draft-ietf-sipping-pstn-call-flows-02.txt   rfc3666.txt 
SIPPING Working Group A. Johnston Network Working Group A. Johnston
Internet Draft WorldCom Request for Comments: 3666 MCI
Document: draft-ietf-sipping-pstn-call-flows-02.txt S. Donovan BCP: 76 S. Donovan
Expires: October 2003 R. Sparks Category: Best Current Practice R. Sparks
C. Cunningham C. Cunningham
dynamicsoft dynamicsoft
K. Summers K. Summers
Sonus Sonus
April 2003 December 2003
Session Initiation Protocol PSTN Call Flows Session Initiation Protocol (SIP)
Public Switched Telephone Network (PSTN) Call Flows
Status of this Memo Status of this Memo
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Abstract Abstract
This document contains best current practice examples of Session This document contains best current practice examples of Session
Initiation Protocol (SIP) call flows showing interworking with the Initiation Protocol (SIP) call flows showing interworking with the
Public Switched Telephone Network (PSTN). Elements in these call Public Switched Telephone Network (PSTN). Elements in these call
flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways. 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. Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP.
PSTN telephony protocols are illustrated using ISDN (Integrated PSTN telephony protocols are illustrated using ISDN (Integrated
Services Digital Network), ISUP (ISDN User Part), and FGB (Feature Services Digital Network), ISUP (ISDN User Part), and FGB (Feature
Group B) circuit associated signaling. PSTN calls are illustrated Group B) circuit associated signaling. PSTN calls are illustrated
using global telephone numbers from the PSTN and private extensions using global telephone numbers from the PSTN and private extensions
served on by a PBX (Private Branch Exchange). Call flow diagrams and served on by a PBX (Private Branch Exchange). Call flow diagrams and
message details are shown. message details are shown.
SIP PSTN Call Flows April 2003
Conventions used in this document
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC-2119 [1].
Table of Contents Table of Contents
1. Overview.......................................................2 1. Overview..................................................... 2
1.1 General Assumptions........................................3 1.1. General Assumptions.................................... 3
1.2 Legend for Message Flows...................................4 1.2. Legend for Message Flows............................... 4
1.3 SIP Protocol Assumptions...................................5 1.3. SIP Protocol Assumptions............................... 5
2. SIP to PSTN Dialing............................................6 2. SIP to PSTN Dialing.......................................... 6
2.1 Successful SIP to ISUP PSTN call...........................7 2.1. Successful SIP to ISUP PSTN call....................... 7
2.2 Successful SIP to ISDN PBX call...........................15 2.2. Successful SIP to ISDN PBX call........................ 15
2.3 Successful SIP to ISUP PSTN call with overflow............23 2.3. Successful SIP to ISUP PSTN call with overflow......... 23
2.4 Session established using ENUM Query......................32 2.4. Session established using ENUM Query................... 32
2.5 Unsuccessful SIP to PSTN call: Treatment from PSTN........38 2.5. Unsuccessful SIP to PSTN call: Treatment from PSTN..... 38
2.6 Unsuccessful SIP to PSTN: REL w/Cause from PSTN...........45 2.6. Unsuccessful SIP to PSTN: REL w/Cause from PSTN........ 45
2.7 Unsuccessful SIP to PSTN: ANM Timeout.....................50 2.7. Unsuccessful SIP to PSTN: ANM Timeout.................. 49
3. PSTN to SIP Dialing...........................................56 3. PSTN to SIP Dialing.......................................... 54
3.1 Successful PSTN to SIP call...............................57 3.1. Successful PSTN to SIP call............................ 55
3.2 Successful PSTN to SIP call, Fast Answer..................64 3.2. Successful PSTN to SIP call, Fast Answer............... 62
3.3 Successful PBX to SIP call................................70 3.3. Successful PBX to SIP call............................. 68
3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL.....77 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......79 3.5. Unsuccessful PSTN to SIP REL, SIP busy mapped to REL... 76
3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones...83 3.6. Unsuccessful PSTN->SIP, SIP error interworking to tones 80
3.7 Unsuccessful PSTN->SIP, ACM timeout.......................87 3.7. Unsuccessful PSTN->SIP, ACM timeout.................... 84
3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy......91 3.8. Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy... 88
3.9 Unsuccessful PSTN->SIP, Caller Abandonment................95 3.9. Unsuccessful PSTN->SIP, Caller Abandonment............. 91
4. PSTN to PSTN Dialing via SIP Network.........................101 4. PSTN to PSTN Dialing via SIP Network......................... 96
4.1 Successful ISUP PSTN to ISUP PSTN call...................102 4.1. Successful ISUP PSTN to ISUP PSTN call................. 97
4.2 Successful FGB PBX to ISDN PBX call with overflow........110 4.2. Successful FGB PBX to ISDN PBX call with overflow...... 105
Security Considerations.........................................118 5. Security Considerations...................................... 113
Normative References............................................120 6. References................................................... 115
Informative References..........................................120 6.1. Normative References................................... 115
Acknowledgments.................................................121 6.2. Informative References................................. 115
Author's Addresses..............................................121 7. Acknowledgments.............................................. 116
8. Intellectual Property Statement.............................. 116
9. Authors' Addresses........................................... 117
10. Full Copyright Statement..................................... 118
1. Overview 1. Overview
The call flows shown in this document were developed in the design of The call flows shown in this document were developed in the design of
a SIP IP communications network. They represent an example a SIP IP communications network. They represent an example of a
minimum set of functionality. minimum set of functionality.
It is the hope of the authors that this document will be useful for It is the hope of the authors that this document will be useful for
SIP implementers, designers, and protocol researchers alike and will SIP implementers, designers, and protocol researchers alike and will
help further the goal of a standard implementation of RFC 3261 [2]. help further the goal of a standard implementation of RFC 3261 [2].
These flows represent carefully checked and working group reviewed These flows represent carefully checked and working group reviewed
SIP PSTN Call Flows April 2003
scenarios of the most common SIP/PSTN interworking examples as a scenarios of the most common SIP/PSTN interworking examples as a
companion to the specifications. companion to the specifications.
These call flows are based on the current version 2.0 of SIP in These call flows are based on the current version 2.0 of SIP in RFC
RFC 3261 [2] with SDP usage described in RFC 3264 [3]. Other RFCs 3261 [2] with SDP usage described in RFC 3264 [3]. Other RFCs also
also comprise the SIP standard but are not used in this set of basic comprise the SIP standard but are not used in this set of basic call
call flows. The SIP/ISUP mapping is based on RFC zzzz [4]. flows. The SIP/ISUP mapping is based on RFC 3398 [4].
Various PSTN signaling protocols are illustrated in this document: Various PSTN signaling protocols are illustrated in this document:
ISDN (Integrated Services Digital Network), ISUP (ISDN User ISDN (Integrated Services Digital Network), ISUP (ISDN User Part) and
Part) and FGB (Feature Group B) circuit associated signaling. This FGB (Feature Group B) circuit associated signaling. This document
document shows mainly ANSI ISUP due to its practical origins. shows mainly ANSI ISUP due to its practical origins. However, as
However, as used in this document, the usage is virtually identical used in this document, the usage is virtually identical to the ITU-T
to the ITU-T International ISUP used as the reference in [4]. International ISUP used as the reference in [4].
Basic SIP call flow examples are contained in a companion document, Basic SIP call flow examples are contained in a companion document,
RFC yyyy [11]. RFC 3665 [10].
1.1 General Assumptions 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 A number of architecture, network, and protocol assumptions underlie
the call flows in this document. Note that these assumptions are not the call flows in this document. Note that these assumptions are not
requirements. They are outlined in this section so that they may be requirements. They are outlined in this section so that they may be
taken into consideration and to aid in the understanding of the call taken into consideration and to aid in the understanding of the call
flow examples. flow examples.
The authentication of SIP User Agents in these example call flows is The authentication of SIP User Agents in these example call flows is
performed using SIP Digest as defined in [3] and [5]. performed using HTTP Digest as defined in [3] and [5].
Some Proxy Servers in these call flows insert Record-Route headers Some Proxy Servers in these call flows insert Record-Route headers
into requests to ensure that they are in the signaling path for into requests to ensure that they are in the signaling path for
future message exchanges. future message exchanges.
These flows show TLS, TCP, and UDP for transport. SCTP [6] could These flows show TLS, TCP, and UDP for transport. SCTP could also be
also be used. See the discussion in RFC 3261 [2] for details on the used. See the discussion in RFC 3261 [2] for details on the
transport issues for SIP. transport issues for SIP.
The SIP Proxy Server has access to a Location Service and other The SIP Proxy Server has access to a Location Service and other
databases. Information present in the Request-URI and the context databases. Information present in the Request-URI and the context
(From header) is sufficient to determine to which proxy or gateway (From header) is sufficient to determine to which proxy or gateway
the message should be routed. In most cases, a primary and secondary the message should be routed. In most cases, a primary and secondary
route will be determined in case of Proxy or Gateway failure route will be determined in case of a Proxy or Gateway failure
downstream. downstream.
Gateways provide tones (ringing, busy, etc) and announcements to the Gateways provide tones (ringing, busy, etc) and announcements to the
PSTN side based on SIP response messages, or pass along audio in-band 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 tones (ringing, busy tone, etc.) in an early media stream to the SIP
side. side.
SIP PSTN Call Flows April 2003
The interactions between the Proxy and Gateway can be summarized as The interactions between the Proxy and Gateway can be summarized as
follows: follows:
. The SIP Proxy Server performs digit analysis and lookup and - The SIP Proxy Server performs digit analysis and lookup and
locates the correct gateway. locates the correct gateway.
. The SIP Proxy Server performs gateway location based on primary - The SIP Proxy Server performs gateway location based on primary
and secondary routing. and secondary routing.
Telephone numbers are usually represented as SIP URIs. Note that an Telephone numbers are usually represented as SIP URIs. Note that an
alternative is the use of the tel URI [7]. alternative is the use of the tel URI [6].
This document shows typical examples of SIP/ISUP interworking. This document shows typical examples of SIP/ISUP interworking.
Although in the spirit of the SIP-T framework [8], these examples do Although in the spirit of the SIP-T framework [7], these examples do
not represent a complete implementation of the framework. The not represent a complete implementation of the framework. The
examples here represent more of a minimal set of examples for very examples here represent more of a minimal set of examples for very
basic SIP to ISUP interworking, rather than the more complex goal of basic SIP to ISUP interworking, rather than the more complex goal of
ISUP transparency. In particular, there are NO examples of ISUP transparency. In particular, there are NO examples of
encapsulated ISUP in this document. If present, these messages would encapsulated ISUP in this document. If present, these messages would
show S/MIME encryption due to the sensitive nature of this show S/MIME encryption due to the sensitive nature of this
information, as discussed in the SIP-T Framework security information, as discussed in the SIP-T Framework security
considerations section. (Note - RFC 3204 [9] contains an example of considerations section. (Note - RFC 3204 [8] contains an example of
an INVITE with encapsulated ISUP.) See the Security Considerations an INVITE with encapsulated ISUP.) See the Security Considerations
section for a more detailed discussion on the security of these call section for a more detailed discussion on the security of these call
flows. flows.
In ISUP, the Calling Party Number is abbreviated as CgPN and the In ISUP, the Calling Party Number is abbreviated as CgPN and the
Called Party Number is abbreviated as CdPN. Other abbreviations Called Party Number is abbreviated as CdPN. Other abbreviations
include Numbering Plan Indicator (NPI) and Nature of Address (NOA). include Numbering Plan Indicator (NPI) and Nature of Address (NOA).
1.2 Legend for Message Flows 1.2. Legend for Message Flows
Dashed lines (---) represent signaling messages that are mandatory to Dashed lines (---) represent signaling messages that are mandatory to
the call scenario. These messages can be SIP or PSTN the call scenario. These messages can be SIP or PSTN signaling. The
signaling. The arrow indicates the direction of message flow. arrow indicates the direction of message flow.
Double dashed lines (===) represent media paths between network Double dashed lines (===) represent media paths between network
elements. elements.
Messages with parentheses around their name represent optional Messages with parentheses around their name represent optional
messages. messages.
Messages are identified in the Figures as F1, F2, etc. This Messages are identified in the Figures as F1, F2, etc. This
references the message details in the list that follows the Figure. references the message details in the list that follows the Figure.
Comments in the message details are shown in the following form: Comments in the message details are shown in the following form:
/* Comments. */ /* Comments. */
SIP PSTN Call Flows April 2003
1.3 SIP Protocol Assumptions 1.3. SIP Protocol Assumptions
This document does not prescribe the flows precisely as they are This document does not prescribe the flows precisely as they are
shown, but rather the flows illustrate the principles for best shown, but rather the flows illustrate the principles for best
practice. They are best practices usages (orderings, syntax, practice. They are best practices usages (orderings, syntax,
selection of features for the purpose, handling of error) of SIP selection of features for the purpose, handling of error) of SIP
methods, headers and parameters. IMPORTANT: The exact flows here methods, headers and parameters. IMPORTANT: The exact flows here
must not be copied as is by an implementer due to specific incorrect must not be copied as is by an implementer due to specific incorrect
characteristics that were introduced into the document for characteristics that were introduced into the document for
convenience and are listed below. To sum up, the SIP/PSTN call flows convenience and are listed below. To sum up, the SIP/PSTN call flows
represent well-reviewed examples of SIP usage, which are best common represent well-reviewed examples of SIP usage, which are best common
practice according to IETF consensus. practice according to IETF consensus.
For simplicity in reading and editing the document, there are a For simplicity in reading and editing the document, there are a
number of differences between some of the examples and actual SIP number of differences between some of the examples and actual SIP
messages. For example, the SIP Digest responses are not actual MD5 messages. For example, the SIP Digest responses are not actual MD5
encodings. Call-IDs are often repeated, and CSeq counts often begin encodings. Call-IDs are often repeated, and CSeq counts often begin
at 1. Header fields are usually shown in the same order. Usually at 1. Header fields are usually shown in the same order. Usually
only the minimum required header field set is shown, others that only the minimum required header field set is shown, others that
would normally be present such as Accept, Supported, Allow, etc are would normally be present, such as Accept, Supported, Allow, etc. are
not shown. not shown.
