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.
Johnston, et al. Best Current Practice [Page 1]
RFC 3666 SIP PSTN Call Flows December 2003
Table of Contents
1. Overview..................................................... 21.1. General Assumptions.................................... 31.2. Legend for Message Flows............................... 41.3. SIP Protocol Assumptions............................... 52. SIP to PSTN Dialing.......................................... 62.1. Successful SIP to ISUP PSTN call....................... 72.2. Successful SIP to ISDN PBX call........................ 152.3. Successful SIP to ISUP PSTN call with overflow......... 232.4. Session established using ENUM Query................... 322.5. Unsuccessful SIP to PSTN call: Treatment from PSTN..... 38
2.6. Unsuccessful SIP to PSTN: REL w/Cause from PSTN........ 452.7. Unsuccessful SIP to PSTN: ANM Timeout.................. 493. PSTN to SIP Dialing.......................................... 543.1. Successful PSTN to SIP call............................ 553.2. Successful PSTN to SIP call, Fast Answer............... 623.3. Successful PBX to SIP call............................. 683.4. 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.................... 843.8. Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy... 88
3.9. Unsuccessful PSTN->SIP, Caller Abandonment............. 914. PSTN to PSTN Dialing via SIP Network......................... 964.1. Successful ISUP PSTN to ISUP PSTN call................. 974.2. Successful FGB PBX to ISDN PBX call with overflow...... 1055. Security Considerations...................................... 1136. References................................................... 1156.1. Normative References................................... 1156.2. Informative References................................. 1157. Acknowledgments.............................................. 1168. Intellectual Property Statement.............................. 1169. Authors' Addresses........................................... 11710. Full Copyright Statement..................................... 118
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].
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.
Johnston, et al. Best Current Practice [Page 3]
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).
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.
Johnston, et al. Best Current Practice [Page 4]
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. */
This document does not prescribe the flows precisely as they are
shown, but rather the flows illustrate the principles for best
practice. They are best practices usages (orderings, syntax,
selection of features for the purpose, handling of error) of SIP
methods, headers and parameters. IMPORTANT: The exact flows here
must not be copied as is by an implementer due to specific incorrect
characteristics that were introduced into the document for
convenience and are listed below. To sum up, the SIP/PSTN call flows
represent well-reviewed examples of SIP usage, which are best common
practice according to IETF consensus.
For simplicity in reading and editing the document, there are a
number of differences between some of the examples and actual SIP
messages. For example, the SIP Digest responses are not actual MD5
encodings. Call-IDs are often repeated, and CSeq counts often begin
at 1. Header fields are usually shown in the same order. Usually
only the minimum required header field set is shown, others that
would normally be present, such as Accept, Supported, Allow, etc. are
not shown.
Actors:
Element Display Name URI IP Address
------- ------------ --- ----------
User Agent Alice sip:alice@a.example.com 192.0.2.101
User Agent Bob sip:bob@b.example.com 192.0.2.200
Proxy Server sip:ss1.a.example.com 192.0.2.111
User Agent (Gateway) sip:gw1.a.example.com 192.0.2.201
User Agent (Gateway) sip:gw2.a.example.com 192.0.2.202
User Agent (Gateway) sip:gw3.a.example.com 192.0.2.203
User Agent (Gateway) sip:ngw1.a.example.com 192.0.2.103
User Agent (Gateway) sip:ngw2.a.example.com 192.0.2.102
Note that NGW 1 and NGW 2 also have device URIs (Contacts) of
sip:ngw1@a.example.com and sip:ngw2@a.example.com which resolve to
the Proxy Server sip:ss1.wcom.com using DNS SRV records.
Johnston, et al. Best Current Practice [Page 5]
RFC 3666 SIP PSTN Call Flows December 2003
In the following scenarios, Alice (sip:alice@a.example.com) is a SIP
phone or other SIP-enabled device. Bob is reachable via the PSTN at
global telephone number +19725552222. Alice places a call to Bob
through a Proxy Server, Proxy 1, and a Network Gateway. In other
scenarios, Alice places calls to Carol, who is served via a PBX
(Private Branch Exchange) and is identified by a private extension
444-3333, or global number +1-918-555-3333. Note that 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.
