
| | |----------->| | |
| | | INVITE F6 | |
| | |------------------------>| SETUP F7 |
| | | 100 F8 |----------->|
| | |<------------------------|CALL PROC F9|
| | | |<-----------|
| | | | ALERT F10 |
| | | 180 F11 |<-----------|
| | 180 F12 |<------------------------| |
| |<-----------| | |
| Ringtone | | |OneWay Voice|
|<===========| | |<===========|
| | | | CONNect F13|
| | | 200 F14 |<-----------|
| | 200 F15 |<------------------------| |
| Seizure |<-----------| | |
|<-----------| ACK F16 | | |
| |----------->| ACK F17 | |
| | |------------------------>|CONN ACK F18|
| | | |----------->|
|BothWayVoice| Both Way RTP Media |BothWayVoice|
|<==========>|<====================================>|<==========>|
| | | | DISC F19 |
| | | |<-----------|
| | | BYE F20 | |
| | BYE F21 |<------------------------| REL F22 |
|Seiz Removal|<-----------| |----------->|
|<-----------| 200 F23 | | |
|Seiz Removal|----------->| 200 F24 | |
|----------->| |------------------------>| REL COM F25|
| | | |<-----------|
| | | | |
Johnston et al Expires - February 2002 [Page 105]
SIP PSTN Call Flows August 2002
PBX User A calls PBX User C via Gateway GW 1 and Proxy 1. During the
attempt to reach User C via GW 2, an error is encountered - Proxy 1
receives a 503 Service Unavailable (F4) response to the forwarded
INVITE. This could be due to all circuits being busy, or some other
outage at GW 2. Proxy 1 recognizes the error and uses an alternative
route via GW 3 to terminate the call. From there, the call proceeds
normally with User C answering the call. The call is terminated when
User C hangs up.
Message Details
PBX A -> GW 1
Seizure
GW 1 -> PBX A
Wink
F1 MF Digits PBX A -> GW 1
KP 444 3333 ST
F2 INVITE GW 1 -> Proxy 1
INVITE sip:4443333@ss1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.atlanta.com>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Johnston et al Expires - February 2002 [Page 106]
SIP PSTN Call Flows August 2002
/* Proxy 1 uses a Location Service function to determine where B is
located. Response is returned listing alternative routes, GW2 and
GW3, which are then tried sequentially. */
F3 INVITE Proxy 1 -> GW 2
INVITE sip:4443333@gw2.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.atlanta.com>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 503 Service Unavailable GW 2 -> Proxy 1
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
;received=192.168.255.111
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F5 ACK Proxy 1 -> GW 2
ACK sip:4443333@ss1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.1
Johnston et al Expires - February 2002 [Page 107]
SIP PSTN Call Flows August 2002
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Max-Forward: 70
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F6 INVITE Proxy 1 -> GW 3
INVITE sip:+19185553333@gw3.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Max-Forwards: 69
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.atlanta.com>
Content-Type: application/sdp
Content-Length: 141
v=0
o=GW 2890844526 2890844526 IN IP4 gw1.atlanta.com
s=-
c=IN IP4 192.168.255.201
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F7 SETUP GW 3 -> PBX C
Protocol discriminator=Q.931
Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio)
Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN; further call
progress information may be available inband)
Called party number:
Type of number and numbering plan ID=33 (National number in ISDN
numbering plan)
Digits=918-555-3333
Johnston et al Expires - February 2002 [Page 108]
SIP PSTN Call Flows August 2002
F8 100 Trying GW 3 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Content-Length: 0
F9 CALL PROCeeding PBX C -> GW 3
Protocol discriminator=Q.931
Message type=CALL PROC
F10 ALERT PBX C -> GW 3
Protocol discriminator=Q.931
Message type=PROG
/* Based on ALERT message, GW 3 returns a 180 response. */
F11 180 Ringing GW 3 -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
;received=192.168.255.111
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.atlanta.com;user=phone>
Content-Length: 0
F12 180 Ringing Proxy 1 -> GW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
Johnston et al Expires - February 2002 [Page 109]
SIP PSTN Call Flows August 2002
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.atlanta.com;user=phone>
Content-Length: 0
F13 CONNect PBX C -> GW 3
Protocol discriminator=Q.931
Message type=CONN
F14 200 OK GW 3 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
;received=192.168.255.111
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 149
v=0
o=GW 987654321 987654321 IN IP4 gw3.atlanta.com
s=-
c=IN IP4 192.168.255.203
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F15 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Record-Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 INVITE
Johnston et al Expires - February 2002 [Page 110]
SIP PSTN Call Flows August 2002
Contact: <sip:+19185553333@gw3.atlanta.com;user=phone>
Content-Type: application/sdp
Content-Length: 149
v=0
o=GW 987654321 987654321 IN IP4 gw3.atlanta.com
s=-
c=IN IP4 192.168.255.203
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000
GW 1 -> PBX A
Seizure
F16 ACK GW 1 -> Proxy 1
ACK sip:+19185553333@gw3.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F17 ACK Proxy 1 -> GW 3
ACK sip:+19185553333@gw3.atlanta.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw1.atlanta.com:5060;branch=z9hG4bKwqwee65
;received=192.168.255.201
Max-Forwards: 69
From: <sip:551313@gw1.atlanta.com>;tag=63412s
To: <sip:4443333@ss1.atlanta.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 ACK