Actors: Actors:
Element Display Name URI IP Address Element Display Name URI IP Address
------- ------------ --- ---------- ------- ------------ --- ----------
User Agent Alice sip:alice@a.example.com 192.0.2.101 User Agent Alice sip:alice@a.example.com 192.0.2.101
User Agent Bob sip:bob@b.example.com 192.0.2.200 User Agent Bob sip:bob@b.example.com 192.0.2.200
Proxy Server sip:ss1.a.example.com 192.0.2.111 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:gw1.a.example.com 192.0.2.201
User Agent (Gateway) sip:gw2.a.example.com 192.0.2.202 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:gw3.a.example.com 192.0.2.203
User Agent (Gateway) sip:ngw1.a.example.com 192.0.2.103 User Agent (Gateway) sip:ngw1.a.example.com 192.0.2.103
User Agent (Gateway) sip:ngw2.a.example.com 192.0.2.102 User Agent (Gateway) sip:ngw2.a.example.com 192.0.2.102
Note that NGW 1 and NGW 2 also have a device URIs (Contacts) of 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 resolves to 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. the Proxy Server sip:ss1.wcom.com using DNS SRV records.
SIP PSTN Call Flows April 2003
2. SIP to PSTN Dialing 2. SIP to PSTN Dialing
In the following scenarios, Alice (Alice sip:alice@a.example.com) is In the following scenarios, Alice (sip:alice@a.example.com) is a SIP
a SIP phone or other SIP-enabled device. Bob is reachable via the phone or other SIP-enabled device. Bob is reachable via the PSTN at
PSTN at global telephone number +19725552222. Alice places a call global telephone number +19725552222. Alice places a call to Bob
to Bob through a Proxy Server Proxy 1 and a Network Gateway. In through a Proxy Server, Proxy 1, and a Network Gateway. In other
other scenarios, Alice places calls to Carol, who is served via a scenarios, Alice places calls to Carol, who is served via a PBX
PBX (Private Branch Exchange) and is identified by a private (Private Branch Exchange) and is identified by a private extension
extension 444-3333, or global number +1-918-555-3333. Note that User 444-3333, or global number +1-918-555-3333. Note that Alice uses
A uses his/her global telephone number +1-314-555-1111 in the From his/her global telephone number +1-314-555-1111 in the From header in
header in the INVITE messages. This then gives the Gateway the the INVITE messages. This then gives the Gateway the option of using
option of using this header to populate the calling party this header to populate the calling party identification field in
identification field in subsequent signaling. Left open is the issue subsequent signaling. Left open is the issue of how the Gateway can
of how the Gateway can determine the accuracy of the telephone determine the accuracy of the telephone number which is necessary
number, necessary before passing it as a valid calling party number before passing it as a valid calling party number in the PSTN.
in the PSTN.
In these scenarios, Alice is a SIP phone or other SIP-enabled In these scenarios, Alice is a SIP phone or other SIP-enabled device.
device. Alice places a call to Bob in the PSTN or Carol on a Alice places a call to Bob in the PSTN or Carol on a PBX through a
PBX through a Proxy Server and a Gateway. Proxy Server and a Gateway.
In the failure scenarios, the call does not complete. In some In the failure scenarios, the call does not complete. In some cases
cases, however, a media stream is still setup. This is due to the however, a media stream is still setup. This is due to the fact that
fact that some failures in dialing to the PSTN result in in-band some failures in dialing to the PSTN result in in-band tones (busy,
tones (busy, reorder tones or announcements - "The number you have reorder tones or announcements - "The number you have dialed has
dialed has changed. The new number is..."). The 183 Session changed. The new number is..."). The 183 Session Progress response
Progress response containing SDP media information is used to containing SDP media information is used to setup this early media
setup this early media path so that the caller Alice knows the final path so that the caller Alice knows the final disposition of the
disposition of the call. call.
The media stream is either terminated by the caller after the tone or 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 announcement has been heard and understood, or by the Gateway after a
timer expires. timer expires.
In other failure scenarios, a SS7 Release with Cause Code is mapped 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 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 used, but the actual failure code is conveyed to the caller by the
SIP User Agent Client. SIP User Agent Client.
SIP PSTN Call Flows April 2003 2.1. Successful SIP to ISUP PSTN call
2.1 Successful SIP to ISUP PSTN call
Alice Proxy 1 NGW 1 Switch B Alice Proxy 1 NGW 1 Switch B
| | | | | | | |
| INVITE F1 | | | | INVITE F1 | | |
|--------------->| | | |--------------->| | |
| 100 F2 | | | | 100 F2 | | |
|<---------------| INVITE F3 | | |<---------------| INVITE F3 | |
| |--------------->| | | |--------------->| |
| | 100 F4 | | | | 100 F4 | |
| |<---------------| IAM F5 | | |<---------------| IAM F5 |
skipping to change at page 7, line 44 skipping to change at page 7, line 42
| BYE F14 | | | | BYE F14 | | |
|--------------->| BYE F15 | | |--------------->| BYE F15 | |
| |--------------->| | | |--------------->| |
| | 200 F16 | | | | 200 F16 | |
| 200 F17 |<---------------| REL F18 | | 200 F17 |<---------------| REL F18 |
|<---------------| |--------------->| |<---------------| |--------------->|
| | | RLC F19 | | | | RLC F19 |
| | |<---------------| | | |<---------------|
| | | | | | | |
Alice dials the globalized E.164 number +19725552222 to reach Alice dials the globalized E.164 number +19725552222 to reach Bob.
Bob. Note that A might have only dialed the last 7 digits, or Note that A might have only dialed the last 7 digits, or some other
some other dialing plan. It is assumed that the SIP User Agent dialing plan. It is assumed that the SIP User Agent Client converts
Client converts the digits into a global number and puts them into a the digits into a global number and puts them into a SIP URI. Note
SIP URI. Note that tel URIs could be used instead of SIP URIs. that tel URIs could be used instead of SIP URIs.
Alice could use either their SIP address (sip:alice@a.example.com) or Alice could use either their SIP address (sip:alice@a.example.com) or
SIP telephone number (sip:+13145551111@ss1.a.example.com;user=phone) SIP telephone number (sip:+13145551111@ss1.a.example.com;user=phone)
in the From header. In this example, the telephone number is in the From header. In this example, the telephone number is
included, and it is shown as being passed as calling party included, and it is shown as being passed as calling party
SIP PSTN Call Flows April 2003
identification through the Network Gateway (NGW 1) to Bob (F5). Note identification through the Network Gateway (NGW 1) to Bob (F5). Note
that for this number to be passed into the SS7 network, it would have that for this number to be passed into the SS7 network, it would have
to be somehow verified for accuracy. to be somehow verified for accuracy.
In this scenario, Bob answers the call then Alice disconnects the In this scenario, Bob answers the call, then Alice disconnects the
call. Signaling between NGW 1 and Bob's telephone switch is ANSI call. Signaling between NGW 1 and Bob's telephone switch is ANSI
ISUP. For the details of SIP to ISUP mapping, refer to [4]. 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 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 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 SIP messages that come through Proxy 1 - any direct signaling will be
ignored. Since this Contact URI may be used outside of this dialog 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 and must be routable (Section 8.1.1.8 in RFC 3261 [2]) the Contact
URI for NGW 1 must resolve to Proxy 1. This Contact URI is an AOR URI for NGW 1 must resolve to Proxy 1. This Contact URI resolves via
which resolves via DNS to Proxy 1 (sip:ss1.a.example.com) which then DNS to Proxy 1 (sip:ss1.a.example.com) which then resolves it to
resolves it to sip:ngw1.a.example.com which is the address of NGW 1. sip:ngw1.a.example.com which is the address of NGW 1.
This flow shows TCP transport. This flow shows TCP transport.
Message Details Message Details
F1 INVITE Alice -> Proxy 1 F1 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
skipping to change at page 9, line 4 skipping to change at page 9, line 4
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
SIP PSTN Call Flows April 2003
F2 100 Trying Proxy 1 -> Alice F2 100 Trying Proxy 1 -> Alice
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
skipping to change at page 10, line 4 skipping to change at page 10, line 4
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1 F4 100 Trying NGW 1 -> Proxy 1
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
SIP PSTN Call Flows April 2003
;received=192.0.2.111 ;received=192.0.2.111
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F5 IAM NGW 1 -> Bob F5 IAM NGW 1 -> Bob
skipping to change at page 11, line 4 skipping to change at page 11, line 4
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
/* NGW 1 sends PSTN audio (ringing) in the RTP path to A */ /* NGW 1 sends PSTN audio (ringing) in the RTP path to A */
SIP PSTN Call Flows April 2003
F8 183 Session Progress Proxy 1 -> Alice F8 183 Session Progress Proxy 1 -> Alice
SIP/2.0 183 Session Progress SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
skipping to change at page 12, line 4 skipping to change at page 12, line 5
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
SIP PSTN Call Flows April 2003
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 gw1.a.example.com c=IN IP4 gw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F11 200 OK Proxy 1 -> Alice F11 200 OK Proxy 1 -> Alice
skipping to change at page 13, line 4 skipping to change at page 13, line 5
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr> Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
F13 ACK Proxy 1 -> NGW 1 F13 ACK Proxy 1 -> NGW 1
ACK sip:ngw1@a.example.com SIP/2.0 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 ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
skipping to change at page 14, line 4 skipping to change at page 14, line 6
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE CSeq: 2 BYE
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
F16 200 OK NGW 1 -> Proxy 1 F16 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
skipping to change at page 15, line 4 skipping to change at page 15, line 4
Content-Length: 0 Content-Length: 0
F18 REL NGW 1 -> B F18 REL NGW 1 -> B
REL REL
CauseCode=16 Normal CauseCode=16 Normal
F19 RLC B -> NGW 1 F19 RLC B -> NGW 1
RLC RLC
SIP PSTN Call Flows April 2003
2.2 Successful SIP to ISDN PBX call 2.2. Successful SIP to ISDN PBX call
Alice Proxy 1 GW 1 PBX C Alice Proxy 1 GW 1 PBX C
| | | | | | | |
| INVITE F1 | | | | INVITE F1 | | |
|--------------->| | | |--------------->| | |
| 100 F2 | | | | 100 F2 | | |
|<---------------| INVITE F3 | | |<---------------| INVITE F3 | |
| |--------------->| | | |--------------->| |
| | 100 F4 | | | | 100 F4 | |
| |<---------------| SETUP F5 | | |<---------------| SETUP F5 |
skipping to change at page 15, line 49 skipping to change at page 15, line 48
| |--------------->| | | |--------------->| |
| | 200 F18 | | | | 200 F18 | |
| 200 F19 |<---------------| DISConnect F20 | | 200 F19 |<---------------| DISConnect F20 |
|<---------------| |--------------->| |<---------------| |--------------->|
| | | RELease F21 | | | | RELease F21 |
| | |<---------------| | | |<---------------|
| | | RELease COM F22| | | | RELease COM F22|
| | |--------------->| | | |--------------->|
| | | | | | | |
Alice is a SIP device while Carol is connected via a Alice is a SIP device while Carol is connected via a Gateway (GW 1)
Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk to a PBX. The PBX connection is via a ISDN trunk group. Alice dials
group. Alice dials Carol's telephone number (918-555-3333) which Carol's telephone number (918-555-3333) which is globalized and put
is globalized and put into a SIP URI. into a SIP URI.
The host portion of the Request-URI in the INVITE F3 is used to The host portion of the Request-URI in the INVITE F3 is used to
SIP PSTN Call Flows April 2003
identify the context (customer, trunk group, or line) in which the identify the context (customer, trunk group, or line) in which the
private number 444-3333 is valid. Otherwise, this INVITE message private number 444-3333 is valid. Otherwise, this INVITE message
could get forwarded by GW 1 and the context of the digits could could get forwarded by GW 1 and the context of the digits could
become lost and the call unroutable. become lost and the call unroutable.
Proxy 1 looks up the telephone number and locates the gateway that Proxy 1 looks up the telephone number and locates the gateway that
serves Carol. Carolis identified by its extension serves Carol. Carol is identified by its extension (444-3333) in the
(444-3333) in the Request-URI sent to GW 1. Request-URI sent to GW 1.
Note that the Contact URI for GW1 as used in messages F8, F9, F12, Note that the Contact URI for GW 1, as used in messages F8, F9, F12,
and F13 is sips:4443333@gw1.a.example.com which does resolve directly and F13, is sips:4443333@gw1.a.example.com, which resolves directly
to the gateway. to the gateway.
This flow shows the use of Secure SIP (sips) URIs. This flow shows the use of Secure SIP (sips) URIs.
