draft-ietf-sipping-3pcc-03.txt   draft-ietf-sipping-3pcc-04.txt 
Internet Engineering Task Force SIPPING WG SIPPING J. Rosenberg
Internet Draft J. Rosenberg Internet-Draft dynamicsoft
dynamicsoft Expires: December 29, 2003 J. Peterson
J. Peterson
Neustar Neustar
H. Schulzrinne H. Schulzrinne
Columbia U. Columbia University
G. Camarillo G. Camarillo
Ericsson Ericsson Advanced Signalling
draft-ietf-sipping-3pcc-03.txt Research Lab
March 2, 2003 June 30, 2003
Expires: September 2003
Best Current Practices for Third Party Call Control Best Current Practices for Third Party Call Control in the Session
in the Session Initiation Protocol Initiation Protocol
draft-ietf-sipping-3pcc-04
STATUS OF THIS MEMO Status of this Memo
This document is an Internet-Draft and is in full conformance with This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026. all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that Task Force (IETF), its areas, and its working groups. Note that other
other groups may also distribute working documents as Internet- groups may also distribute working documents as Internet-Drafts.
Drafts.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress". material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at The list of current Internet-Drafts can be accessed at http://
http://www.ietf.org/ietf/1id-abstracts.txt www.ietf.org/ietf/1id-abstracts.txt.
To view the list Internet-Draft Shadow Directories, see The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html. http://www.ietf.org/shadow.html.
This Internet-Draft will expire on December 29, 2003.
Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract Abstract
Third party call control refers to the ability of one entity to Third party call control refers to the ability of one entity to
create a call in which communications is actually between other create a call in which communications is actually between other
parties. Third party call control is possible using the mechanisms parties. Third party call control is possible using the mechanisms
specified within the Session Initiation Protocol (SIP). However, specified within the Session Initiation Protocol (SIP). However,
there are several possible approaches, each with different benefits there are several possible approaches, each with different benefits
and drawbacks. This document discusses best current practices for the and drawbacks. This document discusses best current practices for the
usage of SIP for third party call control. usage of SIP for third party call control.
Table of Contents Table of Contents
1 Introduction ........................................ 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2 Terminology ......................................... 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . 4
3 Definitions ......................................... 3 3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . 5
4 3pcc Call Establishment ............................. 4 4. 3pcc Call Establishment . . . . . . . . . . . . . . . . . . 6
4.1 Flow I .............................................. 4 4.1 Flow I . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.2 Flow II ............................................. 5 4.2 Flow II . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.3 Flow III ............................................ 7 4.3 Flow III . . . . . . . . . . . . . . . . . . . . . . . . . . 9
4.4 Flow IV ............................................. 8 4.4 Flow IV . . . . . . . . . . . . . . . . . . . . . . . . . . 11
4.5 Recommendations ..................................... 9 5. Recommendations . . . . . . . . . . . . . . . . . . . . . . 13
5 Error Handling ...................................... 10 6. Error Handling . . . . . . . . . . . . . . . . . . . . . . . 14
6 Continued Processing ................................ 11 7. Continued Processing . . . . . . . . . . . . . . . . . . . . 16
7 3pcc and Early Media ................................ 12 8. 3pcc and Early Media . . . . . . . . . . . . . . . . . . . . 18
8 Third Party Call Control and SDP Preconditions ...... 16 9. Third Party Call Control and SDP Preconditions . . . . . . . 21
8.1 Controller Initiates ................................ 16 9.1 Controller Initiates . . . . . . . . . . . . . . . . . . . . 21
8.2 Party A Initiates ................................... 17 9.2 Party A Initiates . . . . . . . . . . . . . . . . . . . . . 23
9 Example Call Flows .................................. 20 10. Example Call Flows . . . . . . . . . . . . . . . . . . . . . 27
9.1 Click to Dial ....................................... 20 10.1 Click to Dial . . . . . . . . . . . . . . . . . . . . . . . 27
9.2 Mid-Call Announcement Capability .................... 22 10.2 Mid-Call Announcement Capability . . . . . . . . . . . . . . 28
10 Implementation Recommendations ...................... 24 11. Implementation Recommendations . . . . . . . . . . . . . . . 31
11 Security Considerations ............................. 24 12. Security Considerations . . . . . . . . . . . . . . . . . . 32
11.1 Identity ............................................ 24 12.1 Identity . . . . . . . . . . . . . . . . . . . . . . . . . . 32
11.2 End-to-End Encryption and Integrity ................. 25 12.2 End-to-End Encryption and Integrity . . . . . . . . . . . . 32
12 IANA Considerations ................................. 26 13. IANA Considerations . . . . . . . . . . . . . . . . . . . . 34
13 Acknowledgements .................................... 26 14. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 35
14 Authors Addresses ................................... 26 Normative References . . . . . . . . . . . . . . . . . . . . 36
15 Normative References ................................ 27 Informative References . . . . . . . . . . . . . . . . . . . 37
16 Informative References .............................. 27 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 37
Intellectual Property and Copyright Statements . . . . . . . 39
1 Introduction 1. Introduction
In the traditional telephony context, third party call control allows In the traditional telephony context, third party call control allows
one entity (which we call the controller) to set up and manage a one entity (which we call the controller) to set up and manage a
communications relationship between two or more other parties. Third communications relationship between two or more other parties. Third
party call control (referred to as 3pcc) is often used for operator party call control (referred to as 3pcc) is often used for operator
services (where an operator creates a call that connects two services (where an operator creates a call that connects two
participants together), and conferencing. participants together), and conferencing.
Similarly, many SIP services are possible through third party call Similarly, many SIP services are possible through third party call
control. These include the traditional ones on the PSTN, but also new control. These include the traditional ones on the PSTN, but also new
skipping to change at page 3, line 38 skipping to change at page 3, line 38
SIP. In particular, the usage of RFC 3312 [2] (used for coupling of SIP. In particular, the usage of RFC 3312 [2] (used for coupling of
signaling to resource reservation) with third party call control is signaling to resource reservation) with third party call control is
non-trivial. Similarly, the usage of early media (where session data non-trivial. Similarly, the usage of early media (where session data
is exchanged before the call is accepted) with third party call is exchanged before the call is accepted) with third party call
control is not trivial. control is not trivial.
