Network Working Group A. Johnston

Network Working Group A. Johnston
Request for Comments: 3666 MCI
BCP: 76 S. Donovan
Category: Best Current Practice R. Sparks
C. Cunningham
dynamicsoft
K. Summers
Sonus
December 2003

Session Initiation Protocol (SIP)
Public Switched Telephone Network (PSTN) Call Flows

Status of this Memo

This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.

Copyright Notice

Copyright (C) The Internet Society (2003). All Rights Reserved.

Abstract

This document contains best current practice examples of Session
Initiation Protocol (SIP) call flows showing interworking with the
Public Switched Telephone Network (PSTN). Elements in these call
flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways.
Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP.
PSTN telephony protocols are illustrated using ISDN (Integrated
Services Digital Network), ISUP (ISDN User Part), and FGB (Feature
Group B) circuit associated signaling. PSTN calls are illustrated
using global telephone numbers from the PSTN and private extensions
served on by a PBX (Private Branch Exchange). Call flow diagrams and
message details are shown.

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RFC 3666 SIP PSTN Call Flows December 2003

Table of Contents

1. Overview..................................................... 2
1.1. General Assumptions.................................... 3
1.2. Legend for Message Flows............................... 4
1.3. SIP Protocol Assumptions............................... 5
2. SIP to PSTN Dialing.......................................... 6
2.1. Successful SIP to ISUP PSTN call....................... 7
2.2. Successful SIP to ISDN PBX call........................ 15
2.3. Successful SIP to ISUP PSTN call with overflow......... 23
2.4. Session established using ENUM Query................... 32
2.5. Unsuccessful SIP to PSTN call: Treatment from PSTN..... 38
2.6. Unsuccessful SIP to PSTN: REL w/Cause from PSTN........ 45
2.7. Unsuccessful SIP to PSTN: ANM Timeout.................. 49
3. PSTN to SIP Dialing.......................................... 54
3.1. Successful PSTN to SIP call............................ 55
3.2. Successful PSTN to SIP call, Fast Answer............... 62
3.3. Successful PBX to SIP call............................. 68
3.4. Unsuccessful PSTN to SIP REL, SIP error mapped to REL.. 74
3.5. Unsuccessful PSTN to SIP REL, SIP busy mapped to REL... 76
3.6. Unsuccessful PSTN->SIP, SIP error interworking to tones 80
3.7. Unsuccessful PSTN->SIP, ACM timeout.................... 84
3.8. Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy... 88
3.9. Unsuccessful PSTN->SIP, Caller Abandonment............. 91
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
5. Security Considerations...................................... 113
6. References................................................... 115
6.1. Normative References................................... 115
6.2. Informative References................................. 115
7. Acknowledgments.............................................. 116
8. Intellectual Property Statement.............................. 116
9. Authors' Addresses........................................... 117
10. Full Copyright Statement..................................... 118

1. Overview

The call flows shown in this document were developed in the design of
a SIP IP communications network. They represent an example of a
minimum set of functionality.

It is the hope of the authors that this document will be useful for
SIP implementers, designers, and protocol researchers alike and will
help further the goal of a standard implementation of RFC 3261 [2].
These flows represent carefully checked and working group reviewed
scenarios of the most common SIP/PSTN interworking examples as a
companion to the specifications.

Johnston, et al. Best Current Practice [Page 2]

RFC 3666 SIP PSTN Call Flows December 2003

These call flows are based on the current version 2.0 of SIP in RFC
3261 [2] with SDP usage described in RFC 3264 [3]. Other RFCs also
comprise the SIP standard but are not used in this set of basic call
flows. The SIP/ISUP mapping is based on RFC 3398 [4].

Various PSTN signaling protocols are illustrated in this document:
ISDN (Integrated Services Digital Network), ISUP (ISDN User Part) and
FGB (Feature Group B) circuit associated signaling. This document
shows mainly ANSI ISUP due to its practical origins. However, as
used in this document, the usage is virtually identical to the ITU-T
International ISUP used as the reference in [4].

Basic SIP call flow examples are contained in a companion document,
RFC 3665 [10].

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 BCP 14, RFC 2119 [1].

1.1. General Assumptions

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

The authentication of SIP User Agents in these example call flows is
performed using HTTP Digest as defined in [3] and [5].

Some Proxy Servers in these call flows insert Record-Route headers
into requests to ensure that they are in the signaling path for
future message exchanges.

These flows show TLS, TCP, and UDP for transport. SCTP could also be
used. See the discussion in RFC 3261 [2] for details on the
transport issues for SIP.

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

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RFC 3666 SIP PSTN Call Flows December 2003

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 [6].

This document shows typical examples of SIP/ISUP interworking.
Although in the spirit of the SIP-T framework [7], these examples do
not represent a complete implementation of the framework. The
examples here represent more of a minimal set of examples for very
basic SIP to ISUP interworking, rather than the more complex goal of
ISUP transparency. In particular, there are NO examples of
encapsulated ISUP in this document. If present, these messages would
show S/MIME encryption due to the sensitive nature of this
information, as discussed in the SIP-T Framework security
considerations section. (Note - RFC 3204 [8] contains an example of
an INVITE with encapsulated ISUP.) See the Security Considerations
section for a more detailed discussion on the security of these call
flows.

In ISUP, the Calling Party Number is abbreviated as CgPN and the
Called Party Number is abbreviated as CdPN. Other abbreviations
include Numbering Plan Indicator (NPI) and Nature of Address (NOA).

1.2. Legend for Message Flows

Dashed lines (---) represent signaling messages that are mandatory to
the call scenario. These messages can be SIP or PSTN signaling. The
arrow indicates the direction of message flow.

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

Messages with parentheses around their name represent optional
messages.

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RFC 3666 SIP PSTN Call Flows December 2003

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 does not prescribe the flows precisely as they are
shown, but rather the flows illustrate the principles for best
practice. They are best practices usages (orderings, syntax,
selection of features for the purpose, handling of error) of SIP
methods, headers and parameters. IMPORTANT: The exact flows here
must not be copied as is by an implementer due to specific incorrect
characteristics that were introduced into the document for
convenience and are listed below. To sum up, the SIP/PSTN call flows
represent well-reviewed examples of SIP usage, which are best common
practice according to IETF consensus.

For simplicity in reading and editing the document, there are a
number of differences between some of the examples and actual SIP
messages. For example, the SIP Digest responses are not actual MD5
encodings. Call-IDs are often repeated, and CSeq counts often begin
at 1. Header fields are usually shown in the same order. Usually
only the minimum required header field set is shown, others that
would normally be present, such as Accept, Supported, Allow, etc. are
not shown.

Actors:

Element Display Name URI IP Address
------- ------------ --- ----------

User Agent Alice sip:[email protected] 192.0.2.101
User Agent Bob sip:[email protected] 192.0.2.200
Proxy Server sip:ss1.a.example.com 192.0.2.111
User Agent (Gateway) sip:gw1.a.example.com 192.0.2.201
User Agent (Gateway) sip:gw2.a.example.com 192.0.2.202
User Agent (Gateway) sip:gw3.a.example.com 192.0.2.203
User Agent (Gateway) sip:ngw1.a.example.com 192.0.2.103
User Agent (Gateway) sip:ngw2.a.example.com 192.0.2.102

Note that NGW 1 and NGW 2 also have device URIs (Contacts) of
sip:[email protected] and sip:[email protected] which resolve to
the Proxy Server sip:ss1.wcom.com using DNS SRV records.

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RFC 3666 SIP PSTN Call Flows December 2003

2. SIP to PSTN Dialing

In the following scenarios, Alice (sip:[email protected]) is a SIP
phone or other SIP-enabled device. Bob is reachable via the PSTN at
global telephone number +19725552222. Alice places a call to Bob
through a Proxy Server, Proxy 1, and a Network Gateway. In other
scenarios, Alice places calls to Carol, who is served via a PBX
(Private Branch Exchange) and is identified by a private extension
444-3333, or global number +1-918-555-3333. Note that Alice uses
his/her global telephone number +1-314-555-1111 in the From header in
the INVITE messages. This then gives the Gateway the option of using
this header to populate the calling party identification field in
subsequent signaling. Left open is the issue of how the Gateway can
determine the accuracy of the telephone number which is necessary
before passing it as a valid calling party number in the PSTN.

In these scenarios, Alice is a SIP phone or other SIP-enabled device.
Alice places a call to Bob in the PSTN or Carol on a PBX through a
Proxy Server and a Gateway.

In the failure scenarios, the call does not complete. In some cases
however, a media stream is still setup. This is due to the fact that
some failures in dialing to the PSTN result in in-band tones (busy,
reorder tones or announcements - "The number you have dialed has
changed. The new number is..."). The 183 Session Progress response
containing SDP media information is used to setup this early media
path so that the caller Alice knows the final disposition of the
call.

The media stream is either terminated by the caller after the tone or
announcement has been heard and understood, or by the Gateway after a
timer expires.

In other failure scenarios, a SS7 Release with Cause Code is mapped
to a SIP response. In these scenarios, the early media path is not
used, but the actual failure code is conveyed to the caller by the
SIP User Agent Client.