Message Details Message Details
F1 INVITE Alice -> Proxy 1 F1 INVITE Alice -> Proxy 1
INVITE sips:+19185553333@ss1.a.example.com;user=phone SIP/2.0 INVITE sips:+19185553333@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
skipping to change at page 17, line 4 skipping to change at page 17, line 4
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> Alice F2 100 Trying Proxy 1 -> Alice
SIP/2.0 100 Trying SIP/2.0 100 Trying
SIP PSTN Call Flows April 2003
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Content-Length: 0 Content-Length: 0
F3 INVITE Proxy 1 -> GW 1 F3 INVITE Proxy 1 -> GW 1
skipping to change at page 18, line 4 skipping to change at page 18, line 4
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
F5 SETUP GW 1 -> Carol F5 SETUP GW 1 -> Carol
Protocol discriminator=Q.931 Protocol discriminator=Q.931
Message type=SETUP Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16 Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio) (3.1 kHz audio)
Channel identification=Preferred or exclusive B-channel Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN;further call Progress indicator=1 (Call is not end-to-end ISDN;further call
progress information may be available inband) progress information may be available inband)
Called party number: Called party number:
skipping to change at page 18, line 48 skipping to change at page 19, line 4
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr> Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com> Contact: <sips:4443333@gw1.a.example.com>
Content-Length: 0 Content-Length: 0
F9 180 Ringing Proxy 1 -> Alice F9 180 Ringing Proxy 1 -> Alice
SIP PSTN Call Flows April 2003
SIP/2.0 180 Ringing SIP/2.0 180 Ringing
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr> Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
skipping to change at page 20, line 4 skipping to change at page 20, line 7
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com> Contact: <sips:4443333@gw1.a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 144 Content-Length: 144
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=- s=-
c=IN IP4 gw1.a.example.com c=IN IP4 gw1.a.example.com
t=0 0 t=0 0
SIP PSTN Call Flows April 2003
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F13 200 OK Proxy 1 -> Alice F13 200 OK Proxy 1 -> Alice
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr> Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
skipping to change at page 20, line 46 skipping to change at page 21, line 4
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr> Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK CSeq: 2 ACK
Content-Length: 0 Content-Length: 0
F15 ACK Proxy 1 -> GW 1 F15 ACK Proxy 1 -> GW 1
ACK sips:4443333@gw1.a.example.com SIP/2.0 ACK sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
SIP PSTN Call Flows April 2003
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK CSeq: 2 ACK
Content-Length: 0 Content-Length: 0
skipping to change at page 21, line 46 skipping to change at page 22, line 4
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE CSeq: 3 BYE
Content-Length: 0 Content-Length: 0
F18 200 OK GW 1 -> Proxy 1 F18 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
SIP PSTN Call Flows April 2003
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9 Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone> From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone> To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE CSeq: 3 BYE
Content-Length: 0 Content-Length: 0
skipping to change at page 23, line 4 skipping to change at page 23, line 4
F21 RELease Carol-> GW 1 F21 RELease Carol-> GW 1
Protocol discriminator=Q.931 Protocol discriminator=Q.931
Message type=REL Message type=REL
F22 RELease COMplete GW 1 -> Carol F22 RELease COMplete GW 1 -> Carol
Protocol discriminator=Q.931 Protocol discriminator=Q.931
Message type=REL COM Message type=REL COM
SIP PSTN Call Flows April 2003
2.3 Successful SIP to ISUP PSTN call with overflow 2.3. Successful SIP to ISUP PSTN call with overflow
Alice Proxy 1 NGW 1 NGW 2 Switch B Alice Proxy 1 NGW 1 NGW 2 Switch B
| | | | | | | | | |
| INVITE F1 | | | | | INVITE F1 | | | |
|------------->| | | | |------------->| | | |
| | INVITE F2 | | | | | INVITE F2 | | |
| 100 F3 |------------->| | | | 100 F3 |------------->| | |
|<-------------| 503 F4 | | | |<-------------| 503 F4 | | |
| |<-------------| | | | |<-------------| | |
| | ACK F5 | | | | | ACK F5 | | |
skipping to change at page 23, line 46 skipping to change at page 23, line 45
| BYE F16 | | | | BYE F16 | | |
|------------->| BYE F17 | | |------------->| BYE F17 | |
| |---------------------------->| | | |---------------------------->| |
| | 200 F18 | | | | 200 F18 | |
| 200 F19 |<----------------------------| REL F20 | | 200 F19 |<----------------------------| REL F20 |
|<-------------| |------------->| |<-------------| |------------->|
| | | RLC F21 | | | | RLC F21 |
| | |<-------------| | | |<-------------|
| | | | | | | |
Alice calls Bob through Proxy 1. Proxy 1 tries to route to a Alice calls Bob through Proxy 1. Proxy 1 tries to route to a Network
Network Gateway NGW 1. NGW 1 is not available and responds with a 503 Gateway NGW 1. NGW 1 is not available and responds with a 503
Service Unavailable (F4). The call is then routed to Network Gateway Service Unavailable (F4). The call is then routed to Network Gateway
NGW 2. Bob answers the call. The call is terminated when Alice NGW 2. Bob answers the call. The call is terminated when Alice
disconnects the call. NGW 2 and Bob's telephone switch use ANSI disconnects the call. NGW 2 and Bob's telephone switch use ANSI ISUP
ISUP signaling. signaling.
NGW 2 also only accepts SIP messages that come through Proxy 1, so NGW 2 also only accepts SIP messages that come through Proxy 1, so
the Contact URI sip:ngw2@a.example.com is used in this flow. the Contact URI sip:ngw2@a.example.com is used in this flow.
SIP PSTN Call Flows April 2003
This flow shows UDP transport. This flow shows UDP transport.
Message Details Message Details
F1 INVITE Alice -> Proxy 1 F1 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
skipping to change at page 25, line 4 skipping to change at page 25, line 5
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
SIP PSTN Call Flows April 2003
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com> Contact: <sip:alice@client.a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
skipping to change at page 25, line 49 skipping to change at page 26, line 4
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=123456789 ;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F5 ACK Proxy 1 -> NGW 1 F5 ACK Proxy 1 -> NGW 1
SIP PSTN Call Flows April 2003
ACK sip:ngw1@a.example.com SIP/2.0 ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com>;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com>;user=phone>
;tag=123456789 ;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK CSeq: 1 ACK
skipping to change at page 26, line 49 skipping to change at page 27, line 4
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F7 IAM NGW 2 -> Bob F7 IAM NGW 2 -> Bob
IAM IAM
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
F8 ACM Bob -> NGW 2 F8 ACM Bob -> NGW 2
SIP PSTN Call Flows April 2003
ACM ACM
F9 183 Session Progress NGW 2 -> Proxy 1 F9 183 Session Progress NGW 2 -> Proxy 1
SIP/2.0 183 Session Progress SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
skipping to change at page 28, line 4 skipping to change at page 28, line 7
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw2@a.example.com> Contact: <sip:ngw2@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
SIP PSTN Call Flows April 2003
o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com
s=- s=-
c=IN IP4 ngw2.a.example.com c=IN IP4 ngw2.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F11 ANM Bob -> NGW 2 F11 ANM Bob -> NGW 2
ANM ANM
skipping to change at page 29, line 4 skipping to change at page 29, line 8
F13 200 OK Proxy 1 -> Alice F13 200 OK Proxy 1 -> Alice
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
SIP PSTN Call Flows April 2003
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw2@a.example.com> Contact: <sip:ngw2@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com
s=- s=-
skipping to change at page 30, line 4 skipping to change at page 30, line 7
Max-Forwards: 69 Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
/* RTP streams are established between A and B(via the GW) */ /* RTP streams are established between A and B(via the GW) */
SIP PSTN Call Flows April 2003
/* Alice Hangs Up with Bob. */ /* Alice Hangs Up with Bob. */
F16 BYE Alice -> Proxy 1 F16 BYE Alice -> Proxy 1
BYE sip:ngw2@a.example.com SIP/2.0 BYE sip:ngw2@a.example.com SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
Route: <ss1.a.example.com;lr> Route: <ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
skipping to change at page 31, line 4 skipping to change at page 31, line 8
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE CSeq: 2 BYE
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
F19 200 OK Proxy 1 -> Alice F19 200 OK Proxy 1 -> Alice
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
skipping to change at page 32, line 4 skipping to change at page 32, line 4
Content-Length: 0 Content-Length: 0
F20 REL NGW 2 -> B F20 REL NGW 2 -> B
REL REL
CauseCode=16 Normal CauseCode=16 Normal
F21 RLC B -> NGW 2 F21 RLC B -> NGW 2
RLC RLC
SIP PSTN Call Flows April 2003
2.4 Successful SIP to SIP using ENUM Query 2.4. Successful SIP to SIP using ENUM Query
Alice DNS Server Proxy 3 Bob Alice DNS Server Proxy 3 Bob
| | | | | | | |
| ENUM Query F1 | | | | ENUM Query F1 | | |
|--------------->| | | |--------------->| | |
| Response F2 | | | | Response F2 | | |
|<---------------| | | |<---------------| | |
| INVITE F3 | | | INVITE F3 | |
|-------------------------------->| INVITE F4 | |-------------------------------->| INVITE F4 |
| 100 F5 |--------------->| | 100 F5 |--------------->|
skipping to change at page 32, line 38 skipping to change at page 32, line 37
| | BYE F12 | | | BYE F12 |
| BYE F13 |<---------------| | BYE F13 |<---------------|
|<--------------------------------| | |<--------------------------------| |
| 200 F14 | | | 200 F14 | |
|-------------------------------->| 200 F15 | |-------------------------------->| 200 F15 |
| |--------------->| | |--------------->|
| | | | | |
In this scenario, Alice places a call to Bob by dialing Bob's In this scenario, Alice places a call to Bob by dialing Bob's
telephone number (9725552222). Alice's UA converts the phone number telephone number (9725552222). Alice's UA converts the phone number
to an E.164 number (+19725552222) performs an ENUM query [10] on the to an E.164 number (+19725552222), and performs an ENUM query [9] on
E.164 number (2.2.2.2.5.5.5.2.7.9.1.e164.arpa) which returns a NAPTR the E.164 number (2.2.2.2.5.5.5.2.7.9.1.e164.arpa), which returns a
record containing a SIP AOR URI for Bob NAPTR record containing a SIP AOR URI for Bob
(sip:+19725552222@b.example.com). As a result, Alice's UA sends an (sip:+19725552222@b.example.com). As a result, Alice's UA sends an
INVITE and the call completes over IP bypassing the PSTN. INVITE and the call completes over IP bypassing the PSTN.
The call is terminated when Bob sends a BYE message. The call is terminated when Bob sends a BYE message.
Message Details Message Details
F1 ENUM Query Alice -> DNS Server F1 ENUM Query Alice -> DNS Server
2.2.2.2.5.5.5.2.7.9.1.e164.arpa 2.2.2.2.5.5.5.2.7.9.1.e164.arpa
SIP PSTN Call Flows April 2003
F2 ENUM NAPTR Set DNS Server -> Alice F2 ENUM NAPTR Set DNS Server -> Alice
$ORIGIN 2.2.2.2.5.5.5.2.7.9.1.e164.arpa. $ORIGIN 2.2.2.2.5.5.5.2.7.9.1.e164.arpa.
IN NAPTR 100 10 "u" "sip+E2U" IN NAPTR 100 10 "u" "sip+E2U"
"!^.*$!sip:+19725552222@b.example.com!". "!^.*$!sip:+19725552222@b.example.com!".
F3 INVITE Alice -> Proxy 3 F3 INVITE Alice -> Proxy 3
INVITE sip:+19725552222@b.example.com SIP/2.0 INVITE sip:+19725552222@b.example.com SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com> To: <tel:+19725552222>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sip:+13145551111@client.a.example.com> Contact: <sip:+13145551111@client.a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
skipping to change at page 33, line 42 skipping to change at page 33, line 40
F4 INVITE Proxy 3 -> Bob F4 INVITE Proxy 3 -> Bob
INVITE sip:+19725552222@client.b.example.com SIP/2.0 INVITE sip:+19725552222@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1 Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sip:ss3.b.example.com;lr> Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com> To: <tel:+19725552222>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sip:+13145551111@client.a.example.com> Contact: <sip:+13145551111@client.a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=UserA 2890844526 2890844526 IN IP4 client.a.example.com o=UserA 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
SIP PSTN Call Flows April 2003
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F5 100 Trying Proxy 3 -> Alice F5 100 Trying Proxy 3 -> Alice
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com> To: <tel:+19725552222>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Content-Length: 0 Content-Length: 0
F6 180 Ringing B -> Proxy 3 F6 180 Ringing B -> Proxy 3
SIP/2.0 180 Ringing SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1 Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233 ;received=192.0.2.233
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr> Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com>;tag=314159 To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com> Contact: <sip:+19725552222@client.b.example.com>
Content-Length: 0 Content-Length: 0
F7 180 Ringing Proxy 3 -> Alice F7 180 Ringing Proxy 3 -> Alice
SIP/2.0 180 Ringing SIP/2.0 180 Ringing
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr> Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com>;tag=314159 To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com> Contact: <sip:+19725552222@client.b.example.com>
Content-Length: 0 Content-Length: 0
F8 200 OK Bob -> Proxy 3 F8 200 OK Bob -> Proxy 3
SIP PSTN Call Flows April 2003
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1 Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233 ;received=192.0.2.233
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr> Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com>;tag=314159 To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com;transport=tcp> Contact: <sip:+19725552222@client.b.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 151 Content-Length: 151
v=0 v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=- s=-
c=IN IP4 client.b.example.com c=IN IP4 client.b.example.com
skipping to change at page 35, line 35 skipping to change at page 35, line 35
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F9 200 OK Proxy -> Alice F9 200 OK Proxy -> Alice
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr> Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com>;tag=314159 To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com> Contact: <sip:+19725552222@client.b.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 151 Content-Length: 151
v=0 v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=- s=-
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
skipping to change at page 35, line 49 skipping to change at page 36, line 4
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 151 Content-Length: 151
v=0 v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=- s=-
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F10 ACK Alice -> Proxy 3 F10 ACK Alice -> Proxy 3
ACK sip:+19725552222@client.b.example.com SIP/2.0 ACK sip:+19725552222@client.b.example.com SIP/2.0
SIP PSTN Call Flows April 2003
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
Max-Forwards: 70 Max-Forwards: 70
Route: <sip:ss3.b.example.com;lr> Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com>;tag=314159 To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK CSeq: 2 ACK
Content-Length: 0 Content-Length: 0
F11 ACK Proxy 3 -> Bob F11 ACK Proxy 3 -> Bob
ACK sip:+19725552222@client.b.example.com SIP/2.0 ACK sip:+19725552222@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1 Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <sip:+19725552222@b.example.com>;tag=314159 To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK CSeq: 2 ACK
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 0 Content-Length: 0
/* RTP streams are established between A and B*/ /* RTP streams are established between A and B*/
/* User B Hangs Up with User A. */ /* User B Hangs Up with User A. */
F12 BYE Bob -> Proxy 3 F12 BYE Bob -> Proxy 3
BYE sip:+13145551111@client.a.example.com SIP/2.0 BYE sip:+13145551111@client.a.example.com SIP/2.0
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2 Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
Max-Forwards: 70 Max-Forwards: 70
Route: <sip:ss3.b.example.com;lr> Route: <sip:ss3.b.example.com;lr>
From: <sip:+19725552222@b.example.com>;tag=314159 From: <tel:+19725552222>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE CSeq: 1 BYE
Content-Length: 0 Content-Length: 0
F13 BYE Proxy 3 -> Alice F13 BYE Proxy 3 -> Alice
BYE sip:+13145551111@client.a.example.com SIP/2.0 BYE sip:+13145551111@client.a.example.com SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1 Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.100 ;received=192.0.2.100
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2 Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
Max-Forwards: 69 Max-Forwards: 69
SIP PSTN Call Flows April 2003 From: <tel:+19725552222>;tag=314159
From: <sip:+19725552222@b.example.com>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE CSeq: 1 BYE
Content-Length: 0 Content-Length: 0
F14 200 OK Alice -> Proxy 3 F14 200 OK Alice -> Proxy 3
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1 Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233 ;received=192.0.2.233
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2 Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
;received=192.0.2.100 ;received=192.0.2.100
From: <sip:+19725552222@b.example.com>;tag=314159 From: <tel:+19725552222>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE CSeq: 1 BYE
Content-Length: 0 Content-Length: 0
F15 200 OK Proxy 3 -> Bob F15 200 OK Proxy 3 -> Bob
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2 Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
;received=192.0.2.100 ;received=192.0.2.100
From: <sip:+19725552222@b.example.com>;tag=314159 From: <tel:+19725552222>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE CSeq: 1 BYE
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
2.5 Unsuccessful SIP to PSTN call: Treatment from PSTN 2.5. Unsuccessful SIP to PSTN call: Treatment from PSTN
Alice Proxy 1 NGW 1 Bob Alice Proxy 1 NGW 1 Bob
| | | | | | | |
| INVITE F1 | | | | INVITE F1 | | |
|--------------->| | | |--------------->| | |
| 100 F2 | | | | 100 F2 | | |
|<---------------| INVITE F3 | | |<---------------| INVITE F3 | |
| |--------------->| | | |--------------->| |
| | 100 F4 | | | | 100 F4 | |
| |<---------------| IAM F5 | | |<---------------| IAM F5 |
skipping to change at page 38, line 45 skipping to change at page 38, line 44
| | 487 F15 |<---------------| | | 487 F15 |<---------------|
| |<---------------| | | |<---------------| |
| | ACK F16 | | | | ACK F16 | |
| 487 F17 |--------------->| | | 487 F17 |--------------->| |
|<---------------| | | |<---------------| | |
| ACK F18 | | | | ACK F18 | | |
|--------------->| | | |--------------->| | |
| | | | | | | |
Alice calls Bob in the PSTN through a proxy server Proxy 1 and a 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- Network Gateway NGW 1. The call is rejected by the PSTN with an
band treatment (tone or recording) played. Alice hears the in-band treatment (tone or recording) played. Alice hears the
treatment and then hangs up, which results in a CANCEL (F9) being 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 sent to terminate the call. (A BYE is not sent since no final
response was ever received by Alice.) response was ever received by Alice.)
