RFC 3666
Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows
Network Working Group A. Johnston Request for Comments: 3666 MCI BCP: 76 S. Donovan Category: Best Current Practice R. Sparks C. Cunningham dynamicsoft K. Summers Sonus December 2003 Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows Status of this Memo This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The Internet Society (2003). All Rights Reserved.Abstract
This document contains best current practice examples of Session Initiation Protocol (SIP) call flows showing interworking with the Public Switched Telephone Network (PSTN). Elements in these call flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways. Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP. PSTN telephony protocols are illustrated using ISDN (Integrated Services Digital Network), ISUP (ISDN User Part), and FGB (Feature Group B) circuit associated signaling. PSTN calls are illustrated using global telephone numbers from the PSTN and private extensions served on by a PBX (Private Branch Exchange). Call flow diagrams and message details are shown.
Table of Contents
1. Overview..................................................... 2 1.1. General Assumptions.................................... 3 1.2. Legend for Message Flows............................... 4 1.3. SIP Protocol Assumptions............................... 5 2. SIP to PSTN Dialing.......................................... 6 2.1. Successful SIP to ISUP PSTN call....................... 7 2.2. Successful SIP to ISDN PBX call........................ 15 2.3. Successful SIP to ISUP PSTN call with overflow......... 23 2.4. Successful SIP to SIP using ENUM Query................. 32 2.5. Unsuccessful SIP to PSTN call: Treatment from PSTN..... 38 2.6. Unsuccessful SIP to PSTN: REL w/Cause from PSTN........ 45 2.7. Unsuccessful SIP to PSTN: ANM Timeout.................. 49 3. PSTN to SIP Dialing.......................................... 54 3.1. Successful PSTN to SIP call............................ 55 3.2. Successful PSTN to SIP call, Fast Answer............... 62 3.3. Successful PBX to SIP call............................. 68 3.4. Unsuccessful PSTN to SIP REL, SIP error mapped to REL.. 74 3.5. Unsuccessful PSTN to SIP REL, SIP busy mapped to REL... 76 3.6. Unsuccessful PSTN->SIP, SIP error interworking to tones 80 3.7. Unsuccessful PSTN->SIP, ACM timeout.................... 84 3.8. Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy... 88 3.9. Unsuccessful PSTN->SIP, Caller Abandonment............. 91 4. PSTN to PSTN Dialing via SIP Network......................... 96 4.1. Successful ISUP PSTN to ISUP PSTN call................. 97 4.2. Successful FGB PBX to ISDN PBX call with overflow...... 105 5. Security Considerations...................................... 113 6. References................................................... 115 6.1. Normative References................................... 115 6.2. Informative References................................. 115 7. Acknowledgments.............................................. 116 8. Intellectual Property Statement.............................. 116 9. Authors' Addresses........................................... 117 10. Full Copyright Statement..................................... 1181. Overview
The call flows shown in this document were developed in the design of a SIP IP communications network. They represent an example of a minimum set of functionality. It is the hope of the authors that this document will be useful for SIP implementers, designers, and protocol researchers alike and will help further the goal of a standard implementation of RFC 3261 [2]. These flows represent carefully checked and working group reviewed scenarios of the most common SIP/PSTN interworking examples as a companion to the specifications.
These call flows are based on the current version 2.0 of SIP in RFC 3261 [2] with SDP usage described in RFC 3264 [3]. Other RFCs also comprise the SIP standard but are not used in this set of basic call flows. The SIP/ISUP mapping is based on RFC 3398 [4]. Various PSTN signaling protocols are illustrated in this document: ISDN (Integrated Services Digital Network), ISUP (ISDN User Part) and FGB (Feature Group B) circuit associated signaling. This document shows mainly ANSI ISUP due to its practical origins. However, as used in this document, the usage is virtually identical to the ITU-T International ISUP used as the reference in [4]. Basic SIP call flow examples are contained in a companion document, RFC 3665 [10]. The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14, RFC 2119 [1].1.1. General Assumptions
A number of architecture, network, and protocol assumptions underlie the call flows in this document. Note that these assumptions are not requirements. They are outlined in this section so that they may be taken into consideration and to aid in the understanding of the call flow examples. The authentication of SIP User Agents in these example call flows is performed using HTTP Digest as defined in [3] and [5]. Some Proxy Servers in these call flows insert Record-Route headers into requests to ensure that they are in the signaling path for future message exchanges. These flows show TLS, TCP, and UDP for transport. SCTP could also be used. See the discussion in RFC 3261 [2] for details on the transport issues for SIP. The SIP Proxy Server has access to a Location Service and other databases. Information present in the Request-URI and the context (From header) is sufficient to determine to which proxy or gateway the message should be routed. In most cases, a primary and secondary route will be determined in case of a Proxy or Gateway failure downstream.