This document serves as a best current practice for implementing This document serves as a best current practice for implementing
third party call control without usage of any extensions specifically third party call control without usage of any extensions specifically
designed for that purpose. Section 4 presents the known call flows designed for that purpose. Section 4 presents the known call flows
that can be used to achieve third party call control, and provides that can be used to achieve third party call control, and provides
guidelines on their usage. Section 8 discusses the interactions of guidelines on their usage. Section 9 discusses the interactions of
RFC 3312 [2] with third party call control. Section 7 discusses the RFC 3312 [2] with third party call control. Section 8 discusses the
interactions of early media with third party call control. Section 9 interactions of early media with third party call control. Section 10
provides example applications that make usage of the flows provides example applications that make usage of the flows
recommended here. recommended here.
2 Terminology 2. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED", In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [3] and and "OPTIONAL" are to be interpreted as described in RFC 2119 [3] and
indicate requirement levels for compliant implementations. indicate requirement levels for compliant implementations.
3 Definitions 3. Definitions
The following terms are used throughout this document: The following terms are used throughout this document:
3pcc: Third Party Call Control, which refers to the general 3pcc: Third Party Call Control, which refers to the general ability
ability to manipulate calls between other parties. to manipulate calls between other parties.
Controller: A controller is a SIP User Agent that wishes to Controller: A controller is a SIP User Agent that wishes to create a
create a session between two other user agents. session between two other user agents.
4 3pcc Call Establishment 4. 3pcc Call Establishment
The primary primitive operation of third party call control is the The primary primitive operation of third party call control is the
establishment of a session between participants A and B. establishment of a session between participants A and B.
Establishment of this session is orchestrated by a third party, Establishment of this session is orchestrated by a third party,
referred to as the controller. referred to as the controller.
This section documents three call flows that the controller can This section documents three call flows that the controller can
utilize in order to provide this primitive operation. utilize in order to provide this primitive operation.
4.1 Flow I 4.1 Flow I
skipping to change at page 4, line 40 skipping to change at page 6, line 33
| |------------------>| | |------------------>|
| |(4) 200 OK answer1 | | |(4) 200 OK answer1 |
| |<------------------| | |<------------------|
| |(5) ACK | | |(5) ACK |
| |------------------>| | |------------------>|
|(6) ACK answer1 | | |(6) ACK answer1 | |
|<------------------| | |<------------------| |
|(7) RTP | | |(7) RTP | |
|.......................................| |.......................................|
Figure 1: 3pcc Flow I Figure 1
The call flow for Flow I is shown in Figure 1. The controller first The call flow for Flow I is shown in Figure 1. The controller first
sends an INVITE A (1). This INVITE has no session description. A's sends an INVITE A (1). This INVITE has no session description. A's
phone rings, and A answers. This results in a 200 OK (2) that phone rings, and A answers. This results in a 200 OK (2) that
contains an offer [4]. The controller needs to send its answer in the contains an offer [4]. The controller needs to send its answer in the
ACK, as mandated by [1]. To obtain the answer, it sends the offer it ACK, as mandated by [1]. To obtain the answer, it sends the offer it
got from A (offer1) in an INVITE to B (3). B's phone rings. When B got from A (offer1) in an INVITE to B (3). B's phone rings. When B
answers, the 200 OK (4) contains the answer to this offer, answer1. answers, the 200 OK (4) contains the answer to this offer, answer1.
The controller sends an ACK to B (5), and then passes answer1 to A in The controller sends an ACK to B (5), and then passes answer1 to A in
an ACK sent to it (6). Because the offer was generated by A, and the an ACK sent to it (6). Because the offer was generated by A, and the
skipping to change at page 5, line 46 skipping to change at page 7, line 34
| |------------------>| | |------------------>|
|(7) INVITE sdp3 | | |(7) INVITE sdp3 | |
|<------------------| | |<------------------| |
|(8) 200 OK sdp2 | | |(8) 200 OK sdp2 | |
|------------------>| | |------------------>| |
|(9) ACK | | |(9) ACK | |
|<------------------| | |<------------------| |
|(10) RTP | | |(10) RTP | |
|.......................................| |.......................................|
Figure 2: 3pcc Flow II Figure 2
An alternative flow, Flow II, is shown in Figure 2. The controller An alternative flow, Flow II, is shown in Figure 2. The controller
first sends an INVITE to user A (1). This is a standard INVITE, first sends an INVITE to user A (1). This is a standard INVITE,
containing an offer (sdp1) with a single audio media line, one codec, containing an offer (sdp1) with a single audio media line, one codec,
a random port number (but not zero), and a connection address of a random port number (but not zero), and a connection address that is
0.0.0.0. This creates an initial media stream that is "black holed", somewhere within the .invalid DNS top level domain (TLD) [8] (for
since no media (or RTCP packets [8]) will flow from A. The INVITE example, rtp.invalid). The use of a connection address in the invalid
causes A's phone to ring. domain creates an initial media stream that is ``black holed'', since
no media (or RTCP packets [9] will flow from A. The INVITE causes A's
phone to ring.
When A answers (2), the 200 OK contains an answer, sdp2. the When A answers (2), the 200 OK contains an answer, sdp2, with a valid
controller sends an ACK (4). It then generates a second INVITE (3). address in the connection line.
This INVITE is addressed to user B, and it contains sdp2 as the offer
to B. Note that the role of sdp2 has changed. In the 200 OK (message A user agent should respond to an offer containing a .invalid
2), it was an answer, but in the INVITE, it is an offer. Fortunatly, domain just as it would any other offer. The answer contains its
all valid answers are valid initial offers. This INVITE causes B's own address in the connection line. However, due to the presence
phone to ring. When it answers, it generates a 200 OK (5) with an of a purposefully invalid domain name, media would not be sent to
answer, sdp3. The controller then generates an ACK (6). Next, it the offerer.
sends a re-INVITE to A (7) containing sdp3 as the offer. Once again,
there has been a reversal of roles. sdp3 was an answer, and now it is The controller sends an ACK (4). It then generates a second INVITE
an offer. Fortunately, an answer to an answer recast as an offer is, (3). This INVITE is addressed to user B, and it contains sdp2 as the
in turn, a valid offer. This re-INVITE generates a 200 OK (8) with offer to B. Note that the role of sdp2 has changed. In the 200 OK
sdp2, assuming that A doesn't decide to change any aspects of the (message 2), it was an answer, but in the INVITE, it is an offer.
session as a result of this re-INVITE. This 200 OK is ACKed (9), and Fortunatly, all valid answers are valid initial offers. This INVITE
then media can flow from A to B. Media from B to A could already causes B's phone to ring. When it answers, it generates a 200 OK (5)
start flowing once message 5 was sent. with an answer, sdp3. The controller then generates an ACK (6). Next,
it sends a re-INVITE to A (7) containing sdp3 as the offer. Once
again, there has been a reversal of roles. sdp3 was an answer, and
now it is an offer. Fortunately, an answer to an answer recast as an
offer is, in turn, a valid offer. This re-INVITE generates a 200 OK
(8) with sdp2, assuming that A doesn't decide to change any aspects
of the session as a result of this re-INVITE. This 200 OK is ACKed
(9), and then media can flow from A to B. Media from B to A could
already start flowing once message 5 was sent.