Message Details Message Details
SIP PSTN Call Flows April 2003
F1 INVITE Alice -> Proxy 1 F1 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp> Contact: <sip:alice@client.a.example.com>
Proxy-Authorization: Digest username="alice", Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40", realm="a.example.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40",
opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone", opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
response="e178fbe430e6680a1690261af8831f40" response="e178fbe430e6680a1690261af8831f40"
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
skipping to change at page 39, line 48 skipping to change at page 40, line 4
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
/* Proxy 1 uses a Location Service function to determine where B is /* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW located. Based upon location analysis the call is forwarded to NGW
1. Client for A prepares to receive data on port 49172 from the 1. Client for A prepares to receive data on port 49172 from the
network. */ network. */
F3 INVITE Proxy 1 -> NGW 1 F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
SIP PSTN Call Flows April 2003
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp> Contact: <sip:alice@client.a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
skipping to change at page 40, line 46 skipping to change at page 41, line 4
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F5 IAM NGW 1 -> Bob F5 IAM NGW 1 -> Bob
IAM IAM
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM Bob -> NGW 1 F6 ACM Bob -> NGW 1
ACM ACM
SIP PSTN Call Flows April 2003
F7 183 Session Progress NGW 1 -> Proxy 1 F7 183 Session Progress NGW 1 -> Proxy 1
SIP/2.0 183 Session Progress SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
skipping to change at page 41, line 44 skipping to change at page 41, line 46
SIP/2.0 183 Session Progress SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
SIP PSTN Call Flows April 2003
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
/* Caller hears the recorded announcement, then hangs up */ /* Caller hears the recorded announcement, then hangs up */
F9 CANCEL Alice -> Proxy 1 F9 CANCEL Alice -> Proxy 1
CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
skipping to change at page 42, line 45 skipping to change at page 43, line 4
CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 CANCEL CSeq: 1 CANCEL
Content-Length: 0 Content-Length: 0
F12 200 OK NGW 1 -> Proxy 1 F12 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
SIP PSTN Call Flows April 2003
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 CANCEL CSeq: 1 CANCEL
Content-Length: 0 Content-Length: 0
F13 REL NGW 1 -> B F13 REL NGW 1 -> B
REL REL
skipping to change at page 44, line 4 skipping to change at page 44, line 8
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
F17 487 Request Terminated Proxy 1 -> A F17 487 Request Terminated Proxy 1 -> A
SIP/2.0 487 Request Terminated SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
skipping to change at page 45, line 4 skipping to change at page 45, line 4
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
2.6 Unsuccessful SIP to PSTN: REL w/Cause from PSTN 2.6. Unsuccessful SIP to PSTN: REL w/Cause from PSTN
Alice Proxy 1 NGW 1 Switch B Alice Proxy 1 NGW 1 Switch B
| | | | | | | |
| INVITE F1 | | | | INVITE F1 | | |
|--------------->| | | |--------------->| | |
| 100 F2 | | | | 100 F2 | | |
|<---------------| INVITE F3 | | |<---------------| INVITE F3 | |
| |--------------->| | | |--------------->| |
| | 100 F4 | | | | 100 F4 | |
| |<---------------| IAM F5 | | |<---------------| IAM F5 |
skipping to change at page 45, line 32 skipping to change at page 45, line 31
| |<---------------| | | |<---------------| |
| | ACK F9 | | | | ACK F9 | |
| |--------------->| | | |--------------->| |
| 404 F10 | | | | 404 F10 | | |
|<---------------| | | |<---------------| | |
| ACK F11 | | | | ACK F11 | | |
|--------------->| | | |--------------->| | |
| | | | | | | |
Alice calls PSTN Bob through a Proxy Server Proxy 1 and a Network 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 Gateway NGW 1. The call is rejected by the PSTN with a ANSI ISUP
ANSI ISUP Release message REL containing a specific Cause code. Release message REL containing a specific Cause code. This cause
This cause value (1) is mapped by the Gateway to a SIP 404 Address value (1) is mapped by the Gateway to a SIP 404 Address Incomplete
Incomplete response which is proxied back to Alice. For more response which is proxied back to Alice. For more details of ISUP
details of ISUP cause value to SIP response mapping refer to [4]. cause value to SIP response mapping, refer to [4].
Message Details Message Details
F1 INVITE Alice -> Proxy 1 F1 INVITE Alice -> Proxy 1
INVITE sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0 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 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone> To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp> Contact: <sip:alice@client.a.example.com;transport=tcp>
Proxy-Authorization: Digest username="alice", Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40", realm="a.example.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",
SIP PSTN Call Flows April 2003
opaque="", uri="sip:+44-1234@ss1.a.example.com;user=phone", opaque="", uri="sip:+44-1234@ss1.a.example.com;user=phone",
response="a451358d46b55512863efe1dccaa2f42" response="a451358d46b55512863efe1dccaa2f42"
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
skipping to change at page 47, line 4 skipping to change at page 47, line 4
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone> To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp> Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
SIP PSTN Call Flows April 2003
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1 F4 100 Trying NGW 1 -> Proxy 1
skipping to change at page 47, line 45 skipping to change at page 48, line 4
REL REL
CauseValue=1 Unallocated number CauseValue=1 Unallocated number
F7 RLC NGW 1 -> Bob F7 RLC NGW 1 -> Bob
RLC RLC
/* Network Gateway maps CauseValue=1 to the SIP message 404 Not /* Network Gateway maps CauseValue=1 to the SIP message 404 Not
Found */ Found */
F8 404 Not Found NGW 1 -> Proxy 1 F8 404 Not Found NGW 1 -> Proxy 1
SIP/2.0 404 Not Found SIP/2.0 404 Not Found
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
SIP PSTN Call Flows April 2003
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159 To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Error-Info: <sip:not-found-ann@ann.a.example.com> Error-Info: <sip:not-found-ann@ann.a.example.com>
Content-Length: 0 Content-Length: 0
skipping to change at page 49, line 4 skipping to change at page 49, line 9
F11 ACK Alice -> Proxy 1 F11 ACK Alice -> Proxy 1
ACK sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0 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 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159 To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK CSeq: 1 ACK
SIP PSTN Call Flows April 2003
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
2.7 Unsuccessful SIP to PSTN: ANM Timeout 2.7. Unsuccessful SIP to PSTN: ANM Timeout
Alice Proxy 1 NGW 1 Switch B Alice Proxy 1 NGW 1 Switch B
| | | | | | | |
| INVITE F1 | | | | INVITE F1 | | |
|--------------->| | | |--------------->| | |
| 100 F2 | | | | 100 F2 | | |
|<---------------| INVITE F3 | | |<---------------| INVITE F3 | |
| |--------------->| | | |--------------->| |
| | 100 F4 | | | | 100 F4 | |
| |<---------------| IAM F5 | | |<---------------| IAM F5 |
skipping to change at page 51, line 4 skipping to change at page 50, line 16
F1 INVITE Alice -> Proxy 1 F1 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70 Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
SIP PSTN Call Flows April 2003
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp> Contact: <sip:alice@client.a.example.com;transport=tcp>
Proxy-Authorization: Digest username="alice", Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40", realm="a.example.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40",
opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone", opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
response="579cb9db184cdc25bf816f37cbc03c7d" response="579cb9db184cdc25bf816f37cbc03c7d"
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
skipping to change at page 52, line 4 skipping to change at page 51, line 17
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 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 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
SIP PSTN Call Flows April 2003
Contact: <sip:alice@client.a.example.com;transport=tcp> Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 154 Content-Length: 154
v=0 v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=- s=-
c=IN IP4 client.a.example.com c=IN IP4 client.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
skipping to change at page 53, line 4 skipping to change at page 52, line 17
F7 183 Session Progress NGW 1 -> Proxy 1 F7 183 Session Progress NGW 1 -> Proxy 1
SIP/2.0 183 Session Progress SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
SIP PSTN Call Flows April 2003
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
skipping to change at page 54, line 4 skipping to change at page 53, line 18
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
/* After NGW 1's timer expires, Network Gateway sends REL to ISUP /* After NGW 1's timer expires, Network Gateway sends REL to ISUP
network and 480 to SIP network */ network and 480 to SIP network */
F9 REL NGW 1 -> Bob F9 REL NGW 1 -> Bob
REL REL
SIP PSTN Call Flows April 2003
CauseCode=18 No user responding CauseCode=18 No user responding
F10 RLC Bob -> NGW 1 F10 RLC Bob -> NGW 1
RLC RLC
F11 480 Temporarily Unavailable NGW 1 -> Proxy 1 F11 480 Temporarily Unavailable NGW 1 -> Proxy 1
SIP/2.0 480 Temporarily Unavailable SIP/2.0 480 Temporarily Unavailable
skipping to change at page 55, line 4 skipping to change at page 54, line 20
F13 480 Temporarily Unavailable F13 Proxy 1 -> Alice F13 480 Temporarily Unavailable F13 Proxy 1 -> Alice
SIP/2.0 480 Temporarily Unavailable SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101 ;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
SIP PSTN Call Flows April 2003
CSeq: 1 INVITE CSeq: 1 INVITE
Error-Info: <sip:temp-unavail-ann@ann.a.example.com> Error-Info: <sip:temp-unavail-ann@ann.a.example.com>
Content-Length: 0 Content-Length: 0
F14 ACK Alice -> Proxy 1 F14 ACK Alice -> Proxy 1
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Max-Forwards: 70 Max-Forwards: 70
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl ;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159 ;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
3. PSTN to SIP Dialing 3. PSTN to SIP Dialing
In these scenarios, Alice is placing calls from the PSTN to Bob In these scenarios, Alice is placing calls from the PSTN to Bob in a
in a SIP network. Alice's telephone switch signals to a Network SIP network. Alice's telephone switch signals to a Network Gateway
Gateway (NGW 1) using ANSI ISUP. (NGW 1) using ANSI ISUP.
Since the called SIP User Agent does not send in-band signaling Since the called SIP User Agent does not send in-band signaling
information, no early media path needs to be established on the IP information, no early media path needs to be established on the IP
side. As a result, the 183 Session Progress response is not used. 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 However, NGW 1 will establish a one way speech path prior to call
completion, and generate ringing for the PSTN caller. Any tones or completion, and generate ringing for the PSTN caller. Any tones or
recordings are generated by NGW 1 and played in this speech path. recordings are generated by NGW 1 and played in this speech path.
When the call completes successfully, NGW 1 bridges the PSTN speech When the call completes successfully, NGW 1 bridges the PSTN speech
path with the IP media path. path with the IP media path.
To reduce the number of messages, only a single proxy server is shown 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 in these flows, which means that the a.example.com proxy server has
access to the b.example.com location service. access to the b.example.com location service.
SIP PSTN Call Flows April 2003 3.1. Successful PSTN to SIP call
3.1 Successful PSTN to SIP call
Switch A NGW 1 Proxy 1 Bob Switch A NGW 1 Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
| | 100 F4 |--------------->| | | 100 F4 |--------------->|
| |<---------------| | | |<---------------| |
| | | 180 F5 | | | | 180 F5 |
| | 180 F6 |<---------------| | | 180 F6 |<---------------|
skipping to change at page 57, line 42 skipping to change at page 55, line 47
| RLC F14 | | | | RLC F14 | | |
|<---------------| BYE F15 | | |<---------------| BYE F15 | |
| |--------------->| BYE F16 | | |--------------->| BYE F16 |
| | |--------------->| | | |--------------->|
| | | 200 F17 | | | | 200 F17 |
| | 200 F18 |<---------------| | | 200 F18 |<---------------|
| |<---------------| | | |<---------------| |
| | | | | | | |
In this scenario, Alice from the PSTN calls Bob through a Network In this scenario, Alice from the PSTN calls Bob through a Network
Gateway NGW1 and Proxy Server Proxy 1. When Bob answers the call 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 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 hangs up the call, with Alice's telephone switch sending an ISUP
RELease message which is mapped to a BYE by NGW 1. RELease message that is mapped to a BYE by NGW 1.