Gateways provide tones (ringing, busy, etc) and announcements to the
PSTN side based on SIP response messages, or pass along audio in-band
tones (ringing, busy tone, etc.) in an early media stream to the SIP
side.
The interactions between the Proxy and Gateway can be summarized as
follows:
- The SIP Proxy Server performs digit analysis and lookup and
locates the correct gateway.
- The SIP Proxy Server performs gateway location based on primary
and secondary routing.
Telephone numbers are usually represented as SIP URIs. Note that an
alternative is the use of the tel URI [6].
This document shows typical examples of SIP/ISUP interworking.
Although in the spirit of the SIP-T framework [7], these examples do
not represent a complete implementation of the framework. The
examples here represent more of a minimal set of examples for very
basic SIP to ISUP interworking, rather than the more complex goal of
ISUP transparency. In particular, there are NO examples of
encapsulated ISUP in this document. If present, these messages would
show S/MIME encryption due to the sensitive nature of this
information, as discussed in the SIP-T Framework security
considerations section. (Note - RFC 3204 [8] contains an example of
an INVITE with encapsulated ISUP.) See the Security Considerations
section for a more detailed discussion on the security of these call
flows.
In ISUP, the Calling Party Number is abbreviated as CgPN and the
Called Party Number is abbreviated as CdPN. Other abbreviations
include Numbering Plan Indicator (NPI) and Nature of Address (NOA).
1.2. Legend for Message Flows
Dashed lines (---) represent signaling messages that are mandatory to
the call scenario. These messages can be SIP or PSTN signaling. The
arrow indicates the direction of message flow.
Double dashed lines (===) represent media paths between network
elements.
Messages with parentheses around their name represent optional
messages.
Messages are identified in the Figures as F1, F2, etc. This references the message details in the list that follows the Figure. Comments in the message details are shown in the following form: /* Comments. */1.3. SIP Protocol Assumptions
This document does not prescribe the flows precisely as they are shown, but rather the flows illustrate the principles for best practice. They are best practices usages (orderings, syntax, selection of features for the purpose, handling of error) of SIP methods, headers and parameters. IMPORTANT: The exact flows here must not be copied as is by an implementer due to specific incorrect characteristics that were introduced into the document for convenience and are listed below. To sum up, the SIP/PSTN call flows represent well-reviewed examples of SIP usage, which are best common practice according to IETF consensus. For simplicity in reading and editing the document, there are a number of differences between some of the examples and actual SIP messages. For example, the SIP Digest responses are not actual MD5 encodings. Call-IDs are often repeated, and CSeq counts often begin at 1. Header fields are usually shown in the same order. Usually only the minimum required header field set is shown, others that would normally be present, such as Accept, Supported, Allow, etc. are not shown. Actors: Element Display Name URI IP Address ------- ------------ --- ---------- User Agent Alice sip:alice@a.example.com 192.0.2.101 User Agent Bob sip:bob@b.example.com 192.0.2.200 Proxy Server sip:ss1.a.example.com 192.0.2.111 User Agent (Gateway) sip:gw1.a.example.com 192.0.2.201 User Agent (Gateway) sip:gw2.a.example.com 192.0.2.202 User Agent (Gateway) sip:gw3.a.example.com 192.0.2.203 User Agent (Gateway) sip:ngw1.a.example.com 192.0.2.103 User Agent (Gateway) sip:ngw2.a.example.com 192.0.2.102 Note that NGW 1 and NGW 2 also have device URIs (Contacts) of sip:ngw1@a.example.com and sip:ngw2@a.example.com which resolve to the Proxy Server sip:ss1.wcom.com using DNS SRV records.