This flow has the advtange that all final responses are immediately This flow has the advtange that all final responses are immediately
ACKed. It therefore does not suffer from the timeout and message ACKed. It therefore does not suffer from the timeout and message
inefficiency problems of flow 1. However, it too has troubles. First inefficiency problems of flow 1. However, it too has troubles. First
off, it requires that the controller know the media types to be used off, it requires that the controller know the media types to be used
for the call (since it must generate a "blackhole" SDP, which for the call (since it must generate a ``blackhole'' SDP, which
requires media lines). Secondly, the first INVITE to A (1) contains requires media lines). Secondly, the first INVITE to A (1) contains
media with a 0.0.0.0 connection address. The controller expects that media with a rtp.invalid connection address to indicate that neither
the response contains a valid, non-zero connection address for A. media nor RTCP packets should be sent. This mechanism for indicating
However, experience has shown that many UAs respond to an offer of a that neither media or RTCP should be sent is different the one
0.0.0.0 connection address with an answer containing a 0.0.0.0 documented in RFC 3264 [4], which uses 0.0.0.0 for this purpose.
connection address. The offer-answer specification [4] now explicitly However, according to RFC 3330 [11], 0.0.0.0 is a special case
tells implementors not to do this, but at the time of publication of address which refers to hosts on this network. This is not the same
this document, many implementations still did. If A should respond as the desired semantic, which is much closer to the meaning of the
with a 0.0.0.0 connection address in sdp2, the flow will not work. .invalid TLD. However, it is unknown how off-the-shelf user agents
will react to an SDP that contains a .invalid TLD in a connection
line.
However, the most serious flaw in this flow is the assumption that However, the most serious flaw in this flow is the assumption that
the 200 OK to the re-INVITE (message 8) contains the same SDP as in the 200 OK to the re-INVITE (message 8) contains the same SDP as in
message 2. This may not be the case. If it is not, the controller message 2. This may not be the case. If it is not, the controller
needs to re-INVITE B with that SDP (say, sdp4), which may result in needs to re-INVITE B with that SDP (say, sdp4), which may result in
getting a different SDP, sdp5 , in the 200 OK from B. Then, the getting a different SDP, sdp5 , in the 200 OK from B. Then, the
controller needs to re-INVITE A again, and so on. The result is an controller needs to re-INVITE A again, and so on. The result is an
infinite loop of re-INVITEs. It is possible to break this cycle by infinite loop of re-INVITEs. It is possible to break this cycle by
having very smart UAs which can return the same SDP whenever having very smart UAs which can return the same SDP whenever
possible, or really smart controllers that can analyze the SDP to possible, or really smart controllers that can analyze the SDP to
skipping to change at page 7, line 34 skipping to change at page 9, line 31
|<---------------------| | |<---------------------| |
|(7) 200 answer2' | | |(7) 200 answer2' | |
|--------------------->| | |--------------------->| |
| |(8) ACK answer2 | | |(8) ACK answer2 |
| |--------------------->| | |--------------------->|
|(9) ACK | | |(9) ACK | |
|<---------------------| | |<---------------------| |
|(10) RTP | | |(10) RTP | |
|.............................................| |.............................................|
Figure 3: 3pcc Flow III Figure 3
A third flow, Flow III, is shown in Figure 3. A third flow, Flow III, is shown in Figure 3
First, the controller sends an INVITE (1) to user A without any SDP First, the controller sends an INVITE (1) to user A without any SDP
(which is good, since it means that the controller doesn't need to (which is good, since it means that the controller doesn't need to
assume anything about the media composition of the session). A's assume anything about the media composition of the session). A's
phone rings. When A answers, a 200 OK is generated (2) containing its phone rings. When A answers, a 200 OK is generated (2) containing its
offer, offer1. The controller generates an immediate ACK containing offer, offer1. The controller generates an immediate ACK containing
an answer (3). This answer is a "black hole" SDP, with its connection an answer (3). This answer is a ``black hole'' SDP, with its
address set to 0.0.0.0. connection address set to any domain in the .invalid TLD, such as
media.invalid.
The controller then sends an INVITE to B without SDP (4). This causes The controller then sends an INVITE to B without SDP (4). This causes
B's phone to ring. When they answer, a 200 OK is sent, containing B's phone to ring. When they answer, a 200 OK is sent, containing
their offer, offer2 (5). This SDP is used to create a re-INVITE back their offer, offer2 (5). This SDP is used to create a re-INVITE back
to A (6). That re-INVITE is based on offer2, but may need to be to A (6). That re-INVITE is based on offer2, but may need to be
reorganized to match up media lines, or to trim media lines. For reorganized to match up media lines, or to trim media lines. For
example, if offer1 contained an audio and a video line, in that example, if offer1 contained an audio and a video line, in that
order, but offer2 contained just an audio line, the controller would order, but offer2 contained just an audio line, the controller would
need to add a video line to the offer (setting its port to zero) to need to add a video line to the offer (setting its port to zero) to
create offer2'. Since this is a re-INVITE, it should complete quickly create offer2'. Since this is a re-INVITE, it should complete quickly
in the general case. Thats good, since user B is retransmitting their in the general case. Thats good, since user B is retransmitting their
200 OK, waiting for an ACK. The SDP in the 200 OK (7) from A, 200 OK, waiting for an ACK. The SDP in the 200 OK (7) from A,
answer2', may also need to be reorganized or trimmed before sending answer2', may also need to be reorganized or trimmed before sending
it an the ACK to B (8) as answer2. Finally, an ACK is sent to A (9), it an the ACK to B (8) as answer2. Finally, an ACK is sent to A (9),
and then media can flow. and then media can flow.