Johnston, et al. Best Current Practice [Page 6]
RFC 3666 SIP PSTN Call Flows December 2003
Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| Both Way RTP Media | One Way Voice |
|<===============================>|<===============|
| | | ANM F9 |
| | 200 F10 |<---------------|
| 200 F11 |<---------------| |
|<---------------| | |
| ACK F12 | | |
|--------------->| ACK F13 | |
| |--------------->| |
| Both Way RTP Media | Both Way Voice |
|<===============================>|<==============>|
| BYE F14 | | |
|--------------->| BYE F15 | |
| |--------------->| |
| | 200 F16 | |
| 200 F17 |<---------------| REL F18 |
|<---------------| |--------------->|
| | | RLC F19 |
| | |<---------------|
| | | |
Alice dials the globalized E.164 number +19725552222 to reach Bob.
Note that A might have only dialed the last 7 digits, or some other
dialing plan. It is assumed that the SIP User Agent Client converts
the digits into a global number and puts them into a SIP URI. Note
that tel URIs could be used instead of SIP URIs.
Alice could use either their SIP address (sip:alice@a.example.com) or
SIP telephone number (sip:+13145551111@ss1.a.example.com;user=phone)
in the From header. In this example, the telephone number is
included, and it is shown as being passed as calling party
identification through the Network Gateway (NGW 1) to Bob (F5). Note
Johnston, et al. Best Current Practice [Page 7]
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:ngw1@a.example.com. This is because NGW 1 only accepts
SIP messages that come through Proxy 1 - any direct signaling will be
ignored. Since this Contact URI may be used outside of this dialog
and must be routable (Section 8.1.1.8 in RFC 3261 [2]) the Contact
URI for NGW 1 must resolve to Proxy 1. This Contact URI resolves via
DNS to Proxy 1 (sip:ss1.a.example.com) which then resolves it to
sip:ngw1.a.example.com which is the address of NGW 1.
This flow shows TCP transport.
Message Details
F1 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Proxy-Authorization: Digest username="alice", realm="a.example.com",
nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
uri="sip:+19725552222@ss1.a.example.com;user=phone",
response="ccdca50cb091d587421457305d097458c"
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Johnston, et al. Best Current Practice [Page 8]
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:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine the gateway
for terminating this call. The call is forwarded to NGW 1. Client
for A prepares to receive data on port 49172 from the
network.*/
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Johnston, et al. Best Current Practice [Page 9]
RFC 3666 SIP PSTN Call Flows December 2003
;received=192.0.2.111
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> Bob
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM Bob -> NGW 1
ACM
F7 183 Session Progress NGW 1 -> Proxy 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* NGW 1 sends PSTN audio (ringing) in the RTP path to A */
Johnston, et al. Best Current Practice [Page 10]
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:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F9 ANM Bob -> NGW 1
ANM
F10 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Johnston, et al. Best Current Practice [Page 11]
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:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F12 ACK Alice -> Proxy 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Johnston, et al. Best Current Practice [Page 12]
RFC 3666 SIP PSTN Call Flows December 2003
Content-Length: 0
F13 ACK Proxy 1 -> NGW 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
/* Alice Hangs Up with Bob. */
F14 BYE Alice -> Proxy 1
BYE sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F15 BYE Proxy 1 -> NGW 1
BYE sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
Johnston, et al. Best Current Practice [Page 13]
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:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F17 200 OK Proxy 1 -> A
SIP/2.0 200 OK
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F18 REL NGW 1 -> B
REL
CauseCode=16 Normal
F19 RLC B -> NGW 1
RLC
Johnston, et al. Best Current Practice [Page 14]
RFC 3666 SIP PSTN Call Flows December 2003
Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | REL(1) F6 |
| | |<---------------|
| | | RLC F7 |
| | 404 F8 |--------------->|
| |<---------------| |
| | ACK F9 | |
| |--------------->| |
| 404 F10 | | |
|<---------------| | |
| ACK F11 | | |
|--------------->| | |
| | | |
Alice calls PSTN Bob through a Proxy Server Proxy 1 and a Network
Gateway NGW 1. The call is rejected by the PSTN with a ANSI ISUP
Release message REL containing a specific Cause code. This cause
value (1) is mapped by the Gateway to a SIP 404 Address Incomplete
response which is proxied back to Alice. For more details of ISUP
cause value to SIP response mapping, refer to [4].