Message Details Message Details
F1 IAM Alice -> NGW 1 F1 IAM Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
SIP PSTN Call Flows April 2003
F2 INVITE Alice -> Proxy 1 F2 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
skipping to change at page 59, line 4 skipping to change at page 57, line 26
Contact: <sip:ngw1@a.example.com> Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
SIP PSTN Call Flows April 2003
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying Bob -> Proxy 1 F4 100 Trying Bob -> Proxy 1
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
skipping to change at page 60, line 4 skipping to change at page 58, line 24
;received=192.0.2.103 ;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com> Contact: <sip:bob@client.b.example.com>
Content-Length: 0 Content-Length: 0
F7 ACM NGW 1 -> Alice F7 ACM NGW 1 -> Alice
SIP PSTN Call Flows April 2003
ACM ACM
F8 200 OK Bob -> Proxy 1 F8 200 OK Bob -> Proxy 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
skipping to change at page 61, line 4 skipping to change at page 59, line 27
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 151 Content-Length: 151
v=0 v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=- s=-
c=IN IP4 client.b.example.com c=IN IP4 client.b.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
SIP PSTN Call Flows April 2003
F10 ACK NGW 1 -> Proxy 1 F10 ACK NGW 1 -> Proxy 1
ACK sip:bob@client.b.example.com SIP/2.0 ACK sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr> Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
skipping to change at page 62, line 4 skipping to change at page 60, line 24
/* Alice Hangs Up with Bob. */ /* Alice Hangs Up with Bob. */
F13 REL Alice -> NGW 1 F13 REL Alice -> NGW 1
REL REL
CauseCode=16 Normal CauseCode=16 Normal
F14 RLC NGW 1 -> Alice F14 RLC NGW 1 -> Alice
RLC RLC
SIP PSTN Call Flows April 2003
F15 BYE NGW 1-> Proxy 1 F15 BYE NGW 1-> Proxy 1
BYE sip:bob@client.b.example.com SIP/2.0 BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr> Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
skipping to change at page 63, line 4 skipping to change at page 61, line 28
Content-Length: 0 Content-Length: 0
F18 200 OK Proxy 1 -> NGW 1 F18 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
SIP PSTN Call Flows April 2003
CSeq: 2 BYE CSeq: 2 BYE
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
3.2 Successful PSTN to SIP call, Fast Answer 3.2. Successful PSTN to SIP call, Fast Answer
Switch A NGW 1 Proxy 1 Bob Switch A NGW 1 Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
| | 100 F4 |--------------->| | | 100 F4 |--------------->|
| |<---------------| | | |<---------------| |
| | | 200 F5 | | | | 200 F5 |
| | 200 F6 |<---------------| | | 200 F6 |<---------------|
skipping to change at page 64, line 34 skipping to change at page 62, line 33
|--------------->| | | |--------------->| | |
| RLC F11 | | | | RLC F11 | | |
|<---------------| BYE F12 | | |<---------------| BYE F12 | |
| |--------------->| BYE F13 | | |--------------->| BYE F13 |
| | |--------------->| | | |--------------->|
| | | 200 F14 | | | | 200 F14 |
| | 200 F15 |<---------------| | | 200 F15 |<---------------|
| |<---------------| | | |<---------------| |
| | | | | | | |
This "fast answer" scenario is similar to 3.1 except that Bob This "fast answer" scenario is similar to 3.1., except that Bob
immediately accepts the call, sending a 200 OK (F5) without sending a immediately accepts the call, sending a 200 OK (F5) without sending a
180 Ringing response. The Gateway then sends an Answer Message (ANM) 180 Ringing response. The Gateway then sends an Answer Message (ANM)
without sending an Address Complete Message (ACM). Note that for without sending an Address Complete Message (ACM). Note that for
ETSI and some other ISUP variants, a CONnect message (CON) would be ETSI and some other ISUP variants, a CONnect message (CON) would be
sent instead of the ANM. sent instead of the ANM.
Message Details Message Details
F1 IAM Alice -> NGW 1 F1 IAM Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1 F2 INVITE NGW 1 -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
SIP PSTN Call Flows April 2003
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
skipping to change at page 65, line 50 skipping to change at page 64, line 4
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1 F4 100 Trying Proxy 1 -> NGW 1
SIP PSTN Call Flows April 2003
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201 ;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
skipping to change at page 67, line 4 skipping to change at page 65, line 6
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com;transport=tcp> Contact: <sip:bob@client.b.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 151 Content-Length: 151
SIP PSTN Call Flows April 2003
v=0 v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=- s=-
c=IN IP4 client.b.example.com c=IN IP4 client.b.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F7 ACK NGW 1 -> Proxy 1 F7 ACK NGW 1 -> Proxy 1
skipping to change at page 67, line 46 skipping to change at page 66, line 4
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
F9 ANM Bob -> NGW 1 F9 ANM Bob -> NGW 1
ANM ANM
/* RTP streams are established between A and B (via the GW) */ /* RTP streams are established between A and B (via the GW) */
/* Alice Hangs Up with Bob. */ /* Alice Hangs Up with Bob. */
F10 REL ser Alice -> NGW 1 F10 REL ser Alice -> NGW 1
REL REL
CauseCode=16 Normal CauseCode=16 Normal
SIP PSTN Call Flows April 2003
F11 RLC NGW 1 -> Alice F11 RLC NGW 1 -> Alice
RLC RLC
F12 BYE NGW 1 -> Proxy 1 F12 BYE NGW 1 -> Proxy 1
BYE sip:bob@client.b.example.com SIP/2.0 BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
skipping to change at page 69, line 4 skipping to change at page 67, line 8
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE CSeq: 2 BYE
Content-Length: 0 Content-Length: 0
F15 200 OK Proxy 1 -> NGW 1 F15 200 OK Proxy 1 -> NGW 1
SIP PSTN Call Flows April 2003
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE CSeq: 2 BYE
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
3.3 Successful PBX to SIP call 3.3. Successful PBX to SIP call
PBX A GW 1 Proxy 1 Bob PBX A GW 1 Proxy 1 Bob
| | | | | | | |
| Seizure | | | | Seizure | | |
|--------------->| | | |--------------->| | |
| Wink | | | | Wink | | |
|<---------------| | | |<---------------| | |
| MF Digits F1 | | | | MF Digits F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
skipping to change at page 70, line 45 skipping to change at page 68, line 44
|<---------------| BYE F11 | | |<---------------| BYE F11 | |
| |--------------->| BYE F12 | | |--------------->| BYE F12 |
| | |--------------->| | | |--------------->|
| | | 200 F13 | | | | 200 F13 |
| | 200 F14 |<---------------| | | 200 F14 |<---------------|
| |<---------------| | | |<---------------| |
| | | | | | | |
In this scenario, Alice dials from PBX A to Bob through GW 1 and 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 Proxy 1. This is an example of a call that appears destined for the
PSTN but instead is routed to a SIP Client. PSTN but is instead routed to a SIP Client.
Signaling between PBX A and GW 1 is Feature Group B (FGB) circuit Signaling between PBX A and GW 1 is Feature Group B (FGB) circuit
associated signaling, in-band Mult-Frequency (MF) outpulsing. After associated signaling, in-band Mult-Frequency (MF) outpulsing. After
the receipt of the 180 Ringing from Bob, GW 1 generates ringing the receipt of the 180 Ringing from Bob, GW 1 generates a ringing
tone for Alice. tone for Alice.
Bob answers the call by sending a 200 OK. The call terminates Bob answers the call by sending a 200 OK. The call terminates when
when Alice hangs up, causing GW1 to send a BYE. Alice hangs up, causing GW1 to send a BYE.
SIP PSTN Call Flows April 2003
The Gateway can only identify the trunk group that the The Gateway can only identify the trunk group that the call came in
call came in on, it cannot identify the individual line on PBX A that on; it cannot identify the individual line on PBX A that is placing
is placing the call. The SIP URI used to identify the caller is the call. The SIP URI used to identify the caller is shown in these
shown in these flows as sip:551313@gw1.a.example.com. flows as sip:551313@gw1.a.example.com.
Message Details Message Details
PBX Alice -> GW 1 PBX Alice -> GW 1
Seizure Seizure
GW 1 -> PBX A GW 1 -> PBX A
Wink Wink
skipping to change at page 72, line 4 skipping to change at page 70, line 4
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 gw1.a.example.com c=IN IP4 gw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service function to determine where the /* Proxy 1 uses a Location Service function to determine where the
phone number +19725552222 is located. Based upon location phone number +19725552222 is located. Based upon location
analysis the call is forwarded to SIP Bob. */ analysis the call is forwarded to SIP Bob. */
SIP PSTN Call Flows April 2003
F3 INVITE Proxy 1 -> Bob F3 INVITE Proxy 1 -> Bob
INVITE sip:bob@client.b.example.com SIP/2.0 INVITE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
skipping to change at page 73, line 4 skipping to change at page 71, line 5
SIP/2.0 180 Ringing SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201 ;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
SIP PSTN Call Flows April 2003
Contact: <sip:bob@client.b.example.com> Contact: <sip:bob@client.b.example.com>
Content-Length: 0 Content-Length: 0
F6 180 Ringing Proxy 1 -> GW 1 F6 180 Ringing Proxy 1 -> GW 1
SIP/2.0 180 Ringing SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
skipping to change at page 73, line 48 skipping to change at page 72, line 4
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 151 Content-Length: 151
v=0 v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=- s=-
c=IN IP4 client.b.example.com c=IN IP4 client.b.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F8 200 OK Proxy 1 -> GW 1 F8 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK SIP/2.0 200 OK
SIP PSTN Call Flows April 2003
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com> Contact: <sip:bob@client.b.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 151 Content-Length: 151
skipping to change at page 74, line 49 skipping to change at page 73, line 4
ACK sip:bob@client.b.example.com SIP/2.0 ACK sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Max-Forwards: 69 Max-Forwards: 69
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
/* RTP streams are established between A and B (via the GW) */ /* RTP streams are established between A and B (via the GW) */
SIP PSTN Call Flows April 2003
/* Alice Hangs Up with Bob. */ /* Alice Hangs Up with Bob. */
F11 BYE GW 1 -> Proxy 1 F11 BYE GW 1 -> Proxy 1
BYE sip:bob@client.b.example.com SIP/2.0 BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70 Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr> Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
skipping to change at page 75, line 44 skipping to change at page 74, line 4
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE CSeq: 2 BYE
Content-Length: 0 Content-Length: 0
F14 200 OK Proxy 1 -> GW 1 F14 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
SIP PSTN Call Flows April 2003
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 2 BYE CSeq: 2 BYE
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
3.4 Unsuccessful PSTN to SIP REL, SIP error mapped to REL 3.4. Unsuccessful PSTN to SIP REL, SIP error mapped to REL
Switch A GW 1 Proxy 1 Bob Switch A GW 1 Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| | | |--------------->| |
| | 604 F3 | | | | 604 F3 | |
| |<---------------| | | |<---------------| |
| | ACK F4 | | | | ACK F4 | |
| |--------------->| | | |--------------->| |
skipping to change at page 78, line 4 skipping to change at page 75, line 15
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: Contact:
<sip:+13145551111@gw1.a.example.com;user=phone;transport=tcp> <sip:+13145551111@gw1.a.example.com;user=phone;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 144 Content-Length: 144
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=- s=-
SIP PSTN Call Flows April 2003
c=IN IP4 gw1.a.example.com c=IN IP4 gw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
/* Proxy 1 uses a Location Service to find a route to +1-972-555- /* 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. */ 9999. A route is not found, so Proxy 1 rejects the call. */
F3 604 Does Not Exist Anywhere Proxy 1 -> GW 1 F3 604 Does Not Exist Anywhere Proxy 1 -> GW 1
skipping to change at page 78, line 36 skipping to change at page 76, line 4
F4 ACK GW 1 -> Proxy 1 F4 ACK GW 1 -> Proxy 1
ACK sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0 ACK sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410 To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
F5 REL GW 1 -> Alice F5 REL GW 1 -> Alice
REL REL
CauseCode=1 CauseCode=1
F6 RLC Alice -> GW 1 F6 RLC Alice -> GW 1
RLC RLC
SIP PSTN Call Flows April 2003
3.5 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL 3.5. Unsuccessful PSTN to SIP REL, SIP busy mapped to REL
Switch A NGW 1 Proxy 1 Bob Switch A NGW 1 Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
| | 100 F4 |--------------->| | | 100 F4 |--------------->|
| |<---------------| | | |<---------------| |
| | | 600 F5 | | | | 600 F5 |
| | |<---------------| | | |<---------------|
skipping to change at page 79, line 28 skipping to change at page 76, line 35
| | 600 F7 |--------------->| | | 600 F7 |--------------->|
| |<---------------| | | |<---------------| |
| | ACK F8 | | | | ACK F8 | |
| |--------------->| | | |--------------->| |
| REL(17) F9 | | | | REL(17) F9 | | |
|<---------------| | | |<---------------| | |
| RLC F10 | | | | RLC F10 | | |
|<-------------->| | | |<-------------->| | |
| | | | | | | |
In this scenario, Alice calls Bob through Network Gateway NGW 1 In this scenario, Alice calls Bob through Network Gateway NGW 1 and
and Proxy 1. The call is routed to Bob by Proxy 1. The call is Proxy 1. The call is routed to Bob by Proxy 1. The call is rejected
rejected by Bob who sends a 600 Busy Everywhere response. The by Bob who sends a 600 Busy Everywhere response. The Gateway sends a
Gateway sends a REL message containing a specific Cause value mapped REL message containing a specific Cause value mapped by the gateway
by the gateway based on the SIP error. based on the SIP error.
Since no interworking is indicated in the IAM (F1), the busy tone is Since no interworking is indicated in the IAM (F1), the busy tone is
generated locally by Alice's telephone switch. In some scenarios, generated locally by Alice's telephone switch. In some scenarios,
the busy signal is generated by the Gateway since interworking is the busy signal is generated by the Gateway since interworking is
indicated. For more discussion on interworking, refer to [4]. indicated. For more discussion on interworking, refer to [4].