This flow has many benefits. First, it will usually operate without This flow has many benefits. First, it will usually operate without
any spurious retransmissions or timeouts (although this may still any spurious retransmissions or timeouts (although this may still
happen if a re-INVITE is not responded to quickly). Secondly, it does happen if a re-INVITE is not responded to quickly). Secondly, it does
not require the controller to guess the media that will be used by not require the controller to guess the media that will be used by
the participants. Thirdly, it does not assume that a device responds the participants.
properly to an INVITE with SDP containing a connection address of
0.0.0.0.
There are some drawbacks. The controller does need to perform SDP There are some drawbacks. The controller does need to perform SDP
manipulations. Specifically, it must take some SDP, and generate manipulations. Specifically, it must take some SDP, and generate
another SDP which has the same media composition, but has connection another SDP which has the same media composition, but has connection
addresses of 0.0.0.0. This is needed for message 3. Secondly, it may addresses within the .invalid TLD. This is needed for message 3.
need to reorder and trim on SDP X, so that its media lines match up Secondly, it may need to reorder and trim on SDP X, so that its media
with those in some other SDP, Y. Thirdly, the offer from B (offer2) lines match up with those in some other SDP, Y. Thirdly, the offer
may have no codecs or media streams in common with the offer from A from B (offer2) may have no codecs or media streams in common with
(offer 1). The controller will need to detect this condition, and the offer from A (offer 1). The controller will need to detect this
terminate the call. Finally, the flow is far more complicated than condition, and terminate the call. Fourth, it requires usage of the
the simple and elegant Flow I (Figure 1). .invalid TLD within SDP, which may not be supported. Finally, the
flow is far more complicated than the simple and elegant Flow I
(Figure 1).
4.4 Flow IV 4.4 Flow IV
Flow IV shows a variation on Flow III that reduces its complexity.
The actual message flow is identical, but the SDP placement and
construction differs. The initial INVITE (1) contains SDP with no
media at all, meaning that there are no m lines. This is valid, and
implies that the media makeup of the session will be established
later through a re-INVITE [4]. Once the INVITE is received, user A is
alerted. When they answer the call, the 200 OK (2) has an answer with
no media either. This is acknowledged by the controller (3). The flow
from this point onwards is identical to Flow III. However, the
manipulations required to convert offer2 to offer2', and answer2' to
answer2, are much simpler. Indeed, no media manipulations are needed
at all. The only change that is needed is to modify the origin lines,
A Controller B A Controller B
|(1) INVITE offer1 | | |(1) INVITE offer1 | |
|no media | | |no media | |
|<---------------------| | |<---------------------| |
|(2) 200 answer1 | | |(2) 200 answer1 | |
|no media | | |no media | |
|--------------------->| | |--------------------->| |
|(3) ACK | | |(3) ACK | |
|<---------------------| | |<---------------------| |
| |(4) INVITE no SDP | | |(4) INVITE no SDP |
skipping to change at page 9, line 28 skipping to change at page 11, line 31
|<---------------------| | |<---------------------| |
|(7) 200 answer2' | | |(7) 200 answer2' | |
|--------------------->| | |--------------------->| |
| |(8) ACK answer2 | | |(8) ACK answer2 |
| |--------------------->| | |--------------------->|
|(9) ACK | | |(9) ACK | |
|<---------------------| | |<---------------------| |
|(10) RTP | | |(10) RTP | |
|.............................................| |.............................................|
Figure 4: 3pcc Flow IV Figure 4
Flow IV shows a variation on Flow III that reduces its complexity.
The actual message flow is identical, but the SDP placement and
construction differs. The initial INVITE (1) contains SDP with no
media at all, meaning that there are no m lines. This is valid, and
implies that the media makeup of the session will be established
later through a re-INVITE [4]. Once the INVITE is received, user A is
alerted. When they answer the call, the 200 OK (2) has an answer with
no media either. This is acknowledged by the controller (3). The flow
from this point onwards is identical to Flow III. However, the
manipulations required to convert offer2 to offer2', and answer2' to
answer2, are much simpler. Indeed, no media manipulations are needed
at all. The only change that is needed is to modify the origin lines,
so that the origin line in offer2' is valid based on the value in so that the origin line in offer2' is valid based on the value in
offer1 (validify requires that the version increments by one, and offer1 (validify requires that the version increments by one, and
that the other parameters remain unchanged). that the other parameters remain unchanged).
There are some limitations associated with this flow. First, user A There are some limitations associated with this flow. First, user A
will be alerted without any media having been established yet. This will be alerted without any media having been established yet. This
means that user A will not be able to reject or accept the call based means that user A will not be able to reject or accept the call based
on its media composition. Secondly, both A and B will end up on its media composition. Secondly, both A and B will end up
answering the call (i.e., generating a 200 OK) before it is known answering the call (i.e., generating a 200 OK) before it is known
whether their is compatible media. If there is no media in common, whether their is compatible media. If there is no media in common,
the call can be terminated later with a BYE. However, the users will the call can be terminated later with a BYE. However, the users will
have already been alerted, resulting in user annoyance and possibly have already been alerted, resulting in user annoyance and possibly
resulting in billing events. resulting in billing events.
4.5 Recommendations 5. Recommendations
Flow I (Figure 1) represents the simplest and the most efficient Flow I (Figure 1) represents the simplest and the most efficient
flow. This flow SHOULD be used by a controller if it knows with flow. This flow SHOULD be used by a controller if it knows with
certainty that user B is actually an automata that will answer the certainty that user B is actually an automata that will answer the
call immediately. This is the case for devices such as media servers, call immediately. This is the case for devices such as media servers,
conferencing servers, and messaging servers, for example. Since we conferencing servers, and messaging servers, for example. Since we
expect a great deal of third party call control to be to automata, expect a great deal of third party call control to be to automata,
special caseing this scenario is reasonable. special caseing this scenario is reasonable.