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RFC 3666 SIP PSTN Call Flows December 2003

2.1. Successful SIP to ISUP PSTN call

Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| Both Way RTP Media | One Way Voice |
|<===============================>|<===============|
| | | ANM F9 |
| | 200 F10 |<---------------|
| 200 F11 |<---------------| |
|<---------------| | |
| ACK F12 | | |
|--------------->| ACK F13 | |
| |--------------->| |
| Both Way RTP Media | Both Way Voice |
|<===============================>|<==============>|
| BYE F14 | | |
|--------------->| BYE F15 | |
| |--------------->| |
| | 200 F16 | |
| 200 F17 |<---------------| REL F18 |
|<---------------| |--------------->|
| | | RLC F19 |
| | |<---------------|
| | | |

Alice dials the globalized E.164 number +19725552222 to reach Bob.
Note that A might have only dialed the last 7 digits, or some other
dialing plan. It is assumed that the SIP User Agent Client converts
the digits into a global number and puts them into a SIP URI. Note
that tel URIs could be used instead of SIP URIs.

Alice could use either their SIP address (sip:[email protected]) or
SIP telephone number (sip:[email protected];user=phone)
in the From header. In this example, the telephone number is
included, and it is shown as being passed as calling party
identification through the Network Gateway (NGW 1) to Bob (F5). Note

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RFC 3666 SIP PSTN Call Flows December 2003

that for this number to be passed into the SS7 network, it would have
to be somehow verified for accuracy.

In this scenario, Bob answers the call, then Alice disconnects the
call. Signaling between NGW 1 and Bob's telephone switch is ANSI
ISUP. For the details of SIP to ISUP mapping, refer to [4].

In this flow, notice that the Contact returned by NGW 1 in messages
F7-11 is sip:[email protected]. This is because NGW 1 only accepts
SIP messages that come through Proxy 1 - any direct signaling will be
ignored. Since this Contact URI may be used outside of this dialog
and must be routable (Section 8.1.1.8 in RFC 3261 [2]) the Contact
URI for NGW 1 must resolve to Proxy 1. This Contact URI resolves via
DNS to Proxy 1 (sip:ss1.a.example.com) which then resolves it to
sip:ngw1.a.example.com which is the address of NGW 1.

This flow shows TCP transport.

Message Details

F1 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Proxy-Authorization: Digest username="alice", realm="a.example.com",
nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
uri="sip:[email protected];user=phone",
response="ccdca50cb091d587421457305d097458c"
Content-Type: application/sdp
Content-Length: 154

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

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RFC 3666 SIP PSTN Call Flows December 2003

F2 100 Trying Proxy 1 -> Alice

SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

/* Proxy 1 uses a Location Service function to determine the gateway
for terminating this call. The call is forwarded to NGW 1. Client
for A prepares to receive data on port 49172 from the
network.*/

F3 INVITE Proxy 1 -> NGW 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 154

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

F4 100 Trying NGW 1 -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1

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RFC 3666 SIP PSTN Call Flows December 2003

;received=192.0.2.111
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 IAM NGW 1 -> Bob

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

F6 ACM Bob -> NGW 1

ACM

F7 183 Session Progress NGW 1 -> Proxy 1

SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

/* NGW 1 sends PSTN audio (ringing) in the RTP path to A */

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RFC 3666 SIP PSTN Call Flows December 2003

F8 183 Session Progress Proxy 1 -> Alice

SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

F9 ANM Bob -> NGW 1

ANM

F10 200 OK NGW 1 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp

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RFC 3666 SIP PSTN Call Flows December 2003

Content-Length: 146

v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F11 200 OK Proxy 1 -> Alice

SIP/2.0 200 OK
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

F12 ACK Alice -> Proxy 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK

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RFC 3666 SIP PSTN Call Flows December 2003

Content-Length: 0

F13 ACK Proxy 1 -> NGW 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

/* Alice Hangs Up with Bob. */

F14 BYE Alice -> Proxy 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F15 BYE Proxy 1 -> NGW 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]

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RFC 3666 SIP PSTN Call Flows December 2003

CSeq: 2 BYE
Content-Length: 0

F16 200 OK NGW 1 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F17 200 OK Proxy 1 -> A

SIP/2.0 200 OK
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F18 REL NGW 1 -> B

REL
CauseCode=16 Normal

F19 RLC B -> NGW 1

RLC

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RFC 3666 SIP PSTN Call Flows December 2003

2.2. Successful SIP to ISDN PBX call

Alice Proxy 1 GW 1 PBX C
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| SETUP F5 |
| | |--------------->|
| | | CALL PROC F6 |
| | |<---------------|
| | | PROGress F7 |
| | 180 F8 |<---------------|
| 180 F9 |<---------------| |
|<---------------| | |
| | | One Way Voice |
| | |<===============|
| | | CONNect F10 |
| | |<---------------|
| | | CONNect ACK F11|
| | 200 F12 |--------------->|
| 200 F13 |<---------------| |
|<---------------| | |
| ACK F14 | | |
|--------------->| ACK F15 | |
| |--------------->| |
| Both Way RTP Media | Both Way Voice |
|<===============================>|<==============>|
| BYE F16 | | |
|--------------->| BYE F17 | |
| |--------------->| |
| | 200 F18 | |
| 200 F19 |<---------------| DISConnect F20 |
|<---------------| |--------------->|
| | | RELease F21 |
| | |<---------------|
| | | RELease COM F22|
| | |--------------->|
| | | |

Alice is a SIP device while Carol is connected via a Gateway (GW 1)
to a PBX. The PBX connection is via a ISDN trunk group. Alice dials
Carol's telephone number (918-555-3333) which is globalized and put
into a SIP URI.

Johnston, et al. Best Current Practice [Page 15]

RFC 3666 SIP PSTN Call Flows December 2003

The host portion of the Request-URI in the INVITE F3 is used to
identify the context (customer, trunk group, or line) in which the
private number 444-3333 is valid. Otherwise, this INVITE message
could get forwarded by GW 1 and the context of the digits could
become lost and the call unroutable.

Proxy 1 looks up the telephone number and locates the gateway that
serves Carol. Carol is identified by its extension (444-3333) in the
Request-URI sent to GW 1.

Note that the Contact URI for GW 1, as used in messages F8, F9, F12,
and F13, is sips:[email protected], which resolves directly
to the gateway.

This flow shows the use of Secure SIP (sips) URIs.

Message Details

F1 INVITE Alice -> Proxy 1

INVITE sips:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sips:[email protected]>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",
opaque="", uri="sips:[email protected];user=phone",
response="6c792f5c9fa360358b93c7fb826bf550"
Content-Type: application/sdp
Content-Length: 154

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

F2 100 Trying Proxy 1 -> Alice

SIP/2.0 100 Trying

Johnston, et al. Best Current Practice [Page 16]

RFC 3666 SIP PSTN Call Flows December 2003

Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 2 INVITE
Content-Length: 0

F3 INVITE Proxy 1 -> GW 1

INVITE sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 154

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

F4 100 Trying GW -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 2 INVITE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 17]

RFC 3666 SIP PSTN Call Flows December 2003

F5 SETUP GW 1 -> Carol

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

F6 CALL PROCeeding Carol-> GW 1

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

F7 PROGress Carol-> GW 1

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

F8 180 Ringing GW 1 -> Proxy 1

SIP/2.0 180 Ringing
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sips:[email protected]>
Content-Length: 0

Johnston, et al. Best Current Practice [Page 18]

RFC 3666 SIP PSTN Call Flows December 2003

F9 180 Ringing Proxy 1 -> Alice

SIP/2.0 180 Ringing
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sips:[email protected]>
Content-Length: 0

F10 CONNect Carol-> GW 1

Protocol discriminator=Q.931
Message type=CONN

F11 CONNect ACK GW 1 -> Carol

Protocol discriminator=Q.931
Message type=CONN ACK

F12 200 OK GW 1 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 144

v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com

Johnston, et al. Best Current Practice [Page 19]

RFC 3666 SIP PSTN Call Flows December 2003

s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F13 200 OK Proxy 1 -> Alice

SIP/2.0 200 OK
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 144

v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F14 ACK Alice -> Proxy 1

ACK sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

Johnston, et al. Best Current Practice [Page 20]

RFC 3666 SIP PSTN Call Flows December 2003

F15 ACK Proxy 1 -> GW 1

ACK sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

/* Alice Hangs Up with Bob. */

F16 BYE Alice -> Proxy 1

BYE sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 3 BYE
Content-Length: 0

F17 BYE Proxy 1 -> GW 1

BYE sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 3 BYE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 21]

RFC 3666 SIP PSTN Call Flows December 2003

F18 200 OK GW 1 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 3 BYE
Content-Length: 0

F19 200 OK Proxy 1 -> A

SIP/2.0 200 OK
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:[email protected];user=phone>
;tag=9fxced76sl
To: Carol <sips:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 3 BYE
Content-Length: 0

F20 DISConnect GW 1 -> Carol

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

F21 RELease Carol-> GW 1

Protocol discriminator=Q.931
Message type=REL

F22 RELease COMplete GW 1 -> Carol

Protocol discriminator=Q.931
Message type=REL COM

Johnston, et al. Best Current Practice [Page 22]

RFC 3666 SIP PSTN Call Flows December 2003

2.3. Successful SIP to ISUP PSTN call with overflow

Alice Proxy 1 NGW 1 NGW 2 Switch B
| | | | |
| INVITE F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| 100 F3 |------------->| | |
|<-------------| 503 F4 | | |
| |<-------------| | |
| | ACK F5 | | |
| |------------->| | |
| | INVITE F6 | |
| |---------------------------->| IAM F7 |
| | |------------->|
| | | ACM F8 |
| | 183 F9 |<-------------|
| 183 F10 |<----------------------------| |
|<-------------| | |
| Two Way RTP Media | One Way Voice|
|<==========================================>|<=============|
| | | ANM F11 |
| | 200 F12 |<-------------|
| 200 F13 |<----------------------------| |
|<-------------| | |
| ACK F14 | | |
|------------->| ACK F15 | |
| |---------------------------->| |
| Both Way RTP Media |Both Way Voice|
|<==========================================>|<============>|
| BYE F16 | | |
|------------->| BYE F17 | |
| |---------------------------->| |
| | 200 F18 | |
| 200 F19 |<----------------------------| REL F20 |
|<-------------| |------------->|
| | | RLC F21 |
| | |<-------------|
| | | |

Alice calls Bob through Proxy 1. Proxy 1 tries to route to a Network
Gateway NGW 1. NGW 1 is not available and responds with a 503
Service Unavailable (F4). The call is then routed to Network Gateway
NGW 2. Bob answers the call. The call is terminated when Alice
disconnects the call. NGW 2 and Bob's telephone switch use ANSI ISUP
signaling.