2. SIP to PSTN Dialing
In the following scenarios, Alice (sip:alice@a.example.com) is a SIP phone or other SIP-enabled device. Bob is reachable via the PSTN at global telephone number +19725552222. Alice places a call to Bob through a Proxy Server, Proxy 1, and a Network Gateway. In other scenarios, Alice places calls to Carol, who is served via a PBX (Private Branch Exchange) and is identified by a private extension 444-3333, or global number +1-918-555-3333. Note that Alice uses his/her global telephone number +1-314-555-1111 in the From header in the INVITE messages. This then gives the Gateway the option of using this header to populate the calling party identification field in subsequent signaling. Left open is the issue of how the Gateway can determine the accuracy of the telephone number which is necessary before passing it as a valid calling party number in the PSTN. In these scenarios, Alice is a SIP phone or other SIP-enabled device. Alice places a call to Bob in the PSTN or Carol on a PBX through a Proxy Server and a Gateway. In the failure scenarios, the call does not complete. In some cases however, a media stream is still setup. This is due to the fact that some failures in dialing to the PSTN result in in-band tones (busy, reorder tones or announcements - "The number you have dialed has changed. The new number is..."). The 183 Session Progress response containing SDP media information is used to setup this early media path so that the caller Alice knows the final disposition of the call. The media stream is either terminated by the caller after the tone or announcement has been heard and understood, or by the Gateway after a timer expires. In other failure scenarios, a SS7 Release with Cause Code is mapped to a SIP response. In these scenarios, the early media path is not used, but the actual failure code is conveyed to the caller by the SIP User Agent Client.
2.1. Successful SIP to ISUP PSTN call
Alice Proxy 1 NGW 1 Switch B | | | | | INVITE F1 | | | |--------------->| | | | 100 F2 | | | |<---------------| INVITE F3 | | | |--------------->| | | | 100 F4 | | | |<---------------| IAM F5 | | | |--------------->| | | | ACM F6 | | | 183 F7 |<---------------| | 183 F8 |<---------------| | |<---------------| | | | Both Way RTP Media | One Way Voice | |<===============================>|<===============| | | | ANM F9 | | | 200 F10 |<---------------| | 200 F11 |<---------------| | |<---------------| | | | ACK F12 | | | |--------------->| ACK F13 | | | |--------------->| | | Both Way RTP Media | Both Way Voice | |<===============================>|<==============>| | BYE F14 | | | |--------------->| BYE F15 | | | |--------------->| | | | 200 F16 | | | 200 F17 |<---------------| REL F18 | |<---------------| |--------------->| | | | RLC F19 | | | |<---------------| | | | | Alice dials the globalized E.164 number +19725552222 to reach Bob. Note that A might have only dialed the last 7 digits, or some other dialing plan. It is assumed that the SIP User Agent Client converts the digits into a global number and puts them into a SIP URI. Note that tel URIs could be used instead of SIP URIs. Alice could use either their SIP address (sip:alice@a.example.com) or SIP telephone number (sip:+13145551111@ss1.a.example.com;user=phone) in the From header. In this example, the telephone number is included, and it is shown as being passed as calling party identification through the Network Gateway (NGW 1) to Bob (F5). Note
that for this number to be passed into the SS7 network, it would have to be somehow verified for accuracy. In this scenario, Bob answers the call, then Alice disconnects the call. Signaling between NGW 1 and Bob's telephone switch is ANSI ISUP. For the details of SIP to ISUP mapping, refer to [4]. In this flow, notice that the Contact returned by NGW 1 in messages F7-11 is sip:ngw1@a.example.com. This is because NGW 1 only accepts SIP messages that come through Proxy 1 - any direct signaling will be ignored. Since this Contact URI may be used outside of this dialog and must be routable (Section 8.1.1.8 in RFC 3261 [2]) the Contact URI for NGW 1 must resolve to Proxy 1. This Contact URI resolves via DNS to Proxy 1 (sip:ss1.a.example.com) which then resolves it to sip:ngw1.a.example.com which is the address of NGW 1. This flow shows TCP transport. Message Details F1 INVITE Alice -> Proxy 1 INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0 Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9 Max-Forwards: 70 From: Alice <sip:+13145551111@ss1.a.example.com;user=phone> ;tag=9fxced76sl To: Bob <sip:+19725552222@ss1.a.example.com;user=phone> Call-ID: 2xTb9vxSit55XU7p8@a.example.com CSeq: 1 INVITE Contact: <sip:alice@client.a.example.com;transport=tcp> Proxy-Authorization: Digest username="alice", realm="a.example.com", nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone", response="ccdca50cb091d587421457305d097458c" Content-Type: application/sdp Content-Length: 154 v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> Alice
SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine the gateway
for terminating this call. The call is forwarded to NGW 1. Client
for A prepares to receive data on port 49172 from the
network.*/
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> Bob
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM Bob -> NGW 1
ACM
F7 183 Session Progress NGW 1 -> Proxy 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* NGW 1 sends PSTN audio (ringing) in the RTP path to A */