For calls to unknown entities, or to entities known to represent For calls to unknown entities, or to entities known to represent
people, it is RECOMMENDED that Flow IV (Figure 4) be used for third people, it is RECOMMENDED that Flow IV (Figure 4) be used for third
party call control. Flow III MAY be used instead, but it provides no party call control. Flow III MAY be used instead, but it provides no
additional benefits over Flow IV. However, Flow II SHOULD NOT be additional benefits over Flow IV. However, Flow II SHOULD NOT be
used, because of the potential for infinite ping-ponging of re- used, because of the potential for infinite ping-ponging of
INVITEs. re-INVITEs.
Several of these flows use a "black hole" connection address of Several of these flows use a ``black hole'' connection address, which
0.0.0.0. This is an IPV4 address with the property that packets sent is any domain in the .invalid domain. A UA MUST be prepared to
to it will never leave the host which sent them; they are just receive offers or answers with connection lines containing addresses
discarded. Those flows are therefore specific to IPv4. For other within this domain. A UA receiving such a domain proceeds with normal
network or address types, an address with an equivalent property RFC 3264 offer/answer processing. However, it will not send media to
SHOULD be used. the address. A UA MAY bypass DNS lookup of the connection line when
it sees it's within the .invalid domain. Of course, a host that
performs a lookup with receive a response indicating that the host
name is invalid.
5 Error Handling 6. Error Handling
There are numerous error cases which merit discussion. There are numerous error cases which merit discussion.
With all of the call flows in Section 4, one call is established to With all of the call flows in Section 4, one call is established to
A, and then the controller attempts to establish a call to B. A, and then the controller attempts to establish a call to B.
However, this call attempt may fail, for any number of reasons. User However, this call attempt may fail, for any number of reasons. User
B might be busy (resulting in a 486 response to the INVITE), there B might be busy (resulting in a 486 response to the INVITE), there
may not be any media in common, the request may time out, and so on. may not be any media in common, the request may time out, and so on.
If the call attempt to B should fail, it is RECOMMENDED that the If the call attempt to B should fail, it is RECOMMENDED that the
controller send a BYE to A. This BYE SHOULD include a Reason header controller send a BYE to A. This BYE SHOULD include a Reason header
[5] which carries the status code from the error response. This will [5] which carries the status code from the error response. This will
inform A of the precise reason for the failure. The information is inform A of the precise reason for the failure. The information is
important from a user interface perspective. For example, if A was important from a user interface perspective. For example, if A was
calling from a black phone, and B generated a 486, the BYE will calling from a black phone, and B generated a 486, the BYE will
contain a Reason code of 486, and this could be used to generate a contain a Reason code of 486, and this could be used to generate a
local busy signal so that A knows that B is busy. local busy signal so that A knows that B is busy.
Another error condition worth discussion is shown in Figure 5. After
the controller establishes the dialog with A (messages 1-3) it
attempts to contact B (message 4). Contacting B may take some time.
During that interval, A could possibly attempt a re-INVITE, providing
an updated offer. However, the controller cannot pass this offer on
to B, since it has an INVITE transaction pending with it. As a
result, the controller needs to reject the request. It is RECOMMENDED
that a 491 response be used. The situation here is similar to the
glare condition described in [1], and thus the same error handling is
A Controller B A Controller B
|(1) INVITE offer1 | | |(1) INVITE offer1 | |
|no media | | |no media | |
|<---------------------| | |<---------------------| |
|(2) 200 answer1 | | |(2) 200 answer1 | |
|no media | | |no media | |
|--------------------->| | |--------------------->| |
|(3) ACK | | |(3) ACK | |
|<---------------------| | |<---------------------| |
| |(4) INVITE no SDP | | |(4) INVITE no SDP |
| |--------------------->| | |--------------------->|
| |(5) 180 | | |(5) 180 |
| |<---------------------| | |<---------------------|
|(6) INVITE offer2 | | |(6) INVITE offer2 | |
|--------------------->| | |--------------------->| |
|(7) 491 | | |(7) 491 | |
|<---------------------| | |<---------------------| |
|(8) ACK | | |(8) ACK | |
|--------------------->| | |--------------------->| |
Figure 5: Glare Error Condition Figure 5
Another error condition worth discussion is shown in Figure 5. After
the controller establishes the dialog with A (messages 1-3) it
attempts to contact B (message 4). Contacting B may take some time.
During that interval, A could possibly attempt a re-INVITE, providing
an updated offer. However, the controller cannot pass this offer on
to B, since it has an INVITE transaction pending with it. As a
result, the controller needs to reject the request. It is RECOMMENDED
that a 491 response be used. The situation here is similar to the
glare condition described in [1], and thus the same error handling is
sensible. However, A is likely to retry its request (as a result of sensible. However, A is likely to retry its request (as a result of
the 491), and this may occur before the exchange with B is completed. the 491), and this may occur before the exchange with B is completed.
In that case, the controller would respond with another 491. In that case, the controller would respond with another 491.
6 Continued Processing 7. Continued Processing
Once the calls are established, both participants believe they are in Once the calls are established, both participants believe they are in
a single point-to-point call. However, they are exchanging media a single point-to-point call. However, they are exchanging media
directly with each other, rather than with the controller. The directly with each other, rather than with the controller. The
controller is involved in two dialogs, yet sees no media. controller is involved in two dialogs, yet sees no media.
Since the controller is still a central point for signaling, it now Since the controller is still a central point for signaling, it now
has complete control over the call. If it receives a BYE from one of has complete control over the call. If it receives a BYE from one of
the participants, it can create a new BYE and hang up with the other the participants, it can create a new BYE and hang up with the other
participant. This is shown in Figure 6. participant. This is shown in Figure 6.
Similarly, if it receives a re-INVITE from one of the participants,
it can forward it to the other participant. Depending on which flow
was used, this may require some manipulation on the SDP before
passing it on.
A Controller B A Controller B
|(1) BYE | | |(1) BYE | |
|------------------>| | |------------------>| |
|(2) 200 OK | | |(2) 200 OK | |
|<------------------| | |<------------------| |
| |(3) BYE | | |(3) BYE |
| |------------------>| | |------------------>|
| |(4) 200 OK | | |(4) 200 OK |
| |<------------------| | |<------------------|
Figure 6: Hanging Up with 3PCC Figure 6
Similarly, if it receives a re-INVITE from one of the participants,
it can forward it to the other participant. Depending on which flow
was used, this may require some manipulation on the SDP before
passing it on.