Johnston, et al. Best Current Practice [Page 23]

RFC 3666 SIP PSTN Call Flows December 2003

NGW 2 also only accepts SIP messages that come through Proxy 1, so
the Contact URI sip:[email protected] is used in this flow.

This flow shows UDP transport.

Message Details

F1 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="b59311c3ba05b401cf80b2a2c5ac51b0",
opaque="", uri="sip:[email protected];user=phone",
response="ba6ab44923fa2614b28e3e3957789ab0"
Content-Type: application/sdp
Content-Length: 154

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

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

F2 INVITE Proxy 1 -> NGW 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl

Johnston, et al. Best Current Practice [Page 24]

RFC 3666 SIP PSTN Call Flows December 2003

To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 154

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

F3 100 Trying Proxy 1 -> Alice

SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F4 503 Service Unavailable NGW 1 -> Proxy 1

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=123456789
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 25]

RFC 3666 SIP PSTN Call Flows December 2003

F5 ACK Proxy 1 -> NGW 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected]>;user=phone>
;tag=123456789
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

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

F6 INVITE Proxy 1 -> NGW 2

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 154

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

F7 IAM NGW 2 -> Bob

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

Johnston, et al. Best Current Practice [Page 26]

RFC 3666 SIP PSTN Call Flows December 2003

F8 ACM Bob -> NGW 2

ACM

F9 183 Session Progress NGW 2 -> Proxy 1

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

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

F10 183 Session Progress Proxy 1 -> Alice

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp

Johnston, et al. Best Current Practice [Page 27]

RFC 3666 SIP PSTN Call Flows December 2003

Content-Length: 146

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

F11 ANM Bob -> NGW 2

ANM

F12 200 OK NGW 2 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

F13 200 OK Proxy 1 -> Alice

SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101

Johnston, et al. Best Current Practice [Page 28]

RFC 3666 SIP PSTN Call Flows December 2003

Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

F14 ACK Alice -> Proxy 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F15 ACK Proxy 1 -> NGW 2

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK

Johnston, et al. Best Current Practice [Page 29]

RFC 3666 SIP PSTN Call Flows December 2003

Content-Length: 0

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

/* Alice Hangs Up with Bob. */

F16 BYE Alice -> Proxy 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F17 BYE Proxy 1 -> NGW 2

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F18 200 OK NGW 2 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>

Johnston, et al. Best Current Practice [Page 30]

RFC 3666 SIP PSTN Call Flows December 2003

;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F19 200 OK Proxy 1 -> Alice

SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F20 REL NGW 2 -> B

REL
CauseCode=16 Normal

F21 RLC B -> NGW 2

RLC

Johnston, et al. Best Current Practice [Page 31]

RFC 3666 SIP PSTN Call Flows December 2003

2.4. Successful SIP to SIP using ENUM Query

Alice DNS Server Proxy 3 Bob
| | | |
| ENUM Query F1 | | |
|--------------->| | |
| Response F2 | | |
|<---------------| | |
| INVITE F3 | |
|-------------------------------->| INVITE F4 |
| 100 F5 |--------------->|
|<--------------------------------| 180 F6 |
| 180 F7 |<---------------|
|<--------------------------------| |
| | 200 F8 |
| 200 F9 |<---------------|
|<--------------------------------| |
| ACK F10 | |
|-------------------------------->| ACK F11 |
| |--------------->|
| Both Way RTP Media |
|<================================================>|
| | BYE F12 |
| BYE F13 |<---------------|
|<--------------------------------| |
| 200 F14 | |
|-------------------------------->| 200 F15 |
| |--------------->|
| | |

In this scenario, Alice places a call to Bob by dialing Bob's
telephone number (9725552222). Alice's UA converts the phone number
to an E.164 number (+19725552222), and performs an ENUM query [9] on
the E.164 number (2.2.2.2.5.5.5.2.7.9.1.e164.arpa), which returns a
NAPTR record containing a SIP AOR URI for Bob
(sip:[email protected]). As a result, Alice's UA sends an
INVITE and the call completes over IP bypassing the PSTN.

The call is terminated when Bob sends a BYE message.

Message Details

F1 ENUM Query Alice -> DNS Server

2.2.2.2.5.5.5.2.7.9.1.e164.arpa

Johnston, et al. Best Current Practice [Page 32]

RFC 3666 SIP PSTN Call Flows December 2003

F2 ENUM NAPTR Set DNS Server -> Alice

$ORIGIN 2.2.2.2.5.5.5.2.7.9.1.e164.arpa.
IN NAPTR 100 10 "u" "sip+E2U"
"!^.*$!sip:[email protected]!".

F3 INVITE Alice -> Proxy 3

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 154

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

F4 INVITE Proxy 3 -> Bob

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 154

v=0
o=UserA 2890844526 2890844526 IN IP4 client.a.example.com
s=-

Johnston, et al. Best Current Practice [Page 33]

RFC 3666 SIP PSTN Call Flows December 2003

c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F5 100 Trying Proxy 3 -> Alice

SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>
Call-ID: [email protected]
CSeq: 2 INVITE
Content-Length: 0

F6 180 Ringing B -> Proxy 3

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Content-Length: 0

F7 180 Ringing Proxy 3 -> Alice

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Content-Length: 0

Johnston, et al. Best Current Practice [Page 34]

RFC 3666 SIP PSTN Call Flows December 2003

F8 200 OK Bob -> Proxy 3

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 151

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

F9 200 OK Proxy -> Alice

SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 151

v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=-
c=IN IP4 192.0.2.100
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

Johnston, et al. Best Current Practice [Page 35]

RFC 3666 SIP PSTN Call Flows December 2003

F10 ACK Alice -> Proxy 3

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
Max-Forwards: 70
Route: <sip:ss3.b.example.com;lr>
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

F11 ACK Proxy 3 -> Bob

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
;received=192.0.2.101
Max-Forwards: 69
From: <sip:[email protected]>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: [email protected]
CSeq: 2 ACK
Content-Type: application/sdp
Content-Length: 0

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

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

F12 BYE Bob -> Proxy 3

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
Max-Forwards: 70
Route: <sip:ss3.b.example.com;lr>
From: <tel:+19725552222>;tag=314159
To: <sip:[email protected]>;tag=9fxced76sl
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

F13 BYE Proxy 3 -> Alice

BYE sip:[email protected] SIP/2.0

Johnston, et al. Best Current Practice [Page 36]

RFC 3666 SIP PSTN Call Flows December 2003

Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.100
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
Max-Forwards: 69
From: <tel:+19725552222>;tag=314159
To: <sip:[email protected]>;tag=9fxced76sl
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

F14 200 OK Alice -> Proxy 3

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
;received=192.0.2.100
From: <tel:+19725552222>;tag=314159
To: <sip:[email protected]>;tag=9fxced76sl
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

F15 200 OK Proxy 3 -> Bob

SIP/2.0 200 OK
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
;received=192.0.2.100
From: <tel:+19725552222>;tag=314159
To: <sip:[email protected]>;tag=9fxced76sl
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 37]

RFC 3666 SIP PSTN Call Flows December 2003

2.5. Unsuccessful SIP to PSTN call: Treatment from PSTN

Alice Proxy 1 NGW 1 Bob
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| Two Way RTP Media | One Way Voice |
|<===============================>|<===============|
| Treatment Applied |
|<=================================================|
| CANCEL F9 | | |
|--------------->| | |
| 200 F10 | | |
|<---------------| CANCEL F11 | |
| |--------------->| |
| | 200 F12 | |
| |<---------------| REL F13 |
| | |--------------->|
| | | RLC F14 |
| | 487 F15 |<---------------|
| |<---------------| |
| | ACK F16 | |
| 487 F17 |--------------->| |
|<---------------| | |
| ACK F18 | | |
|--------------->| | |
| | | |

Alice calls Bob in the PSTN through a proxy server Proxy 1 and a
Network Gateway NGW 1. The call is rejected by the PSTN with an
in-band treatment (tone or recording) played. Alice hears the
treatment and then hangs up, which results in a CANCEL (F9) being
sent to terminate the call. (A BYE is not sent since no final
response was ever received by Alice.)