Message Details Message Details
F1 IAM Alice -> NGW 1 F1 IAM Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1 F2 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
SIP PSTN Call Flows April 2003
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 144 Content-Length: 144
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
skipping to change at page 81, line 4 skipping to change at page 78, line 18
s=- s=-
c=IN IP4 gw1.a.example.com c=IN IP4 gw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1 F4 100 Trying Proxy 1 -> NGW 1
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
SIP PSTN Call Flows April 2003
;received=192.0.2.201 ;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F5 600 Busy Everywhere Bob -> Proxy 1 F5 600 Busy Everywhere Bob -> Proxy 1
SIP/2.0 600 Busy Everywhere SIP/2.0 600 Busy Everywhere
skipping to change at page 82, line 4 skipping to change at page 79, line 18
;received=192.0.2.201 ;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F8 ACK NGW 1 -> Proxy 1 F8 ACK NGW 1 -> Proxy 1
ACK bob@b.example.com SIP/2.0 ACK bob@b.example.com SIP/2.0
SIP PSTN Call Flows April 2003
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
F9 REL NGW 1 -> Alice F9 REL NGW 1 -> Alice
REL REL
CauseCode=17 Busy CauseCode=17 Busy
F10 RLC Alice -> NGW 1 F10 RLC Alice -> NGW 1
RLC RLC
SIP PSTN Call Flows April 2003
3.6 Unsuccessful PSTN->SIP, SIP error interworking to tones 3.6. Unsuccessful PSTN->SIP, SIP error interworking to tones
Switch A NGW 1 Proxy 1 Bob Switch A NGW 1 Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
| | 100 F4 |--------------->| | | 100 F4 |--------------->|
| |<---------------| | | |<---------------| |
| | | 600 F5 | | | | 600 F5 |
| | |<---------------| | | |<---------------|
skipping to change at page 83, line 33 skipping to change at page 80, line 32
| One Way Voice | | | | One Way Voice | | |
|<===============| | | |<===============| | |
| Busy Tone | | | | Busy Tone | | |
|<===============| | | |<===============| | |
| REL(16) F10 | | | | REL(16) F10 | | |
|--------------->| | | |--------------->| | |
| RLC F11 | | | | RLC F11 | | |
|<---------------| | | |<---------------| | |
| | | | | | | |
In this scenario, Alice calls Bob through Network Gateway NGW1 In this scenario, Alice calls Bob through Network Gateway NGW 1 and
and Proxy 1. The call is routed to Bob by Proxy 1. The call is Proxy 1. The call is routed to Bob by Proxy 1. The call is rejected
rejected by the Bob client. NGW 1 sets up a two way voice path to by the Bob client. NGW 1 sets up a two way voice path to Alice and
Alice and plays busy tone. The caller then disconnects plays busy tone. The caller then disconnects
NGW 1 plays the busy tone since the IAM (F1) indicates the NGW 1 plays the busy tone since the IAM (F1) indicates the
interworking is present. In scenario 5.2.2, with no interworking, interworking is present. In scenario 5.2.2., with no interworking,
the busy indication is carried in the REL Cause value and is the busy indication is carried in the REL Cause value and is
generated locally instead. generated locally instead.
Again, note that for ETSI or ITU ISUP, a CONnect message would be Again, note that for ETSI or ITU ISUP, a CONnect message would be
sent instead of the Answer Message. sent instead of the Answer Message.
Message Details Message Details
F1 IAM Alice -> NGW 1 F1 IAM Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
SIP PSTN Call Flows April 2003
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
Interworking=encountered Interworking=encountered
F2 INVITE NGW1 -> Proxy 1 F2 INVITE NGW1 -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com;transport=tcp>
skipping to change at page 85, line 4 skipping to change at page 81, line 48
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
SIP PSTN Call Flows April 2003
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying Bob -> Proxy 1 F4 100 Trying Bob -> Proxy 1
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
skipping to change at page 85, line 46 skipping to change at page 83, line 4
F6 ACK Proxy 1 -> Bob F6 ACK Proxy 1 -> Bob
ACK bob@b.example.com SIP/2.0 ACK 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 ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
F7 600 Busy Everywhere Proxy 1 -> NGW 1 F7 600 Busy Everywhere Proxy 1 -> NGW 1
SIP/2.0 600 Busy Everywhere SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
SIP PSTN Call Flows April 2003
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F8 ACK NGW 1 -> Proxy 1 F8 ACK NGW 1 -> Proxy 1
ACK sip:ngw1@a.example.com SIP/2.0 ACK sip:ngw1@a.example.com SIP/2.0
skipping to change at page 87, line 4 skipping to change at page 84, line 4
/* Call Released after Alice hangs up. */ /* Call Released after Alice hangs up. */
F10 REL Alice -> NGW 1 F10 REL Alice -> NGW 1
REL REL
CauseCode=16 CauseCode=16
F11 RLC NGW 1 -> Alice F11 RLC NGW 1 -> Alice
RLC RLC
SIP PSTN Call Flows April 2003
3.7 Unsuccessful PSTN->SIP, ACM timeout 3.7. Unsuccessful PSTN->SIP, ACM timeout
Switch A NGW 1 Proxy 1 Bob Switch A NGW 1 Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
| | 100 F4 |--------------->| | | 100 F4 |--------------->|
| |<---------------| | | |<---------------| |
| | | INVITE F5 | | | | INVITE F5 |
| | |--------------->| | | |--------------->|
skipping to change at page 87, line 36 skipping to change at page 84, line 35
|--------------->| | | |--------------->| | |
| RLC F11 | | | | RLC F11 | | |
|<---------------| | | |<---------------| | |
| | CANCEL F12 | | | | CANCEL F12 | |
| |--------------->| | | |--------------->| |
| | 200 F13 | | | | 200 F13 | |
| |<---------------| | | |<---------------| |
Alice calls Bob through NGW 1 and Proxy 1. Proxy 1 re-sends the 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 INVITE after the expiration of SIP timer T1 without receiving any
response from Bob. Bob never responds with 180 Ringing or any response from Bob. Bob never responds with 180 Ringing or any other
other response (it is reachable but unresponsive). After the response (it is reachable but unresponsive). After the expiration of
expiration of a timer, Alice's network disconnects the call by a timer, Alice's network disconnects the call by sending a Release
sending a Release message REL. The Gateway maps this to a CANCEL. message REL. The Gateway maps this to a CANCEL.
Message Details Message Details
F1 IAM Alice -> NGW 1 F1 IAM Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1 F2 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
SIP PSTN Call Flows April 2003
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com> Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
skipping to change at page 88, line 45 skipping to change at page 86, line 4
Contact: <sip:ngw1@a.example.com> Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
c c=IN IP4 ngw1.a.example.com c c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1 F4 100 Trying Proxy 1 -> NGW 1
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
SIP PSTN Call Flows April 2003
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F5 INVITE Proxy 1 -> Bob F5 INVITE Proxy 1 -> Bob
Same as Message F3 Same as Message F3
F6 INVITE Proxy 1 -> Bob F6 INVITE Proxy 1 -> Bob
skipping to change at page 89, line 40 skipping to change at page 87, line 4
/* Timer expires in Alice's access network. */ /* Timer expires in Alice's access network. */
F10 REL Alice -> NGW 1 F10 REL Alice -> NGW 1
REL REL
CauseCode=16 Normal CauseCode=16 Normal
F11 RLC NGW 1 -> Alice F11 RLC NGW 1 -> Alice
RLC RLC
F12 CANCEL NGW 1 -> Proxy 1 F12 CANCEL NGW 1 -> Proxy 1
CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
SIP PSTN Call Flows April 2003
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL CSeq: 1 CANCEL
Content-Length: 0 Content-Length: 0
F13 200 OK Proxy 1 -> NGW 1 F13 200 OK Proxy 1 -> NGW 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL CSeq: 1 CANCEL
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
3.8 Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy 3.8. Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy
Switch A NGW 1 Stateless Proxy 1 Bob Switch A NGW 1 Stateless Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
| | INVITE F4 |--------------->| | | INVITE F4 |--------------->|
| |--------------->| INVITE F5 | | |--------------->| INVITE F5 |
| | INVITE F6 |--------------->| | | INVITE F6 |--------------->|
| |--------------->| INVITE F7 | | |--------------->| INVITE F7 |
skipping to change at page 91, line 29 skipping to change at page 88, line 28
| | INVITE F10 |--------------->| | | INVITE F10 |--------------->|
| |--------------->| INVITE F11 | | |--------------->| INVITE F11 |
| | INVITE F12 |--------------->| | | INVITE F12 |--------------->|
| |--------------->| INVITE F13 | | |--------------->| INVITE F13 |
| | |--------------->| | | |--------------->|
| REL F14 | | | | REL F14 | | |
|--------------->| | | |--------------->| | |
| RLC F15 | | | | RLC F15 | | |
|<---------------| | | |<---------------| | |
In this scenario, Alice calls Bob through NGW 1 and Proxy 1. In this scenario, Alice calls Bob through NGW 1 and Proxy 1. Since
Since Proxy 1 is stateless (it does not send a 100 Trying response), Proxy 1 is stateless (it does not send a 100 Trying response), NGW 1
NGW 1 re-sends the INVITE message after the expiration of re-sends the INVITE message after the expiration of SIP timer T1.
SIP timer T1. Bob does not respond with 180 Ringing. Alice's Bob does not respond with 180 Ringing. Alice's network disconnects
network disconnects the call with a release REL (CauseCode=102 the call with a release REL (CauseCode=102 Timeout).
Timeout).
Message Details Message Details
F1 IAM Alice -> NGW 1 F1 IAM Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1 F2 INVITE NGW 1 -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
SIP PSTN Call Flows April 2003
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com> Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
skipping to change at page 93, line 4 skipping to change at page 90, line 4
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 INVITE NGW 1 -> Proxy 1 F4 INVITE NGW 1 -> Proxy 1
Same as Message F2 Same as Message F2
F5 INVITE Proxy 1 -> Bob F5 INVITE Proxy 1 -> Bob
Same as Message F3 Same as Message F3
SIP PSTN Call Flows April 2003
F6 INVITE NGW 1 -> Proxy 1 F6 INVITE NGW 1 -> Proxy 1
Same as Message F2 Same as Message F2
F7 INVITE Proxy 1 -> Bob F7 INVITE Proxy 1 -> Bob
Same as Message F3 Same as Message F3
F8 INVITE NGW 1 -> Proxy 1 F8 INVITE NGW 1 -> Proxy 1
skipping to change at page 94, line 4 skipping to change at page 91, line 4
F13 INVITE Proxy 1 -> Bob F13 INVITE Proxy 1 -> Bob
Same as Message F3 Same as Message F3
/* A timer expires in Alice's access network. */ /* A timer expires in Alice's access network. */
F14 REL Alice -> NGW 1 F14 REL Alice -> NGW 1
REL REL
CauseCode=102 Timeout CauseCode=102 Timeout
SIP PSTN Call Flows April 2003
F15 RLC NGW 1 -> Alice F15 RLC NGW 1 -> Alice
RLC RLC
SIP PSTN Call Flows April 2003
3.9 Unsuccessful PSTN->SIP, Caller Abandonment 3.9. Unsuccessful PSTN->SIP, Caller Abandonment
Switch A NGW 1 Proxy 1 Bob Switch A NGW 1 Proxy 1 Bob
| | | | | | | |
| IAM F1 | | | | IAM F1 | | |
|--------------->| INVITE F2 | | |--------------->| INVITE F2 | |
| |--------------->| INVITE F3 | | |--------------->| INVITE F3 |
| | 100 F4 |--------------->| | | 100 F4 |--------------->|
| |<---------------| | | |<---------------| |
| | | 180 F5 | | | | 180 F5 |
| | 180 F6 |<---------------| | | 180 F6 |<---------------|
skipping to change at page 95, line 44 skipping to change at page 91, line 46
| | |<---------------| | | |<---------------|
| | | 487 F14 | | | | 487 F14 |
| | |<---------------| | | |<---------------|
| | | ACK F15 | | | | ACK F15 |
| | 487 F16 |--------------->| | | 487 F16 |--------------->|
| |<---------------| | | |<---------------| |
| | ACK F17 | | | | ACK F17 | |
| |--------------->| | | |--------------->| |
| | | | | | | |
In this scenario, Alice calls Bob through NGW 1 and Proxy 1. In this scenario, Alice calls Bob through NGW 1 and Proxy 1. Bob
Bob does not respond with 200 OK. NGW 1 plays ringing tone since does not respond with 200 OK. NGW 1 plays ringing tone since the ACM
the ACM indicates that interworking has been encountered. Alice indicates that interworking has been encountered. Alice disconnects
disconnects the call with a Release message REL which is mapped by the call with a Release message REL which is mapped by NGW 1 to a
NGW 1 to a CANCEL. Note that if Bob had sent a 200 OK response CANCEL. Note that if Bob had sent a 200 OK response after the REL,
after the REL, NGW 1 would have sent an ACK then a BYE to properly NGW 1 would have sent an ACK and then a BYE to properly terminate the
terminate the call. call.
Message Details Message Details
SIP PSTN Call Flows April 2003
F1 IAM Alice -> NGW 1 F1 IAM Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1 F2 INVITE Alice -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
skipping to change at page 97, line 4 skipping to change at page 93, line 9
;received=192.0.2.103 ;received=192.0.2.103
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp> Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
SIP PSTN Call Flows April 2003
v=0 v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=- s=-
c=IN IP4 ngw1.a.example.com c=IN IP4 ngw1.a.example.com
t=0 0 t=0 0
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
F4 100 Trying Bob -> Proxy 1 F4 100 Trying Bob -> Proxy 1
skipping to change at page 98, line 4 skipping to change at page 94, line 10
F6 180 Ringing Proxy 1 -> NGW 1 F6 180 Ringing Proxy 1 -> NGW 1
SIP/2.0 180 Ringing SIP/2.0 180 Ringing
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
SIP PSTN Call Flows April 2003
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com> Contact: <sip:bob@client.b.example.com>
Content-Length: 0 Content-Length: 0
F7 ACM NGW 1 -> Alice F7 ACM NGW 1 -> Alice
ACM ACM
/* Alice hangs up */ /* Alice hangs up */
skipping to change at page 99, line 4 skipping to change at page 95, line 9
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL CSeq: 1 CANCEL
Content-Length: 0 Content-Length: 0
F12 CANCEL Proxy 1 -> Bob F12 CANCEL Proxy 1 -> Bob
SIP PSTN Call Flows April 2003
CANCEL sip:bob@b.example.com SIP/2.0 CANCEL sip:bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone> To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL CSeq: 1 CANCEL
Content-Length: 0 Content-Length: 0
skipping to change at page 100, line 4 skipping to change at page 96, line 11
ACK sip:bob@b.example.com SIP/2.0 ACK sip:bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
F16 487 Request Terminated Proxy 1 -> NGW 1 F16 487 Request Terminated Proxy 1 -> NGW 1
SIP PSTN Call Flows April 2003
SIP/2.0 487 Request Terminated SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F17 ACK NGW 1 -> Proxy 1 F17 ACK NGW 1 -> Proxy 1
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2 Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159 To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
SIP PSTN Call Flows April 2003
4. PSTN to PSTN Dialing via SIP Network 4. PSTN to PSTN Dialing via SIP Network
In these scenarios, both the caller and the called party are in the In these scenarios, both the caller and the called party are in the
telephone network, either normal PSTN subscribers or PBX extensions. telephone network, either normal PSTN subscribers or PBX extensions.