Content-Length: 0
F18 CONNect ACK GW 3 -> PBX C
Protocol discriminator=Q.931
Message type=CONN ACK
Johnston et al Expires - February 2002 [Page 111]
SIP PSTN Call Flows August 2002
/* RTP streams are established between GW 1 and GW 3. */
/* User B Hangs Up with User A. */
F19 DISConnect PBX C -> GW 3
Protocol discriminator=Q.931
Message type=DISC
Cause=16 (Normal clearing)
F20 BYE GW 3 -> Proxy 1
BYE sip:551313@gw1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP gw3.atlanta.com:5060;branch=z9hG4bKkdjuwq
Max-Forwards: 70
Route: <sip:ss1.atlanta.com;lr>
From: <sip:4443333@ss1.atlanta.com>;tag=123456789
To: <sip:551313@gw1.atlanta.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 BYE
Content-Length: 0
F21 BYE Proxy 1 -> GW 1
BYE sip:551313@gw1.atlanta.com SIP/2.0
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw3.atlanta.com:5060;branch=z9hG4bKkdjuwq
;received=192.168.255.203
Max-Forwards: 69
From: <sip:4443333@ss1.atlanta.com>;tag=123456789
To: <sip:551313@gw1.atlanta.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 BYE
Content-Length: 0
GW 1 -> PBX A
Seizure removal
F22 RELease GW 3 -> PBX C
Protocol discriminator=Q.931
Message type=REL
Johnston et al Expires - February 2002 [Page 112]
SIP PSTN Call Flows August 2002
F23 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.atlanta.com:5060;branch=z9hG4bK2d4790.2
;received=192.168.255.111
Via: SIP/2.0/UDP gw3.atlanta.com:5060;branch=z9hG4bKkdjuwq
;received=192.168.255.203
From: <sip:4443333@ss1.atlanta.com>;tag=123456789
To: <sip:551313@gw1.atlanta.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 BYE
Content-Length: 0
F24 200 OK Proxy 1 -> GW 3
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw3.atlanta.com:5060;branch=z9hG4bKkdjuwq
;received=192.168.255.203
From: <sip:4443333@ss1.atlanta.com>;tag=123456789
To: <sip:551313@gw1.atlanta.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.atlanta.com
CSeq: 1 BYE
Content-Length: 0
F25 RELease COMplete PBX C -> GW 3
Protocol discriminator=Q.931
Message type=REL COM
PBX A -> GW 1
Seizure removal
Security Considerations
Since this document represents NON NORMATIVE examples of SIP session
establishment, the security considerations in RFC 3261 [2] apply.
References
Johnston et al Expires - February 2002 [Page 113]
SIP PSTN Call Flows August 2002
1 Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997
2 J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: Session
Initiation Protocol", RFC 3261, June 2002.
3 J.Rosenberg and H.Schulzrinne, "An Offer/Answer Model with SDP",
Internet Engineering Task Force, RFC 3264, April 2002.
4 G. Camarillo, "Best Current Practice for ISUP to SIP Mapping",
Internet Draft, Internet Engineering Task Force, Work in
progress.
5 Johnston, A., Donovan, S., Sparks, R., Cunningham, C., Summers,
K., "Session Initiation Protocol Basic Call Flow Examples", RFC
yyyy, August 2002.
6 Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach,
P., Luotonen, A. and L. Stewart, "HTTP authentication: Basic and
Digest Access Authentication", RFC 2617, June 1999.
7 Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", RFC 2246,
January 1999.
8 S. Kent, R. Atkinson, "Security Architecture for the Internet
Protocol", RFC 2401, November 1998.
9 A. Vaha-Sipila, "URLs for Telephone Calls", Internet Draft,
Internet Engineering Task Force, RFC 2806, April 2000.
Acknowledgments
Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings,
and Tom Taylor for their detailed comments during the final final
review. Thanks to Dean Willis for his early contributions to the
development of this document.
The authors wish to thank Neil Deason for his additions to the
Torture Test messages and Kundan Singh for performing parser
validation of messages.
Johnston et al Expires - February 2002 [Page 114]
SIP PSTN Call Flows August 2002
The authors wish to thank the following individuals for their
participation in the final review of this call flows document: Aseem
Agarwal, Rafi Assadi, Ben Campbell, Sunitha Kumar, Jon Peterson, Marc
Petit-Huguenin, Vidhi Rastogi, and Bodgey Yin Shaohua.
The authors also wish to thank the following individuals for their
assistance: Jean-Francois Mule, Hemant Agrawal, Henry Sinnreich,
David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole
MCI WorldCom IPOP Design team, Scott Orton, Greg Osterhout, Pat
Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise
Ingram, Denise Caballero, Tom Redman, Ilya Slain, Pat Sollee, John
Truetken, and others from MCI WorldCom, 3Com, Cisco, Lucent and
Nortel.
Author's Addresses
All listed authors actively contributed large amounts of text to this
document.
Alan Johnston
WorldCom
100 South 4th Street
St. Louis, MO 63102
USA
EMail: alan.johnston@wcom.com
Steve Donovan
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA
EMail: sdonovan@dynamicsoft.com
Robert Sparks
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA
EMail: rsparks@dynamicsoft.com
Chris Cunningham
dynamicsoft, Inc.
Johnston et al Expires - February 2002 [Page 115]
SIP PSTN Call Flows August 2002
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA
EMail: ccunningham@dynamicsoft.com
Kevin Summers
Sonus
1701 North Collins Blvd, Suite 3000
Richardson, TX 75080
USA
Email: kevin.summers@sonusnet.com
Copyright Notice
"Copyright (C) The Internet Society 2002. All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Acknowledgement
Johnston et al Expires - February 2002 [Page 116]
SIP PSTN Call Flows August 2002
Funding for the RFC Editor function is currently provided by the
Internet Society.
Johnston et al Expires - February 2002 [Page 117]