Johnston, et al. Best Current Practice [Page 38]

RFC 3666 SIP PSTN Call Flows December 2003

Message Details

F1 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40",
opaque="", uri="sip:[email protected];user=phone",
response="e178fbe430e6680a1690261af8831f40"
Content-Type: application/sdp
Content-Length: 154

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

F2 100 Trying Proxy 1 -> A

SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
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. */

Johnston, et al. Best Current Practice [Page 39]

RFC 3666 SIP PSTN Call Flows December 2003

F3 INVITE Proxy 1 -> NGW 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 154

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

F4 100 Trying NGW 1 -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 IAM NGW 1 -> Bob

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

Johnston, et al. Best Current Practice [Page 40]

RFC 3666 SIP PSTN Call Flows December 2003

F6 ACM Bob -> NGW 1

ACM

F7 183 Session Progress NGW 1 -> Proxy 1

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

F8 183 Session Progress Proxy 1 -> Alice

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

Johnston, et al. Best Current Practice [Page 41]

RFC 3666 SIP PSTN Call Flows December 2003

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

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

F9 CANCEL Alice -> Proxy 1

CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F10 200 OK Proxy 1 -> A

SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F11 CANCEL Proxy 1 -> NGW 1

CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

Johnston, et al. Best Current Practice [Page 42]

RFC 3666 SIP PSTN Call Flows December 2003

F12 200 OK NGW 1 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F13 REL NGW 1 -> B

REL
CauseCode=18 No user responding

F14 RLC B -> NGW 1

RLC

F15 487 Request Terminated NGW 1 -> Proxy 1

SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F16 ACK Proxy 1 -> NGW 1

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>

Johnston, et al. Best Current Practice [Page 43]

RFC 3666 SIP PSTN Call Flows December 2003

;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F17 487 Request Terminated Proxy 1 -> A

SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F18 ACK Alice -> Proxy 1

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

Johnston, et al. Best Current Practice [Page 44]

RFC 3666 SIP PSTN Call Flows December 2003

2.6. Unsuccessful SIP to PSTN: REL w/Cause from PSTN

Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | REL(1) F6 |
| | |<---------------|
| | | RLC F7 |
| | 404 F8 |--------------->|
| |<---------------| |
| | ACK F9 | |
| |--------------->| |
| 404 F10 | | |
|<---------------| | |
| ACK F11 | | |
|--------------->| | |
| | | |

Alice calls PSTN Bob through a Proxy Server Proxy 1 and a Network
Gateway NGW 1. The call is rejected by the PSTN with a ANSI ISUP
Release message REL containing a specific Cause code. This cause
value (1) is mapped by the Gateway to a SIP 404 Address Incomplete
response which is proxied back to Alice. For more details of ISUP
cause value to SIP response mapping, refer to [4].

Message Details

F1 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",
opaque="", uri="sip:[email protected];user=phone",

Johnston, et al. Best Current Practice [Page 45]

RFC 3666 SIP PSTN Call Flows December 2003

response="a451358d46b55512863efe1dccaa2f42"
Content-Type: application/sdp
Content-Length: 154

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

F2 100 Trying Proxy 1 -> A

SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
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:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 154

Johnston, et al. Best Current Practice [Page 46]

RFC 3666 SIP PSTN Call Flows December 2003

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

F4 100 Trying NGW 1 -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 IAM NGW 1 -> Bob

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

F6 REL Bob -> NGW 1

REL
CauseValue=1 Unallocated number

F7 RLC NGW 1 -> Bob

RLC

/* Network Gateway maps CauseValue=1 to the SIP message 404 Not
Found */

Johnston, et al. Best Current Practice [Page 47]

RFC 3666 SIP PSTN Call Flows December 2003

F8 404 Not Found NGW 1 -> Proxy 1

SIP/2.0 404 Not Found
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Error-Info: <sip:[email protected]>
Content-Length: 0

F9 ACK Proxy 1 -> NGW 1

ACK sip:[email protected];user=phone SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F10 404 Not Found Proxy 1 -> Alice

SIP/2.0 404 Not Found
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Error-Info: <sip:[email protected]>
Content-Length: 0

F11 ACK Alice -> Proxy 1

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70

Johnston, et al. Best Current Practice [Page 48]

RFC 3666 SIP PSTN Call Flows December 2003

From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

2.7. Unsuccessful SIP to PSTN: ANM Timeout

Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| | Timer on NGW 1 Expires |
| | | |
| | | REL F9 |
| | |--------------->|
| | | RLC F10 |
| | 480 F11 |<---------------|
| |<---------------| |
| | ACK F12 | |
| |--------------->| |
| 480 F13 | | |
|<---------------| | |
| ACK F14 | | |
|--------------->| | |

Alice calls Bob in the PSTN through a proxy server Proxy 1 and
Network Gateway NGW 1. The call is released by the Gateway after a
timer expires due to no ANswer Message (ANM) being received. The
Gateway sends an ISUP Release REL message to the PSTN and a 480
Temporarily Unavailable response to Alice in the SIP network.

Johnston, et al. Best Current Practice [Page 49]

RFC 3666 SIP PSTN Call Flows December 2003

Message Details

F1 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40",
opaque="", uri="sip:[email protected];user=phone",
response="579cb9db184cdc25bf816f37cbc03c7d"
Content-Type: application/sdp
Content-Length: 154

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

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

F2 100 Trying Proxy 1 -> A

SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 50]

RFC 3666 SIP PSTN Call Flows December 2003

F3 INVITE Proxy 1 -> NGW 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 154

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

F4 100 Trying NGW 1 -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 IAM NGW 1 -> Bob

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

Johnston, et al. Best Current Practice [Page 51]

RFC 3666 SIP PSTN Call Flows December 2003

F6 ACM Bob -> NGW 1

ACM

F7 183 Session Progress NGW 1 -> Proxy 1

SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

F8 183 Session Progress Proxy 1 -> Alice

SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

Johnston, et al. Best Current Practice [Page 52]

RFC 3666 SIP PSTN Call Flows December 2003

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

/* After NGW 1's timer expires, Network Gateway sends REL to ISUP
network and 480 to SIP network */

F9 REL NGW 1 -> Bob

REL

CauseCode=18 No user responding

F10 RLC Bob -> NGW 1

RLC

F11 480 Temporarily Unavailable NGW 1 -> Proxy 1

SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Error-Info: <sip:[email protected]>
Content-Length: 0

F12 ACK Proxy 1 -> NGW 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl

Johnston, et al. Best Current Practice [Page 53]

RFC 3666 SIP PSTN Call Flows December 2003

To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F13 480 Temporarily Unavailable F13 Proxy 1 -> Alice

SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Error-Info: <sip:[email protected]>
Content-Length: 0

F14 ACK Alice -> Proxy 1

ACK sip:[email protected];user=phone SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
From: Alice <sip:[email protected];user=phone>
;tag=9fxced76sl
To: Bob <sip:[email protected];user=phone>
;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

3. PSTN to SIP Dialing

In these scenarios, Alice is placing calls from the PSTN to Bob in a
SIP network. Alice's telephone switch signals to a Network Gateway
(NGW 1) using ANSI ISUP.

Since the called SIP User Agent does not send in-band signaling
information, no early media path needs to be established on the IP
side. As a result, the 183 Session Progress response is not used.
However, NGW 1 will establish a one way speech path prior to call
completion, and generate ringing for the PSTN caller. Any tones or

Johnston, et al. Best Current Practice [Page 54]

RFC 3666 SIP PSTN Call Flows December 2003

recordings are generated by NGW 1 and played in this speech path.
When the call completes successfully, NGW 1 bridges the PSTN speech
path with the IP media path.

To reduce the number of messages, only a single proxy server is shown
in these flows, which means that the a.example.com proxy server has
access to the b.example.com location service.

3.1. Successful PSTN to SIP call

Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| ACM F7 |<---------------| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | 200 F8 |
|<===============| 200 F9 |<---------------|
| |<---------------| |
| | ACK F10 | |
| ANM F12 |--------------->| ACK F11 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| REL F13 | | |
|--------------->| | |
| RLC F14 | | |
|<---------------| BYE F15 | |
| |--------------->| BYE F16 |
| | |--------------->|
| | | 200 F17 |
| | 200 F18 |<---------------|
| |<---------------| |
| | | |

In this scenario, Alice from the PSTN calls Bob through a Network
Gateway NGW1 and Proxy Server Proxy 1. When Bob answers the call,
the media path is setup end-to-end. The call terminates when Alice
hangs up the call, with Alice's telephone switch sending an ISUP
RELease message that is mapped to a BYE by NGW 1.

Johnston, et al. Best Current Practice [Page 55]

RFC 3666 SIP PSTN Call Flows December 2003

Message Details

F1 IAM Alice -> NGW 1

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

F2 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

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

Johnston, et al. Best Current Practice [Page 56]

RFC 3666 SIP PSTN Call Flows December 2003

F3 INVITE Proxy 1 -> Bob

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

F4 100 Trying Bob -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 180 Ringing Bob -> Proxy 1

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals

Johnston, et al. Best Current Practice [Page 57]

RFC 3666 SIP PSTN Call Flows December 2003

To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Length: 0

F6 180 Ringing Proxy 1 -> NGW 1

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Length: 0

F7 ACM NGW 1 -> Alice

ACM

F8 200 OK Bob -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
Contact: <sip:[email protected]>
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 151

v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=-
c=IN IP4 client.b.example.com
t=0 0
m=audio 3456 RTP/AVP 0

Johnston, et al. Best Current Practice [Page 58]

RFC 3666 SIP PSTN Call Flows December 2003

a=rtpmap:0 PCMU/8000

F9 200 OK Proxy 1 -> NGW 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 151

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

F10 ACK NGW 1 -> Proxy 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F11 ACK Proxy 1 -> Bob

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159

Johnston, et al. Best Current Practice [Page 59]

RFC 3666 SIP PSTN Call Flows December 2003

Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F12 ANM Bob -> NGW 1

ANM

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

/* Alice Hangs Up with Bob. */

F13 REL Alice -> NGW 1

REL
CauseCode=16 Normal

F14 RLC NGW 1 -> Alice

RLC

F15 BYE NGW 1-> Proxy 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F16 BYE Proxy 1 -> Bob

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]

Johnston, et al. Best Current Practice [Page 60]

RFC 3666 SIP PSTN Call Flows December 2003

CSeq: 2 BYE
Content-Length: 0

F17 200 OK Bob -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F18 200 OK Proxy 1 -> NGW 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 61]

RFC 3666 SIP PSTN Call Flows December 2003

3.2. Successful PSTN to SIP call, Fast Answer

Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 200 F5 |
| | 200 F6 |<---------------|
| |<---------------| |
| | ACK F7 | |
| ANM F9 |--------------->| ACK F8 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| REL F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| BYE F12 | |
| |--------------->| BYE F13 |
| | |--------------->|
| | | 200 F14 |
| | 200 F15 |<---------------|
| |<---------------| |
| | | |

This "fast answer" scenario is similar to 3.1., except that Bob
immediately accepts the call, sending a 200 OK (F5) without sending a
180 Ringing response. The Gateway then sends an Answer Message (ANM)
without sending an Address Complete Message (ACM). Note that for
ETSI and some other ISUP variants, a CONnect message (CON) would be
sent instead of the ANM.