The calls route through two Gateways and at least one SIP Proxy The calls route through two Gateways and at least one SIP Proxy
Server. The Proxy Server performs the authentication and location of Server. The Proxy Server performs the authentication and location of
the Gateways. the Gateways.
Again it is noted that the intent of this call flows document is not 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 to provide a detailed parameter level mapping of SIP to PSTN
protocols. For information on SIP to ISUP mapping, the reader is protocols. For information on SIP to ISUP mapping, the reader is
referred to other references [4]. referred to other references [4].
In these scenarios, the call is successfully completed between the In these scenarios, the call is successfully completed between the
two Gateways allowing the PSTN or PBX users to communicate. The 183 two Gateways, allowing the PSTN or PBX users to communicate. The 183
Session Progress response is used to indicate in-band alerting may Session Progress response is used to indicate that in-band alerting
flow from the called party telephone switch to the caller. may flow from the called party telephone switch to the caller.
SIP PSTN Call Flows April 2003
4.1 Successful ISUP PSTN to ISUP PSTN call 4.1. Successful ISUP PSTN to ISUP PSTN call
Switch A NGW 1 Proxy 1 GW 2 Switch C Switch A NGW 1 Proxy 1 GW 2 Switch C
| | | | | | | | | |
| IAM F1 | | | | | IAM F1 | | | |
|------------->| | | | |------------->| | | |
| | INVITE F2 | | | | | INVITE F2 | | |
| |------------->| INVITE F3 | | | |------------->| INVITE F3 | |
| | |------------->| IAM F4 | | | |------------->| IAM F4 |
| | | |------------->| | | | |------------->|
| | | | ACM F5 | | | | | ACM F5 |
skipping to change at page 102, line 51 skipping to change at page 97, line 49
| |------------->| 200 F20 | | | |------------->| 200 F20 | |
| | |------------->| | | | |------------->| |
| REL F21 | | | | | REL F21 | | | |
|<-------------| | | | |<-------------| | | |
| RLC F22 | | | | | RLC F22 | | | |
|------------->| | | | |------------->| | | |
| | | | | | | | | |
In this scenario, Alice in the PSTN calls Carol who is an extension 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 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 NGW 1, while Carol's PBX signals via SS7 with the Gateway GW
Gateway GW 2. The CdPN and CgPN are mapped by GW1 into SIP URIs and 2. The CdPN and CgPN are mapped by GW 1 into SIP URIs and placed in
placed in the To and From headers. Proxy 1 looks up the dialed the To and From headers. Proxy 1 looks up the dialed digits in the
digits in the Request-URI and maps the digits to the PBX extension of Request-URI and maps the digits to the PBX extension of Carol, which
SIP PSTN Call Flows April 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
Carol which is served by GW 2. The Proxy in F3 uses the host portion referenced. The INVITE is then forwarded to GW 2 for call
of the Request-URI to identify what private dialing plan is being completion. An early media path is established end-to-end so that
referenced. The INVITE is then forwarded to GW 2 for call completion. Alice can hear the ringing tone generated by PBX C.
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 Carol answers the call and the media path is cut through in both
directions. Bob hangs up terminating the call. directions. Bob hangs up terminating the call.
Message Details Message Details
F1 IAM Switch Alice -> NGW 1 F1 IAM Switch Alice -> NGW 1
IAM IAM
CgPN=314-555-1111,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National
skipping to change at page 104, line 4 skipping to change at page 99, line 4
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
/* Proxy 1 consults Location Service and translates the dialed number /* Proxy 1 consults Location Service and translates the dialed number
to a private number in the Request-URI*/ to a private number in the Request-URI*/
F3 INVITE Proxy 1 -> GW 2 F3 INVITE Proxy 1 -> GW 2
INVITE sips:4443333@gw2.a.example.com SIP/2.0 INVITE sips:4443333@gw2.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKwqwee65 Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKwqwee65
SIP PSTN Call Flows April 2003
;received=192.0.2.103 ;received=192.0.2.103
Max-Forwards: 69 Max-Forwards: 69
Record-Route: <sips:ss1.a.example.com;lr> Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone> To: <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sips:ngw1@a.example.com> Contact: <sips:ngw1@a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 146 Content-Length: 146
skipping to change at page 105, line 4 skipping to change at page 100, line 5
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr> Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sips:4443333@gw2.a.example.com> Contact: <sips:4443333@gw2.a.example.com>
Content-Type: application/sdp Content-Type: application/sdp
SIP PSTN Call Flows April 2003
Content-Length: 143 Content-Length: 143
v=0 v=0
o=GW 987654321 987654321 IN IP4 gw2.a.example.com o=GW 987654321 987654321 IN IP4 gw2.a.example.com
s=- s=-
c=IN IP4 gw2.a.example.com c=IN IP4 gw2.a.example.com
t=0 0 t=0 0
m=audio 14918 RTP/AVP 0 m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
skipping to change at page 105, line 47 skipping to change at page 101, line 4
/* NGW 1 receives packets from GW 2 with encoded ringback, tones or /* NGW 1 receives packets from GW 2 with encoded ringback, tones or
other audio. NGW 1 decodes this and places it on the originating other audio. NGW 1 decodes this and places it on the originating
trunk. */ trunk. */
F8 ACM NGW 1 -> Switch A F8 ACM NGW 1 -> Switch A
ACM ACM
/* Bob answers */ /* Bob answers */
F9 ANM Switch C -> GW 2 F9 ANM Switch C -> GW 2
ANM ANM
SIP PSTN Call Flows April 2003
F10 200 OK GW 2 -> Proxy 1 F10 200 OK GW 2 -> Proxy 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103 ;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr> Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
skipping to change at page 107, line 4 skipping to change at page 102, line 7
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 143 Content-Length: 143
v=0 v=0
o=GW 987654321 987654321 IN IP4 gw2.a.example.com o=GW 987654321 987654321 IN IP4 gw2.a.example.com
s=- s=-
c=IN IP4 gw2.a.example.com c=IN IP4 gw2.a.example.com
t=0 0 t=0 0
m=audio 14918 RTP/AVP 0 m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
SIP PSTN Call Flows April 2003
F12 ANM NGW 1 -> Switch A F12 ANM NGW 1 -> Switch A
ANM ANM
F13 ACK NGW 1 -> Proxy 1 F13 ACK NGW 1 -> Proxy 1
ACK sips:4443333@gw2.a.example.com SIP/2.0 ACK sips:4443333@gw2.a.example.com SIP/2.0
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2 Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
Max-Forwards: 70 Max-Forwards: 70
skipping to change at page 107, line 43 skipping to change at page 103, line 4
Content-Length: 0 Content-Length: 0
/* RTP streams are established between NGW 1 and GW 2. */ /* RTP streams are established between NGW 1 and GW 2. */
/* Bob Hangs Up with Alice. */ /* Bob Hangs Up with Alice. */
F15 REL Switch C -> GW 2 F15 REL Switch C -> GW 2
REL REL
CauseCode=16 Normal CauseCode=16 Normal
F16 BYE GW 2 -> Proxy 1 F16 BYE GW 2 -> Proxy 1
BYE sips:ngw1@a.example.com SIP/2.0 BYE sips:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6 Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
Max-Forwards: 70 Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr> Route: <sips:ss1.a.example.com;lr>
SIP PSTN Call Flows April 2003
From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 4 BYE CSeq: 4 BYE
Content-Length: 0 Content-Length: 0
F17 RLC GW 2 -> Switch C F17 RLC GW 2 -> Switch C
RLC RLC
skipping to change at page 108, line 41 skipping to change at page 104, line 4
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1 Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6 Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202 ;received=192.0.2.202
From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 4 BYE CSeq: 4 BYE
Content-Length: 0 Content-Length: 0
F20 200 OK Proxy 1 -> GW 2 F20 200 OK Proxy 1 -> GW 2
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6 Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202 ;received=192.0.2.202
From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159 From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 4 BYE CSeq: 4 BYE
SIP PSTN Call Flows April 2003
Content-Length: 0 Content-Length: 0
F21 REL Switch C -> GW 2 F21 REL Switch C -> GW 2
REL REL
CauseCode=16 Normal CauseCode=16 Normal
F22 RLC GW 2 -> Switch C F22 RLC GW 2 -> Switch C
RLC RLC
SIP PSTN Call Flows April 2003
4.2 Successful FGB PBX to ISDN PBX call with overflow 4.2. Successful FGB PBX to ISDN PBX call with overflow
PBX A GW 1 Proxy 1 GW 2 GW 3 PBX C PBX A GW 1 Proxy 1 GW 2 GW 3 PBX C
| | | | | | | | | | | |
| Seizure | | | | | | Seizure | | | | |
|----------->| | | | | |----------->| | | | |
| Wink | | | | | | Wink | | | | |
|<-----------| | | | | |<-----------| | | | |
|MF Digits F1| | | | | |MF Digits F1| | | | |
|----------->| | | | | |----------->| | | | |
| | INVITE F2 | | | | | | INVITE F2 | | | |
skipping to change at page 111, line 4 skipping to change at page 106, line 4
| | | | DISC F19 | | | | | DISC F19 |
| | | |<-----------| | | | |<-----------|
| | | BYE F20 | | | | | BYE F20 | |
| | BYE F21 |<------------------------| REL F22 | | | BYE F21 |<------------------------| REL F22 |
|Seiz Removal|<-----------| |----------->| |Seiz Removal|<-----------| |----------->|
|<-----------| 200 F23 | | | |<-----------| 200 F23 | | |
|Seiz Removal|----------->| 200 F24 | | |Seiz Removal|----------->| 200 F24 | |
|----------->| |------------------------>| REL COM F25| |----------->| |------------------------>| REL COM F25|
| | | |<-----------| | | | |<-----------|
| | | | | | | | | |
SIP PSTN Call Flows April 2003
PBX Alice calls PBX Carol via Gateway GW 1 and Proxy 1. During the 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 attempt to reach Carol via GW 2, an error is encountered - Proxy 1
receives a 503 Service Unavailable (F4) response to the forwarded receives a 503 Service Unavailable (F4) response to the forwarded
INVITE. This could be due to all circuits being busy, or some other 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 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 route via GW 3 to terminate the call. From there, the call proceeds
normally with Carol answering the call. The call is terminated when normally with Carol answering the call. The call is terminated when
Carol hangs up. Carol hangs up.