Message Details
F1 INVITE Alice -> Proxy 1
INVITE sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",
opaque="", uri="sip:+44-1234@ss1.a.example.com;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:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW1.
Client for A prepares to receive data on port 49172 from the network.
*/
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 154
Johnston, et al. 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:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> Bob
IAM
CdPN=44-1234,NPI=E.164,NOA=International
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 REL Bob -> NGW 1
REL
CauseValue=1 Unallocated number
F7 RLC NGW 1 -> Bob
RLC
/* Network Gateway maps CauseValue=1 to the SIP message 404 Not
Found */
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:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Error-Info: <sip:not-found-ann@ann.a.example.com>
Content-Length: 0
F9 ACK Proxy 1 -> NGW 1
ACK sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
F10 404 Not Found Proxy 1 -> Alice
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Error-Info: <sip:not-found-ann@ann.a.example.com>
Content-Length: 0
F11 ACK Alice -> Proxy 1
ACK sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Johnston, et al. Best Current Practice [Page 48]
RFC 3666 SIP PSTN Call Flows December 2003
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
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.
Switch A GW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| |
| | 604 F3 | |
| |<---------------| |
| | ACK F4 | |
| |--------------->| |
| REL F5 | | |
|<---------------| | |
| RLC F6 | | |
|--------------->| | |
| | | |
Alice attempts to place a call through Gateway GW 1 and Proxy 1,
which is unable to find any routing for the number. The call is
rejected by Proxy 1 with a REL message containing a specific Cause
value mapped by the gateway based on the SIP error.
Message Details
F1 IAM Alice -> GW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-9999,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1
INVITE sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
Johnston, et al. Best Current Practice [Page 74]
RFC 3666 SIP PSTN Call Flows December 2003
To: <sip:+1972559999@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Contact:
<sip:+13145551111@gw1.a.example.com;user=phone;transport=tcp>
Content-Type: application/sdp
Content-Length: 144
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service to find a route to +1-972-555-
9999. A route is not found, so Proxy 1 rejects the call. */
F3 604 Does Not Exist Anywhere Proxy 1 -> GW 1
SIP/2.0 604 Does Not Exist Anywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Error-Info: <sip:does-not-exist@ann.a.example.com>
Content-Length: 0
F4 ACK GW 1 -> Proxy 1
ACK sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
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
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:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */
F3 INVITE Proxy 1 -> Bob
INVITE bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
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:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 600 Busy Everywhere Bob -> Proxy 1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F6 ACK Proxy 1 -> Bob
ACK bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
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:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F8 ACK NGW 1 -> Proxy 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F9 ACM NGW 1 -> Alice
ACM
/* A one way speech path is established between NGW 1 and Alice. */
/* Call Released after Alice hangs up. */
F10 REL Alice -> NGW 1
REL
CauseCode=16
F11 RLC NGW 1 -> Alice
RLC
Johnston, et al. Best Current Practice [Page 83]
RFC 3666 SIP PSTN Call Flows December 2003
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
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
[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.
[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
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.
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
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The IETF invites any interested party to bring to its attention any
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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
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: alan.johnston@mci.com
Steve Donovan
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA
EMail: sdonovan@dynamicsoft.com
Robert Sparks
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA
EMail: rsparks@dynamicsoft.com
Chris Cunningham
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA
EMail: ccunningham@dynamicsoft.com
Kevin Summers
Sonus
1701 North Collins Blvd, Suite 3000
Richardson, TX 75080
USA
EMail: kevin.summers@sonusnet.com
Johnston, et al. Best Current Practice [Page 117]
RFC 3666 SIP PSTN Call Flows December 2003
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Johnston, et al. Best Current Practice [Page 118]