Message Details

F1 IAM Alice -> NGW 1

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

F2 INVITE NGW 1 -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2

Johnston, et al. Best Current Practice [Page 62]

RFC 3666 SIP PSTN Call Flows December 2003

Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

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

F3 INVITE Proxy 1 -> Bob

INVITE [email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

Johnston, et al. Best Current Practice [Page 63]

RFC 3666 SIP PSTN Call Flows December 2003

F4 100 Trying Proxy 1 -> NGW 1

SIP/2.0 100 Trying
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 200 OK Bob -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 151

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

F6 200 OK Proxy 1 -> NGW 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>

Johnston, et al. Best Current Practice [Page 64]

RFC 3666 SIP PSTN Call Flows December 2003

Content-Type: application/sdp
Content-Length: 151

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

F7 ACK NGW 1 -> Proxy 1

ACK [email protected] SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F8 ACK Proxy 1 -> Bob

ACK [email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=130.131.132.14
Max-Forwards: 69
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F9 ANM Bob -> NGW 1

ANM

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

/* Alice Hangs Up with Bob. */

Johnston, et al. Best Current Practice [Page 65]

RFC 3666 SIP PSTN Call Flows December 2003

F10 REL ser Alice -> NGW 1

REL
CauseCode=16 Normal

F11 RLC NGW 1 -> Alice

RLC

F12 BYE NGW 1 -> Proxy 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F13 BYE Proxy 1 -> Bob

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F14 200 OK Bob -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]

Johnston, et al. Best Current Practice [Page 66]

RFC 3666 SIP PSTN Call Flows December 2003

CSeq: 2 BYE
Content-Length: 0

F15 200 OK Proxy 1 -> NGW 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 67]

RFC 3666 SIP PSTN Call Flows December 2003

3.3. Successful PBX to SIP call

PBX A GW 1 Proxy 1 Bob
| | | |
| Seizure | | |
|--------------->| | |
| Wink | | |
|<---------------| | |
| MF Digits F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| |<---------------| |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | 200 F7 |
|<===============| 200 F8 |<---------------|
| |<---------------| |
| | ACK F9 | |
| Seizure |--------------->| ACK F10 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| Seizure Removal| | |
|--------------->| | |
| Seizure Removal| | |
|<---------------| BYE F11 | |
| |--------------->| BYE F12 |
| | |--------------->|
| | | 200 F13 |
| | 200 F14 |<---------------|
| |<---------------| |
| | | |

In this scenario, Alice dials from PBX A to Bob through GW 1 and
Proxy 1. This is an example of a call that appears destined for the
PSTN but is instead routed to a SIP Client.

Signaling between PBX A and GW 1 is Feature Group B (FGB) circuit
associated signaling, in-band Mult-Frequency (MF) outpulsing. After
the receipt of the 180 Ringing from Bob, GW 1 generates a ringing
tone for Alice.

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

Johnston, et al. Best Current Practice [Page 68]

RFC 3666 SIP PSTN Call Flows December 2003

The Gateway can only identify the trunk group that the call came in
on; it cannot identify the individual line on PBX A that is placing
the call. The SIP URI used to identify the caller is shown in these
flows as sip:[email protected].

Message Details

PBX Alice -> GW 1

Seizure

GW 1 -> PBX A

Wink

F1 MF Digits PBX Alice -> GW 1

KP 1 972 555 2222 ST

F2 INVITE GW 1 -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];user=phone>
Content-Type: application/sdp
Content-Length: 146

v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/* Proxy 1 uses a Location Service function to determine where the
phone number +19725552222 is located. Based upon location
analysis the call is forwarded to SIP Bob. */

Johnston, et al. Best Current Practice [Page 69]

RFC 3666 SIP PSTN Call Flows December 2003

F3 INVITE Proxy 1 -> Bob

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];user=phone>
Content-Type: application/sdp
Content-Length: 146

v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F4 100 Trying Proxy 1 -> GW 1

SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 180 Ringing Bob -> Proxy 1

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]

Johnston, et al. Best Current Practice [Page 70]

RFC 3666 SIP PSTN Call Flows December 2003

CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Length: 0

F6 180 Ringing Proxy 1 -> GW 1

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Length: 0

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

F7 200 OK Bob -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
Contact: <sip:[email protected]>
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 151

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

Johnston, et al. Best Current Practice [Page 71]

RFC 3666 SIP PSTN Call Flows December 2003

F8 200 OK Proxy 1 -> GW 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 151

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

F9 ACK GW 1 -> Proxy 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F10 ACK Proxy 1 -> Bob

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

Johnston, et al. Best Current Practice [Page 72]

RFC 3666 SIP PSTN Call Flows December 2003

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

/* Alice Hangs Up with Bob. */

F11 BYE GW 1 -> Proxy 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F12 BYE Proxy 1 -> Bob

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

F13 200 OK Bob -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 73]

RFC 3666 SIP PSTN Call Flows December 2003

F14 200 OK Proxy 1 -> GW 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=jwdkallkzm
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

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

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

Alice attempts to place a call through Gateway GW 1 and Proxy 1,
which is unable to find any routing for the number. The call is
rejected by Proxy 1 with a REL message containing a specific Cause
value mapped by the gateway based on the SIP error.

Message Details

F1 IAM Alice -> GW 1

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

F2 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=076342s

Johnston, et al. Best Current Practice [Page 74]

RFC 3666 SIP PSTN Call Flows December 2003

To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact:
<sip:[email protected];user=phone;transport=tcp>
Content-Type: application/sdp
Content-Length: 144

v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

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

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

SIP/2.0 604 Does Not Exist Anywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=076342s
To: <sip:[email protected];user=phone>;tag=6a34d410
Call-ID: [email protected]
CSeq: 1 INVITE
Error-Info: <sip:[email protected]>
Content-Length: 0

F4 ACK GW 1 -> Proxy 1

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=076342s
To: <sip:[email protected];user=phone>;tag=6a34d410
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

Johnston, et al. Best Current Practice [Page 75]

RFC 3666 SIP PSTN Call Flows December 2003

F5 REL GW 1 -> Alice

REL
CauseCode=1

F6 RLC Alice -> GW 1

RLC

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

Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 600 F5 |
| | |<---------------|
| | | ACK F6 |
| | 600 F7 |--------------->|
| |<---------------| |
| | ACK F8 | |
| |--------------->| |
| REL(17) F9 | | |
|<---------------| | |
| RLC F10 | | |
|<-------------->| | |
| | | |

In this scenario, Alice calls Bob through Network Gateway NGW 1 and
Proxy 1. The call is routed to Bob by Proxy 1. The call is rejected
by Bob who sends a 600 Busy Everywhere response. The Gateway sends a
REL message containing a specific Cause value mapped by the gateway
based on the SIP error.

Since no interworking is indicated in the IAM (F1), the busy tone is
generated locally by Alice's telephone switch. In some scenarios,
the busy signal is generated by the Gateway since interworking is
indicated. For more discussion on interworking, refer to [4].

Johnston, et al. Best Current Practice [Page 76]

RFC 3666 SIP PSTN Call Flows December 2003

Message Details

F1 IAM Alice -> NGW 1

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

F2 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 144

v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */

F3 INVITE F3 Proxy 1 -> Bob

INVITE [email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp

Johnston, et al. Best Current Practice [Page 77]

RFC 3666 SIP PSTN Call Flows December 2003

Content-Length: 144

v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F4 100 Trying Proxy 1 -> NGW 1

SIP/2.0 100 Trying
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 600 Busy Everywhere Bob -> Proxy 1

SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F6 ACK Proxy 1 -> Bob

ACK [email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

Johnston, et al. Best Current Practice [Page 78]

RFC 3666 SIP PSTN Call Flows December 2003

F7 600 Busy Everywhere Proxy 1 -> NGW 1

SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F8 ACK NGW 1 -> Proxy 1

ACK [email protected] SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F9 REL NGW 1 -> Alice

REL
CauseCode=17 Busy

F10 RLC Alice -> NGW 1

RLC

Johnston, et al. Best Current Practice [Page 79]

RFC 3666 SIP PSTN Call Flows December 2003

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

Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 600 F5 |
| | |<---------------|
| | | ACK F6 |
| | 600 F7 |--------------->|
| |<---------------| |
| | ACK F8 | |
| ACM F9 |--------------->| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Busy Tone | | |
|<===============| | |
| REL(16) F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| | |
| | | |

In this scenario, Alice calls Bob through Network Gateway NGW 1 and
Proxy 1. The call is routed to Bob by Proxy 1. The call is rejected
by the Bob client. NGW 1 sets up a two way voice path to Alice and
plays busy tone. The caller then disconnects

NGW 1 plays the busy tone since the IAM (F1) indicates the
interworking is present. In scenario 5.2.2., with no interworking,
the busy indication is carried in the REL Cause value and is
generated locally instead.