Message Details Message Details
skipping to change at page 112, line 4 skipping to change at page 107, line 4
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 155 Content-Length: 155
v=0 v=0
o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com
s=- s=-
c=IN IP4 gw1.a.example.com c=IN IP4 gw1.a.example.com
t=0 0 t=0 0
m=audio 49172 RTP/AVP 0 m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
SIP PSTN Call Flows April 2003
/* Proxy 1 uses a Location Service function to determine where B is /* Proxy 1 uses a Location Service function to determine where B is
located. Response is returned listing alternative routes, GW2 and located. Response is returned listing alternative routes, GW2 and
GW3, which are then tried sequentially. */ GW3, which are then tried sequentially. */
F3 INVITE Proxy 1 -> GW 2 F3 INVITE Proxy 1 -> GW 2
INVITE sip:4443333@gw2.a.example.com SIP/2.0 INVITE sip:4443333@gw2.a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
skipping to change at page 113, line 4 skipping to change at page 108, line 4
From: <sip:551313@gw1.a.example.com>;tag=63412s From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=314159 To: <sip:4443333@ss1.a.example.com>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
F5 ACK Proxy 1 -> GW 2 F5 ACK Proxy 1 -> GW 2
ACK sip:4443333@ss1.a.example.com SIP/2.0 ACK sip:4443333@ss1.a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
SIP PSTN Call Flows April 2003
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Max-Forward: 70 Max-Forward: 70
From: <sip:551313@gw1.a.example.com>;tag=63412s From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=314159 To: <sip:4443333@ss1.a.example.com>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
F6 INVITE Proxy 1 -> GW 3 F6 INVITE Proxy 1 -> GW 3
skipping to change at page 114, line 4 skipping to change at page 109, line 4
Message type=SETUP Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16 Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio) (3.1 kHz audio)
Channel identification=Preferred or exclusive B-channel Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN; further call Progress indicator=1 (Call is not end-to-end ISDN; further call
progress information may be available inband) progress information may be available inband)
Called party number: Called party number:
Type of number and numbering plan ID=33 (National number in ISDN Type of number and numbering plan ID=33 (National number in ISDN
numbering plan) numbering plan)
Digits=918-555-3333 Digits=918-555-3333
SIP PSTN Call Flows April 2003
F8 100 Trying GW 3 -> Proxy 1 F8 100 Trying GW 3 -> Proxy 1
SIP/2.0 100 Trying SIP/2.0 100 Trying
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
From: <sip:551313@gw1.a.example.com>;tag=63412s From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com> To: <sip:4443333@ss1.a.example.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Content-Length: 0 Content-Length: 0
skipping to change at page 115, line 4 skipping to change at page 110, line 6
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.a.example.com;user=phone> Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
Content-Length: 0 Content-Length: 0
F12 180 Ringing Proxy 1 -> GW 1 F12 180 Ringing Proxy 1 -> GW 1
SIP/2.0 180 Ringing SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
SIP PSTN Call Flows April 2003
From: <sip:551313@gw1.a.example.com>;tag=63412s From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789 To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.a.example.com;user=phone> Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
Content-Length: 0 Content-Length: 0
F13 CONNect PBX C -> GW 3 F13 CONNect PBX C -> GW 3
Protocol discriminator=Q.931 Protocol discriminator=Q.931
skipping to change at page 116, line 4 skipping to change at page 111, line 7
F15 200 OK Proxy 1 -> GW 1 F15 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr> Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789 To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE CSeq: 1 INVITE
SIP PSTN Call Flows April 2003
Contact: <sip:+19185553333@gw3.a.example.com;user=phone> Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 143 Content-Length: 143
v=0 v=0
o=GW 987654321 987654321 IN IP4 gw3.a.example.com o=GW 987654321 987654321 IN IP4 gw3.a.example.com
s=- s=-
c=IN IP4 gw3.a.example.com c=IN IP4 gw3.a.example.com
t=0 0 t=0 0
m=audio 14918 RTP/AVP 0 m=audio 14918 RTP/AVP 0
skipping to change at page 116, line 46 skipping to change at page 112, line 4
ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0 ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65 Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201 ;received=192.0.2.201
Max-Forwards: 69 Max-Forwards: 69
From: <sip:551313@gw1.a.example.com>;tag=63412s From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789 To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 ACK CSeq: 1 ACK
Content-Length: 0 Content-Length: 0
F18 CONNect ACK GW 3 -> PBX C F18 CONNect ACK GW 3 -> PBX C
Protocol discriminator=Q.931 Protocol discriminator=Q.931
Message type=CONN ACK Message type=CONN ACK
SIP PSTN Call Flows April 2003
/* RTP streams are established between GW 1 and GW 3. */ /* RTP streams are established between GW 1 and GW 3. */
/* Bob Hangs Up with Alice. */ /* Bob Hangs Up with Alice. */
F19 DISConnect PBX C -> GW 3 F19 DISConnect PBX C -> GW 3
Protocol discriminator=Q.931 Protocol discriminator=Q.931
Message type=DISC Message type=DISC
Cause=16 (Normal clearing) Cause=16 (Normal clearing)
skipping to change at page 117, line 44 skipping to change at page 113, line 4
Max-Forwards: 69 Max-Forwards: 69
From: <sip:4443333@ss1.a.example.com>;tag=123456789 From: <sip:4443333@ss1.a.example.com>;tag=123456789
To: <sip:551313@gw1.a.example.com>;tag=63412s To: <sip:551313@gw1.a.example.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 BYE CSeq: 1 BYE
Content-Length: 0 Content-Length: 0
GW 1 -> PBX A GW 1 -> PBX A
Seizure removal Seizure removal
F22 RELease GW 3 -> PBX C F22 RELease GW 3 -> PBX C
Protocol discriminator=Q.931 Protocol discriminator=Q.931
Message type=REL Message type=REL
SIP PSTN Call Flows April 2003
F23 200 OK GW 1 -> Proxy 1 F23 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2 Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111 ;received=192.0.2.111
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
;received=192.0.2.203 ;received=192.0.2.203
From: <sip:4443333@ss1.a.example.com>;tag=123456789 From: <sip:4443333@ss1.a.example.com>;tag=123456789
To: <sip:551313@gw1.a.example.com>;tag=63412s To: <sip:551313@gw1.a.example.com>;tag=63412s
skipping to change at page 118, line 39 skipping to change at page 113, line 42
F25 RELease COMplete PBX C -> GW 3 F25 RELease COMplete PBX C -> GW 3
Protocol discriminator=Q.931 Protocol discriminator=Q.931
Message type=REL COM Message type=REL COM
PBX Alice -> GW 1 PBX Alice -> GW 1
Seizure removal Seizure removal
Security Considerations 5. Security Considerations
This document provides examples of mapping from SIP to ISUP and ISUP This document provides examples of mapping from SIP to ISUP and ISUP
to SIP. The gateways in these examples are compliant with the to SIP. The gateways in these examples are compliant with the
Security Considerations Section of RFC zzzz [4] which is summarized Security Considerations Section of RFC 3398 [4] which is summarized
here. here.
SIP PSTN Call Flows April 2003
There are few security concerns relating to the mapping of ISUP to There are few security concerns relating to the mapping of ISUP to
SIP besides privacy considerations in the calling party number SIP besides privacy considerations in the calling party number
passing. Some concerns relating to the mapping from tel URI passing. Some concerns relating to the mapping from tel URI
parameters to ISUP including the user creation of parameters and parameters to ISUP include the user creation of parameters and codes
codes relating to called number and local number portability (LNP). relating to called number and local number portability (LNP). An
An operator of a gateway should use policies similar to those present operator of a gateway should use policies similar to those present in
in PSTN switches to avoid security problems. PSTN switches to avoid security problems.
The mapping from a SIP response code to an ISUP Cause Code presents a The mapping from a SIP response code to an ISUP Cause Code presents a
theoretical risk, so a gateway operator may implement policies theoretical risk, so a gateway operator may implement policies
controlling this mapping. Gateways should also not rely on the controlling this mapping. Gateways should also not rely on the
contents of the From header field for identity information, as it may contents of the From header field for identity information, as it may
be arbitrarily populated by a user. Instead, some sort of be arbitrarily populated by a user. Instead, some sort of
cryptographic authentication and authorization should be used for cryptographic authentication and authorization should be used for
identity determination. These flows show both HTTP Digest for identity determination. These flows show both HTTP Digest for
authentication of users, although for brevity the challenge is not authentication of users, although for brevity, the challenge is not
always shown. always shown.
The early media cut-through shown in some flows is another potential The early media cut-through shown in some flows is another potential
security risk, but it is also required for proper interaction with security risk, but it is also required for proper interaction with
the PSTN. Again, a gateway operator should use proper policies the PSTN. Again, a gateway operator should use proper policies
relating to early media to prevent fraud and misuse. Finally, a user relating to early media to prevent fraud and misuse. Finally, a user
agent (even a properly authenticated one) can launch multiple agent (even a properly authenticated one) can launch multiple
simultaneous requests through a gateway, constituting a denial of simultaneous requests through a gateway, constituting a denial of
service attack. The adoption of policies to limit the number of service attack. The adoption of policies to limit the number of
simultaneous requests from a single entity may be used to prevent simultaneous requests from a single entity may be used to prevent
this attack. this attack.
As discussed in the SIP-T framework [8] SIP/ISUP interworking can be As discussed in the SIP-T framework [7], SIP/ISUP interworking can be
employed as an interdomain signaling mechanism that may be subject to employed as an interdomain signaling mechanism that may be subject to
pre-existing trust relationships between administrative domains. Any pre-existing trust relationships between administrative domains. Any
administrative domain implementing SIP-T or SIP/ISUP interworking administrative domain implementing SIP-T or SIP/ISUP interworking
should have an adequate security apparatus (including elements that should have an adequate security apparatus (including elements that
manage any appropriate policies to manage fraud and billing in an manage any appropriate policies to manage fraud and billing in an
interdomain environment) in place to ensure that the translation of interdomain environment) in place to ensure that the translation of
ISUP information does not result in any security violations. ISUP information does not result in any security violations.
Although no examples of this are shown in this document, transporting Although no examples of this are shown in this document, transporting
ISUP in SIP bodies may provide opportunities for abuse, fraud, and ISUP in SIP bodies may provide opportunities for abuse, fraud, and
privacy concerns, especially when SIP-T requests can be generated, privacy concerns, especially when SIP-T requests can be generated,
inspected or modified by arbitrary SIP endpoints. ISUP MIME bodies inspected or modified by arbitrary SIP endpoints. ISUP MIME bodies
should be secured (preferably with S/MIME as detailed in RFC 3261 should be secured (preferably with S/MIME as detailed in RFC 3261
[2]) to alleviate this concern. Authentication properties provided by [2]) to alleviate this concern. Authentication properties provided
S/MIME would allow the recipient of a SIP-T message to ensure that 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 the ISUP MIME body was generated by an authorized entity. Encryption
would ensure that only carriers possessing a particular decryption would ensure that only carriers possessing a particular decryption
key are capable of inspecting encapsulated ISUP MIME bodies in a SIP key are capable of inspecting encapsulated ISUP MIME bodies in a SIP
request. request.
SIP PSTN Call Flows April 2003 6. References
Normative References 6.1. Normative References
1 Bradner, S., "Key words for use in RFCs to Indicate Requirement [1] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997 Levels", BCP 14, RFC 2119, March 1997.
2 Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., [2] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M., and Schooler, E., "SIP: Peterson, J., Sparks, R., Handley, M. E. and Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002. Session Initiation Protocol", RFC 3261, June 2002.
3 Rosenberg, J. and Schulzrinne, H., "An Offer/Answer Model with [3] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
SDP", Internet Engineering Task Force, RFC 3264, April 2002. the Session Description Protocol (SDP)", RFC 3264, June 2002.
4 G. Camarillo, A. Roach, J. Peterson, L. Ong, "ISUP to SIP
Mapping", Internet Draft, Internet Engineering Task Force, Work in
progress. August 2002.
5 Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach,
P., Luotonen, A. and L. Stewart, "HTTP authentication: Basic and
Digest Access Authentication", RFC 2617, June 1999.
6 J. Rosenberg, H. Schulzrinne, and G. Camarillo, "The Stream [4] Camarillo, G., Roach, A. B., Peterson, J. and L. Ong,
Control Transmission Protocol as a Transport for the Session "Integrated Services Digital Network (ISDN) User Part (ISUP) to
Initiation Protocol," Internet Draft, Internet Engineering Task Session Initiation Protocol (SIP) Mapping", RFC 3398, December
Force, Work in progress. June 2002. 2002.
7 A. Vaha-Sipila, "URLs for Telephone Calls", Internet Draft, [5] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Internet Engineering Task Force, RFC 2806, April 2000. Leach, P., Luotonen, A. and L. Stewart, "HTTP Authentication:
Basic and Digest Access Authentication", RFC 2617, June 1999.
8 A. Vemuri and J. Peterson, "Session Initiation Protocol for [6] Vaha-Sipila, A., "URLs for Telephone Calls", RFC 2806, April
Telephones (SIP-T): Context and Architectures," RFC 3372, 2000.
September 2002.
9 E. Zimmerer, J. Peterson, A. Vemuri, L. Ong, F. Audet, M. Watson, [7] Vemuri, A. and J. Peterson, "Session Initiation Protocol for
M. Zonoun, "MIME media types for ISUP and QSIG Objects," RFC 3204, Telephones (SIP-T): Context and Architectures", BCP 63, RFC
December 2001. 3372, September 2002.
10 P. Faltstrom, "E.164 Numbers and DNS," RFC 2916, September 2000. [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.
Informative References [9] Faltstrom, P., "E.164 number and DNS", RFC 2916, September 2000.
11 Johnston, A., Donovan, S., Sparks, R., Cunningham, C., Summers, 6.2. Informative References
K., "Session Initiation Protocol Basic Call Flow Examples", RFC
yyyv, August 2002.
SIP PSTN Call Flows April 2003 [10] Johnston, A., Donovan, S., Sparks, R., Cunningham, C. and K.
Summers, "Session Initiation Protocol (SIP) Basic Call Flow
Examples", RFC 3665, December 2003.
Acknowledgments 7. Acknowledgments
Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings, Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings,
and Tom Taylor for their detailed comments during the final review. and Tom Taylor for their detailed comments during the final review.
Thanks to Dean Willis for his early contributions to the development Thanks to Dean Willis for his early contributions to the development
of this document. Thanks to Jon Peterson for his help on the of this document. Thanks to Jon Peterson for his help on the
security section. security section.
The authors wish to thank Kundan Singh for performing parser The authors wish to thank Kundan Singh for performing parser
validation of messages. validation of messages.
skipping to change at page 121, line 32 skipping to change at page 116, line 30
The authors also wish to thank the following individuals for their The authors also wish to thank the following individuals for their
assistance: Jean-Francois Mule, Hemant Agrawal, Henry Sinnreich, assistance: Jean-Francois Mule, Hemant Agrawal, Henry Sinnreich,
David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole
MCI WorldCom IPOP Design team, Scott Orton, Greg Osterhout, Pat MCI WorldCom IPOP Design team, Scott Orton, Greg Osterhout, Pat
Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise
Ingram, Denise Caballero, Tom Redman, Ilya Slain, Pat Sollee, John Ingram, Denise Caballero, Tom Redman, Ilya Slain, Pat Sollee, John
Truetken, and others from MCI WorldCom, 3Com, Cisco, Lucent and Truetken, and others from MCI WorldCom, 3Com, Cisco, Lucent and
Nortel. Nortel.
Author's Addresses 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.
9. Authors' Addresses
All listed authors actively contributed large amounts of text to this All listed authors actively contributed large amounts of text to this
document. document.
Alan Johnston Alan Johnston
WorldCom MCI
100 South 4th Street 100 South 4th Street
St. Louis, MO 63102 St. Louis, MO 63102
USA USA
EMail: alan.johnston@wcom.com EMail: alan.johnston@mci.com
Steve Donovan Steve Donovan
dynamicsoft, Inc. dynamicsoft, Inc.
5100 Tennyson Parkway 5100 Tennyson Parkway
Suite 1200 Suite 1200
Plano, Texas 75024 Plano, Texas 75024
USA USA
EMail: sdonovan@dynamicsoft.com EMail: sdonovan@dynamicsoft.com
SIP PSTN Call Flows April 2003
Robert Sparks Robert Sparks
dynamicsoft, Inc. dynamicsoft, Inc.
5100 Tennyson Parkway 5100 Tennyson Parkway
Suite 1200 Suite 1200
Plano, Texas 75024 Plano, Texas 75024
USA USA
EMail: rsparks@dynamicsoft.com EMail: rsparks@dynamicsoft.com
skipping to change at page 122, line 30 skipping to change at page 117, line 51
USA USA
EMail: ccunningham@dynamicsoft.com EMail: ccunningham@dynamicsoft.com
Kevin Summers Kevin Summers
Sonus Sonus
1701 North Collins Blvd, Suite 3000 1701 North Collins Blvd, Suite 3000
Richardson, TX 75080 Richardson, TX 75080
USA USA
Email: kevin.summers@sonusnet.com EMail: kevin.summers@sonusnet.com
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.
SIP PSTN Call Flows April 2003
Full Copyright Statement 10. Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved. Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing document itself may not be modified in any way, such as by removing
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