F8 183 Session Progress Proxy 1 -> Alice
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F9 ANM Bob -> NGW 1
ANM
F10 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F11 200 OK Proxy 1 -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F12 ACK Alice -> Proxy 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
F13 ACK Proxy 1 -> NGW 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
/* Alice Hangs Up with Bob. */
F14 BYE Alice -> Proxy 1
BYE sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F15 BYE Proxy 1 -> NGW 1
BYE sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F16 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F17 200 OK Proxy 1 -> A
SIP/2.0 200 OK
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F18 REL NGW 1 -> B
REL
CauseCode=16 Normal
F19 RLC B -> NGW 1
RLC
2.2. Successful SIP to ISDN PBX call
Alice Proxy 1 GW 1 PBX C | | | | | INVITE F1 | | | |--------------->| | | | 100 F2 | | | |<---------------| INVITE F3 | | | |--------------->| | | | 100 F4 | | | |<---------------| SETUP F5 | | | |--------------->| | | | CALL PROC F6 | | | |<---------------| | | | PROGress F7 | | | 180 F8 |<---------------| | 180 F9 |<---------------| | |<---------------| | | | | | One Way Voice | | | |<===============| | | | CONNect F10 | | | |<---------------| | | | CONNect ACK F11| | | 200 F12 |--------------->| | 200 F13 |<---------------| | |<---------------| | | | ACK F14 | | | |--------------->| ACK F15 | | | |--------------->| | | Both Way RTP Media | Both Way Voice | |<===============================>|<==============>| | BYE F16 | | | |--------------->| BYE F17 | | | |--------------->| | | | 200 F18 | | | 200 F19 |<---------------| DISConnect F20 | |<---------------| |--------------->| | | | RELease F21 | | | |<---------------| | | | RELease COM F22| | | |--------------->| | | | | Alice is a SIP device while Carol is connected via a Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk group. Alice dials Carol's telephone number (918-555-3333) which is globalized and put into a SIP URI.
The host portion of the Request-URI in the INVITE F3 is used to
identify the context (customer, trunk group, or line) in which the
private number 444-3333 is valid. Otherwise, this INVITE message
could get forwarded by GW 1 and the context of the digits could
become lost and the call unroutable.
Proxy 1 looks up the telephone number and locates the gateway that
serves Carol. Carol is identified by its extension (444-3333) in the
Request-URI sent to GW 1.
Note that the Contact URI for GW 1, as used in messages F8, F9, F12,
and F13, is sips:4443333@gw1.a.example.com, which resolves directly
to the gateway.
This flow shows the use of Secure SIP (sips) URIs.
Message Details
F1 INVITE Alice -> Proxy 1
INVITE sips:+19185553333@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:alice@client.a.example.com>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",
opaque="", uri="sips:+19185553333@ss1.a.example.com;user=phone",
response="6c792f5c9fa360358b93c7fb826bf550"
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> Alice
SIP/2.0 100 Trying
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Content-Length: 0
F3 INVITE Proxy 1 -> GW 1
INVITE sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:alice@client.a.example.com>
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying GW -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Content-Length: 0
F5 SETUP GW 1 -> Carol
Protocol discriminator=Q.931
Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio)
Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN;further call
progress information may be available inband)
Called party number:
Type of number unknown
Digits=444-3333
F6 CALL PROCeeding Carol-> GW 1
Protocol discriminator=Q.931
Message type=CALL PROC
Channel identification=Exclusive B-channel
F7 PROGress Carol-> GW 1
Protocol discriminator=Q.931
Message type=PROG
Progress indicator=1 (Call is not end-to-end ISDN;further call
progress information may be available inband)
F8 180 Ringing GW 1 -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Length: 0
F9 180 Ringing Proxy 1 -> Alice
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Length: 0
F10 CONNect Carol-> GW 1
Protocol discriminator=Q.931
Message type=CONN
F11 CONNect ACK GW 1 -> Carol
Protocol discriminator=Q.931
Message type=CONN ACK
F12 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Type: application/sdp
Content-Length: 144
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F13 200 OK Proxy 1 -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Type: application/sdp
Content-Length: 144
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F14 ACK Alice -> Proxy 1
ACK sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK
Content-Length: 0
F15 ACK Proxy 1 -> GW 1
ACK sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK
Content-Length: 0
/* Alice Hangs Up with Bob. */
F16 BYE Alice -> Proxy 1
BYE sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F17 BYE Proxy 1 -> GW 1
BYE sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F18 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F19 200 OK Proxy 1 -> A
SIP/2.0 200 OK
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F20 DISConnect GW 1 -> Carol
Protocol discriminator=Q.931
Message type=DISC
Cause=16 (Normal clearing)
F21 RELease Carol-> GW 1
Protocol discriminator=Q.931
Message type=REL
F22 RELease COMplete GW 1 -> Carol
Protocol discriminator=Q.931
Message type=REL COM