Again, note that for ETSI or ITU ISUP, a CONnect message would be
sent instead of the Answer Message.

Message Details

F1 IAM Alice -> NGW 1

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

Johnston, et al. Best Current Practice [Page 80]

RFC 3666 SIP PSTN Call Flows December 2003

F2 INVITE NGW1 -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */

F3 INVITE Proxy 1 -> Bob

INVITE [email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

Johnston, et al. Best Current Practice [Page 81]

RFC 3666 SIP PSTN Call Flows December 2003

a=rtpmap:0 PCMU/8000

F4 100 Trying Bob -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 600 Busy Everywhere Bob -> Proxy 1

SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F6 ACK Proxy 1 -> Bob

ACK [email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

Johnston, et al. Best Current Practice [Page 82]

RFC 3666 SIP PSTN Call Flows December 2003

F7 600 Busy Everywhere Proxy 1 -> NGW 1

SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F8 ACK NGW 1 -> Proxy 1

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F9 ACM NGW 1 -> Alice

ACM

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

/* Call Released after Alice hangs up. */

F10 REL Alice -> NGW 1

REL
CauseCode=16

F11 RLC NGW 1 -> Alice

RLC

Johnston, et al. Best Current Practice [Page 83]

RFC 3666 SIP PSTN Call Flows December 2003

3.7. Unsuccessful PSTN->SIP, ACM timeout

Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | INVITE F5 |
| | |--------------->|
| | | INVITE F6 |
| | |--------------->|
| | | INVITE F7 |
| | |--------------->|
| | | INVITE F8 |
| | |--------------->|
| | | INVITE F9 |
| | |--------------->|
| REL F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| | |
| | CANCEL F12 | |
| |--------------->| |
| | 200 F13 | |
| |<---------------| |

Alice calls Bob through NGW 1 and Proxy 1. Proxy 1 re-sends the
INVITE after the expiration of SIP timer T1 without receiving any
response from Bob. Bob never responds with 180 Ringing or any other
response (it is reachable but unresponsive). After the expiration of
a timer, Alice's network disconnects the call by sending a Release
message REL. The Gateway maps this to a CANCEL.

Message Details

F1 IAM Alice -> NGW 1

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

F2 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2

Johnston, et al. Best Current Practice [Page 84]

RFC 3666 SIP PSTN Call Flows December 2003

Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */

F3 INVITE Proxy 1 -> Bob

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

Johnston, et al. Best Current Practice [Page 85]

RFC 3666 SIP PSTN Call Flows December 2003

F4 100 Trying Proxy 1 -> NGW 1

SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 INVITE Proxy 1 -> Bob

Same as Message F3

F6 INVITE Proxy 1 -> Bob

Same as Message F3

F7 INVITE Proxy 1 -> Bob

Same as Message F3

F8 INVITE Proxy 1 -> Bob

Same as Message F3

F9 INVITE Proxy 1 -> Bob

Same as Message F3

/* Timer expires in Alice's access network. */

F10 REL Alice -> NGW 1

REL
CauseCode=16 Normal

F11 RLC NGW 1 -> Alice

RLC

Johnston, et al. Best Current Practice [Page 86]

RFC 3666 SIP PSTN Call Flows December 2003

F12 CANCEL NGW 1 -> Proxy 1

CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F13 200 OK Proxy 1 -> NGW 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

Johnston, et al. Best Current Practice [Page 87]

RFC 3666 SIP PSTN Call Flows December 2003

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

Switch A NGW 1 Stateless Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | INVITE F4 |--------------->|
| |--------------->| INVITE F5 |
| | INVITE F6 |--------------->|
| |--------------->| INVITE F7 |
| | INVITE F8 |--------------->|
| |--------------->| INVITE F9 |
| | INVITE F10 |--------------->|
| |--------------->| INVITE F11 |
| | INVITE F12 |--------------->|
| |--------------->| INVITE F13 |
| | |--------------->|
| REL F14 | | |
|--------------->| | |
| RLC F15 | | |
|<---------------| | |

In this scenario, Alice calls Bob through NGW 1 and Proxy 1. Since
Proxy 1 is stateless (it does not send a 100 Trying response), NGW 1
re-sends the INVITE message after the expiration of SIP timer T1.
Bob does not respond with 180 Ringing. Alice's network disconnects
the call with a release REL (CauseCode=102 Timeout).

Message Details

F1 IAM Alice -> NGW 1

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

F2 INVITE NGW 1 -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE

Johnston, et al. Best Current Practice [Page 88]

RFC 3666 SIP PSTN Call Flows December 2003

Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */

F3 INVITE Proxy 1 -> Bob

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
Max-Forwards: 69
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 146

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

F4 INVITE NGW 1 -> Proxy 1

Same as Message F2

F5 INVITE Proxy 1 -> Bob

Same as Message F3

Johnston, et al. Best Current Practice [Page 89]

RFC 3666 SIP PSTN Call Flows December 2003

F6 INVITE NGW 1 -> Proxy 1

Same as Message F2

F7 INVITE Proxy 1 -> Bob

Same as Message F3

F8 INVITE NGW 1 -> Proxy 1

Same as Message F2

F9 INVITE Proxy 1 -> Bob

Same as Message F3

F10 INVITE NGW 1 -> Proxy 1

Same as Message F2

F11 INVITE Proxy 1 -> Bob

Same as Message F3

F12 INVITE NGW 1 -> Proxy 1

Same as Message F2

F13 INVITE Proxy 1 -> Bob

Same as Message F3

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

F14 REL Alice -> NGW 1

REL
CauseCode=102 Timeout

Johnston, et al. Best Current Practice [Page 90]

RFC 3666 SIP PSTN Call Flows December 2003

F15 RLC NGW 1 -> Alice

RLC

3.9. Unsuccessful PSTN->SIP, Caller Abandonment

Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| ACM F7 |<---------------| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | |
|<===============| | |
| | | |
| REL F8 | | |
|--------------->| | |
| RLC F9 | | |
|<---------------| CANCEL F10 | |
| |--------------->| |
| | 200 F11 | |
| |<---------------| |
| | | CANCEL F12 |
| | |--------------->|
| | | 200 F13 |
| | |<---------------|
| | | 487 F14 |
| | |<---------------|
| | | ACK F15 |
| | 487 F16 |--------------->|
| |<---------------| |
| | ACK F17 | |
| |--------------->| |
| | | |

In this scenario, Alice calls Bob through NGW 1 and Proxy 1. Bob
does not respond with 200 OK. NGW 1 plays ringing tone since the ACM
indicates that interworking has been encountered. Alice disconnects
the call with a Release message REL which is mapped by NGW 1 to a

Johnston, et al. Best Current Practice [Page 91]

RFC 3666 SIP PSTN Call Flows December 2003

CANCEL. Note that if Bob had sent a 200 OK response after the REL,
NGW 1 would have sent an ACK and then a BYE to properly terminate the
call.

Message Details

F1 IAM Alice -> NGW 1

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

F2 INVITE Alice -> Proxy 1

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */

F3 INVITE Proxy 1 -> Bob

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>

Johnston, et al. Best Current Practice [Page 92]

RFC 3666 SIP PSTN Call Flows December 2003

Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Type: application/sdp
Content-Length: 146

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

F4 100 Trying Bob -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 180 Ringing Bob -> Proxy 1

SIP/2.0 180 Ringing
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=tcp>
Content-Length: 0

F6 180 Ringing Proxy 1 -> NGW 1

SIP/2.0 180 Ringing

Johnston, et al. Best Current Practice [Page 93]

RFC 3666 SIP PSTN Call Flows December 2003

Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Length: 0

F7 ACM NGW 1 -> Alice

ACM

/* Alice hangs up */

F8 REL Alice -> NGW 1

REL
CauseCode=16 Normal

F9 RLC NGW 1 -> Alice

RLC

F10 CANCEL NGW 1 -> Proxy 1

CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F11 200 OK Proxy 1 -> NGW 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>

Johnston, et al. Best Current Practice [Page 94]

RFC 3666 SIP PSTN Call Flows December 2003

Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F12 CANCEL Proxy 1 -> Bob

CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F13 200 OK Bob -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

F14 487 Request Terminated Bob -> Proxy 1

SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F15 ACK Proxy 1 -> Bob

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70

Johnston, et al. Best Current Practice [Page 95]

RFC 3666 SIP PSTN Call Flows December 2003

From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F16 487 Request Terminated Proxy 1 -> NGW 1

SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F17 ACK NGW 1 -> Proxy 1

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=7643kals
To: <sip:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

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 that in-band alerting
may flow from the called party telephone switch to the caller.