However, the controller need not "proxy" the SIP messages received However, the controller need not "proxy" the SIP messages received
from one of the parties. Since it is a B2BUA, it can invoke any from one of the parties. Since it is a B2BUA, it can invoke any
signaling mechanism on each dialog, as it sees fit. For example, if signaling mechanism on each dialog, as it sees fit. For example, if
the controller receives a BYE from A, it can generate a new INVITE to the controller receives a BYE from A, it can generate a new INVITE to
a third party, C, and connect B to that participant instead. A call a third party, C, and connect B to that participant instead. A call
flow for this is shown in Figure 7, assuming the case where C flow for this is shown in Figure 7, assuming the case where C
represents an end user, not an automata. Note that it is just Flow represents an end user, not an automata. Note that it is just Flow
IV. IV.
From here, new parties can be added, removed, transferred, and so on,
as the controller sees fit.
It is important to point out that the call need not have been
established by the controller in order for the processing of this
section to be used. Rather, the controller could have acted as a
B2BUA during a call established by A towards B (or vice a versa).
7 3pcc and Early Media
Early media represents the condition where the session is established
(as a result of the completion of an offer/answer exchange), yet the
call itself has not been accepted. This is usually used to convey
tones or announcements regarding progress of the call. Handling of
early media in a third party call is straightforward.
Figure 8 shows the case where user B generates early media before
answering the call. The flow is almost identical to Flow IV from
Figure 4. The only difference is that user B generates a reliable
provisional response (5) [6] instead of a final response, and answer2
is carried in a PRACK (8) instead of an ACK. When party B finally
A Controller B C A Controller B C
|(1) BYE | | | |(1) BYE | | |
|--------------->| | | |--------------->| | |
|(2) 200 OK | | | |(2) 200 OK | | |
|<---------------| | | |<---------------| | |
| |(3) INV no media| | | |(3) INV no media| |
| |-------------------------------->| | |-------------------------------->|
| |(4) 200 no media| | | |(4) 200 no media| |
| |<--------------------------------| | |<--------------------------------|
| |(5) ACK | | | |(5) ACK | |
skipping to change at page 13, line 30 skipping to change at page 17, line 31
| |-------------------------------->| | |-------------------------------->|
| |(9) 200 answer3'| | | |(9) 200 answer3'| |
| |<--------------------------------| | |<--------------------------------|
| |(10) ACK | | | |(10) ACK | |
| |-------------------------------->| | |-------------------------------->|
| |(11) ACK answer3| | | |(11) ACK answer3| |
| |--------------->| | | |--------------->| |
| | |(12) RTP | | | |(12) RTP |
| | |................| | | |................|
Figure 7: Alternative to Hangup Figure 7
does accept the call (11), there is no change in the session state, From here, new parties can be added, removed, transferred, and so on,
and therefore, no signaling needs to be done with user A. The as the controller sees fit.
controller simply ACKs the 200 OK (12) to confirm the dialog.
It is important to point out that the call need not have been
established by the controller in order for the processing of this
section to be used. Rather, the controller could have acted as a
B2BUA during a call established by A towards B (or vice a versa).
8. 3pcc and Early Media
Early media represents the condition where the session is established
(as a result of the completion of an offer/answer exchange), yet the
call itself has not been accepted. This is usually used to convey
tones or announcements regarding progress of the call. Handling of
early media in a third party call is straightforward.
The case where user A generates early media is more complicated, and
is shown in Figure 9. The flow is based on Flow IV. The controller
sends an INVITE to user A (1), with an offer containing no media
streams. User A generates a reliable provisional response (2)
containing an answer with no media streams. The controller PRACKs
this provisional response (3). Now, the controller sends an INVITE
without SDP to user B (5). User B's phone rings, and they answer,
resulting in a 200 OK (6) with an offer, offer2. The controller now
needs to update the session parameters with user A. However, since
the call has not been answered, it cannot use a re-INVITE. Rather, it
uses a SIP UPDATE request (7) [7], passing the offer (after modifying
A Controller B A Controller B
| | | | | |
|(1) INVITE offer1 | | |(1) INVITE offer1 | |
|no media | | |no media | |
|<---------------------| | |<---------------------| |
| | | | | |
|<ring> | | |<ring> | |
| | | | | |
| | |
|<answer> | | |<answer> | |
| | | | | |
|(2) 200 answer1 | | |(2) 200 answer1 | |
|no media | | |no media | |
|--------------------->| | |--------------------->| |
| | |
|(3) ACK | | |(3) ACK | |
|<---------------------| | |<---------------------| |
| | |
| |(4) INVITE no SDP | | |(4) INVITE no SDP |
| |--------------------->| | |--------------------->|
| | |
| | |<ring> | | |<ring>
| | |
| | |
| |(5) 183 offer2 | | |(5) 183 offer2 |
| |<---------------------| | |<---------------------|
| | |
|(6) INVITE offer2' | | |(6) INVITE offer2' | |
|<---------------------| | |<---------------------| |
| | |
|(7) 200 answer2' | | |(7) 200 answer2' | |
|--------------------->| | |--------------------->| |
| | |
|(8) ACK | | |(8) ACK | |
|<---------------------| | |<---------------------| |
| | |
| |(9) PRACK answer2 | | |(9) PRACK answer2 |
| |--------------------->| | |--------------------->|
| | |
| |(10) 200 PRACK | | |(10) 200 PRACK |
| |<---------------------| | |<---------------------|
| | |
|(11) RTP | | |(11) RTP | |
|.............................................| |.............................................|
| | |
| | |<answer> | | |<answer>
| | |
| | |
| |(12) 200 OK | | |(12) 200 OK |
| |<---------------------| | |<---------------------|
| | |
| |(13) ACK | | |(13) ACK |
| |--------------------->| | |--------------------->|
| | | Figure 8
| | |
| | | Figure 8 shows the case where user B generates early media before
answering the call. The flow is almost identical to Flow IV from
Figure 4. The only difference is that user B generates a reliable
provisional response (5) [6] instead of a final response, and answer2
is carried in a PRACK (8) instead of an ACK. When party B finally
does accept the call (11), there is no change in the session state,
and therefore, no signaling needs to be done with user A. The
controller simply ACKs the 200 OK (12) to confirm the dialog.