Johnston, et al. Best Current Practice [Page 96]

RFC 3666 SIP PSTN Call Flows December 2003

4.1. Successful ISUP PSTN to ISUP PSTN call

Switch A NGW 1 Proxy 1 GW 2 Switch C
| | | | |
| IAM F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| |------------->| INVITE F3 | |
| | |------------->| IAM F4 |
| | | |------------->|
| | | | ACM F5 |
| | | 183 F6 |<-------------|
| | 183 F7 |<-------------| |
| ACM F8 |<-------------| | |
|<-------------| | | |
| One Way Voice| Two Way RTP Media | One Way Voice|
|<=============|<===========================>|<=============|
| | | | ANM F9 |
| | | 200 F10 |<-------------|
| | 200 F11 |<-------------| |
| ANM F12 |<-------------| | |
|<-------------| | | |
| | ACK F13 | | |
| |------------->| ACK F14 | |
| | |------------->| |
|Both Way Voice| Both Way RTP Media |Both Way Voice|
|<=============|<===========================>|<=============|
| | | | REL F15 |
| | | |<-------------|
| | | BYE F16 | |
| | BYE F18 |<-------------| RLC F17 |
| |<-------------| |------------->|
| | | | |
| | 200 F19 | | |
| |------------->| 200 F20 | |
| | |------------->| |
| REL F21 | | | |
|<-------------| | | |
| RLC F22 | | | |
|------------->| | | |
| | | | |

In this scenario, Alice in the PSTN calls Carol who is an extension
on a PBX. Alice's telephone switch signals via SS7 to the Network
Gateway NGW 1, while Carol's PBX signals via SS7 with the Gateway GW
2. The CdPN and CgPN are mapped by GW 1 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 Carol, which

Johnston, et al. Best Current Practice [Page 97]

RFC 3666 SIP PSTN Call Flows December 2003

is served by GW 2. The Proxy in F3 uses the host portion of the
Request-URI to identify what private dialing plan is being
referenced. The INVITE is then forwarded to GW 2 for call
completion. An early media path is established end-to-end so that
Alice can hear the ringing tone generated by PBX C.

Carol answers the call and the media path is cut through in both
directions. Bob hangs up terminating the call.

Message Details

F1 IAM Switch Alice -> NGW 1

IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=918-555-3333,NPI=E.164,NOA=National

F2 INVITE NGW 1 -> Proxy 1

INVITE sips:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 146

v=0
o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/* Proxy 1 consults Location Service and translates the dialed number
to a private number in the Request-URI*/

F3 INVITE Proxy 1 -> GW 2

INVITE sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKwqwee65

Johnston, et al. Best Current Practice [Page 98]

RFC 3666 SIP PSTN Call Flows December 2003

;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 146

v=0
o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F4 IAM GW 2 -> Switch C

IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=444-3333,NPI=Private,NOA=Subscriber

F5 ACM Switch C -> GW 2

ACM

/* Based on the ACM message, GW 2 returns a 183 response. In-band
call progress indications are sent to Alice through NGW 1. */

F6 183 Session Progress GW 2 -> Proxy 1

SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sips:[email protected]>

Johnston, et al. Best Current Practice [Page 99]

RFC 3666 SIP PSTN Call Flows December 2003

Content-Type: application/sdp
Content-Length: 143

v=0
o=GW 987654321 987654321 IN IP4 gw2.a.example.com
s=-
c=IN IP4 gw2.a.example.com
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F7 183 Session Progress Proxy 1 -> GW 1

SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 143

v=0
o=GW 987654321 987654321 IN IP4 gw2.a.example.com
s=-
c=IN IP4 gw2.a.example.com
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/* NGW 1 receives packets from GW 2 with encoded ringback, tones or
other audio. NGW 1 decodes this and places it on the originating
trunk. */

F8 ACM NGW 1 -> Switch A

ACM

/* Bob answers */

Johnston, et al. Best Current Practice [Page 100]

RFC 3666 SIP PSTN Call Flows December 2003

F9 ANM Switch C -> GW 2

ANM

F10 200 OK GW 2 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 143

v=0
o=GW 987654321 987654321 IN IP4 gw2.a.example.com
s=-
c=IN IP4 gw2.a.example.com
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F11 200 OK Proxy 1 -> NGW 1

SIP/2.0 200 OK
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sips:[email protected]>
Content-Type: application/sdp
Content-Length: 143

v=0
o=GW 987654321 987654321 IN IP4 gw2.a.example.com
s=-
c=IN IP4 gw2.a.example.com

Johnston, et al. Best Current Practice [Page 101]

RFC 3666 SIP PSTN Call Flows December 2003

t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F12 ANM NGW 1 -> Switch A

ANM

F13 ACK NGW 1 -> Proxy 1

ACK sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F14 ACK Proxy 1 -> GW 2

ACK sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sips:[email protected];user=phone>;tag=7643kals
To: <sips:[email protected];user=phone>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

/* RTP streams are established between NGW 1 and GW 2. */

/* Bob Hangs Up with Alice. */

F15 REL Switch C -> GW 2

REL
CauseCode=16 Normal

Johnston, et al. Best Current Practice [Page 102]

RFC 3666 SIP PSTN Call Flows December 2003

F16 BYE GW 2 -> Proxy 1

BYE sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: <sips:[email protected];user=phone>;tag=314159
To: <sips:[email protected];user=phone>;tag=7643kals
Call-ID: [email protected]
CSeq: 4 BYE
Content-Length: 0

F17 RLC GW 2 -> Switch C

RLC

F18 BYE Proxy 1 -> NGW 1

BYE sips:[email protected] SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202
Max-Forwards: 69
From: <sips:[email protected];user=phone>;tag=314159
To: <sips:[email protected];user=phone>;tag=7643kals
Call-ID: [email protected]
CSeq: 4 BYE
Content-Length: 0

F19 200 OK NGW 1 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202
From: <sips:[email protected];user=phone>;tag=314159
To: <sips:[email protected];user=phone>;tag=7643kals
Call-ID: [email protected]
CSeq: 4 BYE
Content-Length: 0

Johnston, et al. Best Current Practice [Page 103]

RFC 3666 SIP PSTN Call Flows December 2003

F20 200 OK Proxy 1 -> GW 2

SIP/2.0 200 OK
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202
From: <sips:[email protected];user=phone>;tag=314159
To: <sips:[email protected];user=phone>;tag=7643kals
Call-ID: [email protected]
CSeq: 4 BYE
Content-Length: 0

F21 REL Switch C -> GW 2

REL
CauseCode=16 Normal

F22 RLC GW 2 -> Switch C

RLC

Johnston, et al. Best Current Practice [Page 104]

RFC 3666 SIP PSTN Call Flows December 2003

4.2. Successful FGB PBX to ISDN PBX call with overflow

PBX A GW 1 Proxy 1 GW 2 GW 3 PBX C
| | | | | |
| Seizure | | | | |
|----------->| | | | |
| Wink | | | | |
|<-----------| | | | |
|MF Digits F1| | | | |
|----------->| | | | |
| | INVITE F2 | | | |
| |----------->| INVITE F3 | | |
| | |----------->| | |
| | | 503 F4 | | |
| | |<-----------| | |
| | | ACK F5 | | |
| | |----------->| | |
| | | INVITE F6 | |
| | |------------------------>| SETUP F7 |
| | | 100 F8 |----------->|
| | |<------------------------|CALL PROC F9|
| | | |<-----------|
| | | | ALERT F10 |
| | | 180 F11 |<-----------|
| | 180 F12 |<------------------------| |
| |<-----------| | |
| Ringtone | | |OneWay Voice|
|<===========| | |<===========|
| | | | CONNect F13|
| | | 200 F14 |<-----------|
| | 200 F15 |<------------------------| |
| Seizure |<-----------| | |
|<-----------| ACK F16 | | |
| |----------->| ACK F17 | |
| | |------------------------>|CONN ACK F18|
| | | |----------->|
|BothWayVoice| Both Way RTP Media |BothWayVoice|
|<==========>|<====================================>|<==========>|
| | | | DISC F19 |
| | | |<-----------|
| | | BYE F20 | |
| | BYE F21 |<------------------------| REL F22 |
|Seiz Removal|<-----------| |----------->|
|<-----------| 200 F23 | | |
|Seiz Removal|----------->| 200 F24 | |
|----------->| |------------------------>| REL COM F25|
| | | |<-----------|
| | | | |

Johnston, et al. Best Current Practice [Page 105]

RFC 3666 SIP PSTN Call Flows December 2003

PBX Alice calls PBX Carol via Gateway GW 1 and Proxy 1. During the
attempt to reach Carol via GW 2, an error is encountered - Proxy 1
receives a 503 Service Unavailable (F4) response to the forwarded
INVITE. This could be due to all circuits being busy, or some other
outage at GW 2. Proxy 1 recognizes the error and uses an alternative
route via GW 3 to terminate the call. From there, the call proceeds
normally with Carol answering the call. The call is terminated when
Carol hangs up.

Message Details

PBX Alice -> GW 1

Seizure

GW 1 -> PBX A

Wink

F1 MF Digits PBX Alice -> GW 1

KP 444 3333 ST

F2 INVITE GW 1 -> Proxy 1

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 155

v=0
o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000

Johnston, et al. Best Current Practice [Page 106]

RFC 3666 SIP PSTN Call Flows December 2003

/* Proxy 1 uses a Location Service function to determine where B is
located. Response is returned listing alternative routes, GW2 and
GW3, which are then tried sequentially. */

F3 INVITE Proxy 1 -> GW 2

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 155

v=0
o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F4 503 Service Unavailable GW 2 -> Proxy 1

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>;tag=314159
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F5 ACK Proxy 1 -> GW 2

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1

Johnston, et al. Best Current Practice [Page 107]

RFC 3666 SIP PSTN Call Flows December 2003

Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forward: 70
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>;tag=314159
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F6 INVITE Proxy 1 -> GW 3

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 155

v=0
o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F7 SETUP GW 3 -> PBX C

Protocol discriminator=Q.931
Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio)
Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN; further call
progress information may be available inband)
Called party number:
Type of number and numbering plan ID=33 (National number in ISDN
numbering plan)
Digits=918-555-3333

Johnston, et al. Best Current Practice [Page 108]

RFC 3666 SIP PSTN Call Flows December 2003

F8 100 Trying GW 3 -> Proxy 1

SIP/2.0 100 Trying
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0

F9 CALL PROCeeding PBX C -> GW 3

Protocol discriminator=Q.931
Message type=CALL PROC

F10 ALERT PBX C -> GW 3

Protocol discriminator=Q.931
Message type=PROG

/* Based on ALERT message, GW 3 returns a 180 response. */

F11 180 Ringing GW 3 -> Proxy 1

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>;tag=123456789
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];user=phone>
Content-Length: 0

F12 180 Ringing Proxy 1 -> GW 1

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65

Johnston, et al. Best Current Practice [Page 109]

RFC 3666 SIP PSTN Call Flows December 2003

;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>;tag=123456789
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];user=phone>
Content-Length: 0

F13 CONNect PBX C -> GW 3

Protocol discriminator=Q.931
Message type=CONN

F14 200 OK GW 3 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>;tag=123456789
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];user=phone>
Content-Type: application/sdp
Content-Length: 143

v=0
o=GW 987654321 987654321 IN IP4 gw3.a.example.com
s=-
c=IN IP4 gw3.a.example.com
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000

F15 200 OK Proxy 1 -> GW 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=63412s

Johnston, et al. Best Current Practice [Page 110]

RFC 3666 SIP PSTN Call Flows December 2003

To: <sip:[email protected]>;tag=123456789
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected];user=phone>
Content-Type: application/sdp
Content-Length: 143

v=0
o=GW 987654321 987654321 IN IP4 gw3.a.example.com
s=-
c=IN IP4 gw3.a.example.com
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000

GW 1 -> PBX A

Seizure

F16 ACK GW 1 -> Proxy 1

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>;tag=123456789
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

F17 ACK Proxy 1 -> GW 3

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
From: <sip:[email protected]>;tag=63412s
To: <sip:[email protected]>;tag=123456789
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

Johnston, et al. Best Current Practice [Page 111]

RFC 3666 SIP PSTN Call Flows December 2003

F18 CONNect ACK GW 3 -> PBX C

Protocol discriminator=Q.931
Message type=CONN ACK

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

/* Bob Hangs Up with Alice. */

F19 DISConnect PBX C -> GW 3

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

F20 BYE GW 3 -> Proxy 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:[email protected]>;tag=123456789
To: <sip:[email protected]>;tag=63412s
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

F21 BYE Proxy 1 -> GW 1

BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
;received=192.0.2.203
Max-Forwards: 69
From: <sip:[email protected]>;tag=123456789
To: <sip:[email protected]>;tag=63412s
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

GW 1 -> PBX A

Seizure removal

Johnston, et al. Best Current Practice [Page 112]

RFC 3666 SIP PSTN Call Flows December 2003

F22 RELease GW 3 -> PBX C

Protocol discriminator=Q.931
Message type=REL

F23 200 OK GW 1 -> Proxy 1

SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
;received=192.0.2.203
From: <sip:[email protected]>;tag=123456789
To: <sip:[email protected]>;tag=63412s
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

F24 200 OK Proxy 1 -> GW 3

SIP/2.0 200 OK
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
;received=192.0.2.203
From: <sip:[email protected]>;tag=123456789
To: <sip:[email protected]>;tag=63412s
Call-ID: [email protected]
CSeq: 1 BYE
Content-Length: 0

F25 RELease COMplete PBX C -> GW 3

Protocol discriminator=Q.931
Message type=REL COM

PBX Alice -> GW 1

Seizure removal

5. Security Considerations

This document provides examples of mapping from SIP to ISUP and ISUP
to SIP. The gateways in these examples are compliant with the
Security Considerations Section of RFC 3398 [4] which is summarized
here.

Johnston, et al. Best Current Practice [Page 113]

RFC 3666 SIP PSTN Call Flows December 2003

There are few security concerns relating to the mapping of ISUP to
SIP besides privacy considerations in the calling party number
passing. Some concerns relating to the mapping from tel URI
parameters to ISUP include the user creation of parameters and codes
relating to called number and local number portability (LNP). An
operator of a gateway should use policies similar to those present in
PSTN switches to avoid security problems.

The mapping from a SIP response code to an ISUP Cause Code presents a
theoretical risk, so a gateway operator may implement policies
controlling this mapping. Gateways should also not rely on the
contents of the From header field for identity information, as it may
be arbitrarily populated by a user. Instead, some sort of
cryptographic authentication and authorization should be used for
identity determination. These flows show both HTTP Digest for
authentication of users, although for brevity, the challenge is not
always shown.

The early media cut-through shown in some flows is another potential
security risk, but it is also required for proper interaction with
the PSTN. Again, a gateway operator should use proper policies
relating to early media to prevent fraud and misuse. Finally, a user
agent (even a properly authenticated one) can launch multiple
simultaneous requests through a gateway, constituting a denial of
service attack. The adoption of policies to limit the number of
simultaneous requests from a single entity may be used to prevent
this attack.

As discussed in the SIP-T framework [7], SIP/ISUP interworking can be
employed as an interdomain signaling mechanism that may be subject to
pre-existing trust relationships between administrative domains. Any
administrative domain implementing SIP-T or SIP/ISUP interworking
should have an adequate security apparatus (including elements that
manage any appropriate policies to manage fraud and billing in an
interdomain environment) in place to ensure that the translation of
ISUP information does not result in any security violations.

Although no examples of this are shown in this document, transporting
ISUP in SIP bodies may provide opportunities for abuse, fraud, and
privacy concerns, especially when SIP-T requests can be generated,
inspected or modified by arbitrary SIP endpoints. ISUP MIME bodies
should be secured (preferably with S/MIME as detailed in RFC 3261
[2]) to alleviate this concern. Authentication properties provided
by S/MIME would allow the recipient of a SIP-T message to ensure that
the ISUP MIME body was generated by an authorized entity. Encryption
would ensure that only carriers possessing a particular decryption
key are capable of inspecting encapsulated ISUP MIME bodies in a SIP
request.

Johnston, et al. Best Current Practice [Page 114]

RFC 3666 SIP PSTN Call Flows December 2003

6. References

6.1. Normative References

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

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

[3] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
the Session Description Protocol (SDP)", RFC 3264, June 2002.

[4] Camarillo, G., Roach, A. B., Peterson, J. and L. Ong,
"Integrated Services Digital Network (ISDN) User Part (ISUP) to
Session Initiation Protocol (SIP) Mapping", RFC 3398, December
2002.

[5] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A. and L. Stewart, "HTTP Authentication:
Basic and Digest Access Authentication", RFC 2617, June 1999.

[6] Vaha-Sipila, A., "URLs for Telephone Calls", RFC 2806, April
2000.

[7] Vemuri, A. and J. Peterson, "Session Initiation Protocol for
Telephones (SIP-T): Context and Architectures", BCP 63, RFC
3372, September 2002.

[8] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F.,
Watson, M. and M. Zonoun, "MIME media types for ISUP and QSIG
Objects", RFC 3204, December 2001.

[9] Faltstrom, P., "E.164 number and DNS", RFC 2916, September 2000.

6.2. Informative References

[10] Johnston, A., Donovan, S., Sparks, R., Cunningham, C. and K.
Summers, "Session Initiation Protocol (SIP) Basic Call Flow
Examples", RFC 3665, December 2003.

Johnston, et al. Best Current Practice [Page 115]

RFC 3666 SIP PSTN Call Flows December 2003

7. Acknowledgments

Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings,
and Tom Taylor for their detailed comments during the final review.
Thanks to Dean Willis for his early contributions to the development
of this document. Thanks to Jon Peterson for his help on the
security section.

The authors wish to thank Kundan Singh for performing parser
validation of messages.

The authors wish to thank the following individuals for their
participation in a detailed review of this call flows document: Aseem
Agarwal, Rafi Assadi, Ben Campbell, Sunitha Kumar, Jon Peterson, Marc
Petit-Huguenin, Vidhi Rastogi, and Bodgey Yin Shaohua.

The authors also wish to thank the following individuals for their
assistance: Jean-Francois Mule, Hemant Agrawal, Henry Sinnreich,
David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole
MCI WorldCom IPOP Design team, Scott Orton, Greg Osterhout, Pat
Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise
Ingram, Denise Caballero, Tom Redman, Ilya Slain, Pat Sollee, John
Truetken, and others from MCI WorldCom, 3Com, Cisco, Lucent and
Nortel.

8. Intellectual Property Statement

The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it
has made any effort to identify any such rights. Information on the
IETF's procedures with respect to rights in standards-track and
standards-related documentation can be found in BCP-11. Copies of
claims of rights made available for publication and any assurances of
licenses to be made available, or the result of an attempt made to
obtain a general license or permission for the use of such
proprietary rights by implementors or users of this specification can
be obtained from the IETF Secretariat.

The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice
this standard. Please address the information to the IETF Executive
Director.

Johnston, et al. Best Current Practice [Page 116]

RFC 3666 SIP PSTN Call Flows December 2003

9. Authors' Addresses

All listed authors actively contributed large amounts of text to this
document.

Alan Johnston
MCI
100 South 4th Street
St. Louis, MO 63102
USA

EMail: [email protected]

Steve Donovan
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA

EMail: [email protected]

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

EMail: [email protected]

Chris Cunningham
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA

EMail: [email protected]

Kevin Summers
Sonus
1701 North Collins Blvd, Suite 3000
Richardson, TX 75080
USA

EMail: [email protected]

Johnston, et al. Best Current Practice [Page 117]

RFC 3666 SIP PSTN Call Flows December 2003

10. Full Copyright Statement

Copyright (C) The Internet Society (2003). 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 assignees.

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

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

Johnston, et al. Best Current Practice [Page 118]