Figure 8: Early Media from User B
A Controller B A Controller B
| | | | | |
|(1) INVITE offer1 | | |(1) INVITE offer1 | |
|no media | | |no media | |
|<---------------------| | |<---------------------| |
| | | | | |
|ring | | |ring | |
| | | | | |
|(2) 183 answer1 | | |(2) 183 answer1 | |
|no media | | |no media | |
|--------------------->| | |--------------------->| |
| | |
|(3) PRACK | | |(3) PRACK | |
|<---------------------| | |<---------------------| |
| | |
|(4) 200 PRACK | | |(4) 200 PRACK | |
|--------------------->| | |--------------------->| |
| | |
| |(5) INVITE no SDP | | |(5) INVITE no SDP |
| |--------------------->| | |--------------------->|
| | |
| | |ring | | |ring
| | | | | |
| | |
| | |answer | | |answer
| | | | | |
| | |
| |(6) 200 OK offer2 | | |(6) 200 OK offer2 |
| |<---------------------| | |<---------------------|
| | |
|(7) UPDATE offer2' | | |(7) UPDATE offer2' | |
|<---------------------| | |<---------------------| |
| | |
|answer | | |answer | |
| | | | | |
| | |
|(8) 200 answer2' | | |(8) 200 answer2' | |
|--------------------->| | |--------------------->| |
| | |
| |(9) ACK answer2 | | |(9) ACK answer2 |
| |--------------------->| | |--------------------->|
| | |
|(10) RTP | | |(10) RTP | |
|.............................................| |.............................................|
| | |
|(11) 200 OK | | |(11) 200 OK | |
|--------------------->| | |--------------------->| |
| | |
|(12) ACK | | |(12) ACK | |
|<---------------------| | |<---------------------| |
| | |
| | |
| | |
Figure 9: Early Media from User A
Figure 9
The case where user A generates early media is more complicated, and
is shown in Figure 9. The flow is based on Flow IV. The controller
sends an INVITE to user A (1), with an offer containing no media
streams. User A generates a reliable provisional response (2)
containing an answer with no media streams. The controller PRACKs
this provisional response (3). Now, the controller sends an INVITE
without SDP to user B (5). User B's phone rings, and they answer,
resulting in a 200 OK (6) with an offer, offer2. The controller now
needs to update the session parameters with user A. However, since
the call has not been answered, it cannot use a re-INVITE. Rather, it
uses a SIP UPDATE request (7) [7], passing the offer (after modifying
it to get the origin field correct). User A generates its answer in it to get the origin field correct). User A generates its answer in
the 200 OK to the UPDATE (8). This answer is passed to user B in the the 200 OK to the UPDATE (8). This answer is passed to user B in the
ACK (9). When user A finally answers (11), there is no change in ACK (9). When user A finally answers (11), there is no change in
session state, so the controller simply ACKs the 200 OK (12). session state, so the controller simply ACKs the 200 OK (12).
Note that it is likely that there will be clipping of media in this Note that it is likely that there will be clipping of media in this
call flow. User A is likely a PSTN gateway, and has generated a call flow. User A is likely a PSTN gateway, and has generated a
provisional response because of early media from the PSTN side. The provisional response because of early media from the PSTN side. The
PSTN will deliver this media even though the gateway does not have PSTN will deliver this media even though the gateway does not have
anywhere to send it, since the initial offer from the controller had anywhere to send it, since the initial offer from the controller had
no media streams. When user B answers, media can begin to flow. no media streams. When user B answers, media can begin to flow.
However, any media sent to the gateway from the PSTN up to that point However, any media sent to the gateway from the PSTN up to that point
will be lost. will be lost.
8 Third Party Call Control and SDP Preconditions 9. Third Party Call Control and SDP Preconditions
A SIP extension has been specified that allows for the coupling of A SIP extension has been specified that allows for the coupling of
signaling and resource reservation [2]. This draft relies on signaling and resource reservation [2]. This specification relies on
exchanges of session descriptions before completion of the call exchanges of session descriptions before completion of the call
setup. These flows are initiated when certain SDP parameters are setup. These flows are initiated when certain SDP parameters are
passed in the initial INVITE. As a result, the interaction of this passed in the initial INVITE. As a result, the interaction of this
mechanism with third party call control is not obvious, and worth mechanism with third party call control is not obvious, and worth
detailing. detailing.
8.1 Controller Initiates 9.1 Controller Initiates
In one usage scenario, the controller wishes to make use of In one usage scenario, the controller wishes to make use of
preconditions in order to avoid the call failure scenarios documented preconditions in order to avoid the call failure scenarios documented
in Section 4.4. Specifically, the controller can use preconditions in in Section 4.4. Specifically, the controller can use preconditions in
order to guarantee that neither party is alerted unless there is a order to guarantee that neither party is alerted unless there is a
common set of media and codecs. It can also provide both parties with common set of media and codecs. It can also provide both parties with
information on the media composition of the call before they decide information on the media composition of the call before they decide
to accept it. to accept it.
User Controller Customer Service
| | |
|(1) INVITE no SDP | |
|require precon | |
|<------------------| |
|(2) 183 offer1 | |
|optional precon | |
|------------------>| |
| | |
| |(3) INVITE offer1 |
| |------------------>|
| | |<ring>
| | |
| | |<answer>
| |(4) 200 OK answer1 |
| |no precon |
| |<------------------|
| |(5) ACK |
| |------------------>|
|(6) PRACK answer1 | |
|<------------------| |
|<ring> | |
| | |
|(7) 200 PRACK | |
|------------------>| |
|<answer> | |
| | |
|(8) 200 INVITE | |
|------------------>| |
|(9) ACK | |
|<------------------| |
Figure 10
The flow for this scenario is shown in Figure 10. In this example, we The flow for this scenario is shown in Figure 10. In this example, we
assume that user B is an automata or agent of some sort which will assume that user B is an automata or agent of some sort which will
answer the call immediately. Therefore, the flow is based on Flow I. answer the call immediately. Therefore, the flow is based on Flow I.
The controller sends an INVITE to user A containing no SDP, but with The controller sends an INVITE to user A containing no SDP, but with
a Require header indicating that preconditions are required. This a Require header indicating that preconditions are required. This
specific scenario (an INVITE without an offer, but with a Require specific scenario (an INVITE without an offer, but with a Require
header indicating preconditions) is not described in [2]. It is header indicating preconditions) is not described in [2]. It is
RECOMMENDED that the UAS respond with an offer in a 1xx including the RECOMMENDED that the UAS respond with an offer in a 1xx including the
media streams it wishes to use for the call, and for each, list all media streams it wishes to use for the call, and for each, list all
preconditions it supports as optional. Of course, the user is not preconditions it supports as optional. Of course, the user is not
skipping to change at page 17, line 29 skipping to change at page 23, line 26
property is achieved using preconditions even though it doesn't property is achieved using preconditions even though it doesn't
matter what specific types of preconditions are supported by user A. matter what specific types of preconditions are supported by user A.
It is also entirely possible that user B does actually desire It is also entirely possible that user B does actually desire
preconditions. In that case, it might generate a 1xx of its own with preconditions. In that case, it might generate a 1xx of its own with
an answer containing preconditions. That answer would still be passed an answer containing preconditions. That answer would still be passed
to user A, and both parties would proceed with whatever measures are to user A, and both parties would proceed with whatever measures are
necessary to meet the preconditions. Neither user would be alerted necessary to meet the preconditions. Neither user would be alerted
until the preconditions were met. until the preconditions were met.
8.2 Party A Initiates 9.2 Party A Initiates
In Section 8.1, the controller requested the use of preconditions to In Section 9.1, the controller requested the use of preconditions to
achieve a specific goal. It is also possible that the controller achieve a specific goal. It is also possible that the controller
doesn't care (or perhaps doesn't even know) about preconditions, but doesn't care (or perhaps doesn't even know) about preconditions, but
one of the participants in the call does care. A call flow for this one of the participants in the call does care. A call flow for this
case is shown in Figure 11. case is shown in Figure 11.
The controller follows Flow IV; it has no specific requirements for
support of the preconditions specification [2]. Therefore, it sends
an INVITE (1) with SDP that contains no media lines. User A is
interested in supporting preconditions, and does not want to ring its
phone until resources are reserved. Since there are no media streams
in the INVITE, it can't reserve resources for media streams, and
therefore it can't ring the phone until they are conveyed in a
subsequent offer and then reserved. Therefore, it generates a 183
with the answer, and doesn't alert the user (2). The controller
PRACKs this (3) and A responds to the PRACK (4).
At this point, the controller attempts to bring B into the call. It
sends B an INVITE without SDP (5). B is interested in having
preconditions for this call. Therefore, it generates its offer in a
User Controller Customer Service
| | |
|(1) INVITE no SDP | |
|require precon | |
|<------------------| |
|(2) 183 offer1 | |
|optional precon | |
|------------------>| |
| | |
| |(3) INVITE offer1 |
| |------------------>|
| | |
| | |<ring>
| | |
| | |
| | |<answer>
| | |
| |(4) 200 OK answer1 |
| |no precon |
| |<------------------|
| | |
| |(5) ACK |
| |------------------>|
| | |
|(6) PRACK answer1 | |
|<------------------| |
| | |
|<ring> | |
| | |
| | |
|(7) 200 PRACK | |
|------------------>| |
| | |
|<answer> | |
| | |
| | |
|(8) 200 INVITE | |
|------------------>| |
| | |
|(9) ACK | |
|<------------------| |
| | |
| | |
| | |
Figure 10: Controller Initiated Preconditions
A Controller B A Controller B
|(1) INVITE offer1 | | |(1) INVITE offer1 | |
|no media | | |no media | |
|<---------------------| | |<---------------------| |
|(2) 183 answer1 | | |(2) 183 answer1 | |
|no media | | |no media | |
|--------------------->| | |--------------------->| |
|(3) PRACK | | |(3) PRACK | |
|<---------------------| | |<---------------------| |
|(4) 200 OK | | |(4) 200 OK | |
skipping to change at page 20, line 4 skipping to change at page 25, line 39
|<answer> | | |<answer> | |
|(23) 200 INVITE | | |(23) 200 INVITE | |
|--------------------->| | |--------------------->| |
|(24) ACK | | |(24) ACK | |
|<---------------------| | |<---------------------| |
| | |<answer> | | |<answer>
| |(25) 200 INVITE | | |(25) 200 INVITE |
| |<---------------------| | |<---------------------|
| |(26) ACK | | |(26) ACK |
| |--------------------->| | |--------------------->|
Figure 11: User A Initiated Preconditions
Figure 11
The controller follows Flow IV; it has no specific requirements for
support of the preconditions specification [2]. Therefore, it sends
an INVITE (1) with SDP that contains no media lines. User A is
interested in supporting preconditions, and does not want to ring its
phone until resources are reserved. Since there are no media streams
in the INVITE, it can't reserve resources for media streams, and
therefore it can't ring the phone until they are conveyed in a
subsequent offer and then reserved. Therefore, it generates a 183
with the answer, and doesn't alert the user (2). The controller
PRACKs this (3) and A responds to the PRACK (4).
At this point, the controller attempts to bring B into the call. It
sends B an INVITE without SDP (5). B is interested in having
preconditions for this call. Therefore, it generates its offer in a
183 that contains the appropriate SDP attributes (6). The controller 183 that contains the appropriate SDP attributes (6). The controller
passes this offer to A in an UPDATE request (7). The controller uses passes this offer to A in an UPDATE request (7). The controller uses
UPDATE because the call has not been answered yet, and therefore, it UPDATE because the call has not been answered yet, and therefore, it
cannot use a re-INVITE. User A sees that its peer is capable of cannot use a re-INVITE. User A sees that its peer is capable of
supporting preconditions. Since it desires preconditions for the supporting preconditions. Since it desires preconditions for the
call, it generates an answer in the 200 OK (8) to the UPDATE. This call, it generates an answer in the 200 OK (8) to the UPDATE. This
answer, in turn, is passed to B in the PRACK for the provisional answer, in turn, is passed to B in the PRACK for the provisional
response (9). Now, both sides perform resource reservation. User A response (9). Now, both sides perform resource reservation. User A
succeeds first, and passes an updated session description in an succeeds first, and passes an updated session description in an
UPDATE request (13). The controller simply passes this to A (after UPDATE request (13). The controller simply passes this to A (after
the manipulation of the origin field, as required in Flow IV) in an the manipulation of the origin field, as required in Flow IV) in an
UPDATE (14), and the answer (15) is passed back to A (16). The same UPDATE (14), and the answer (15) is passed back to A (16). The same
flow happens, but from B to A, when B's reservation succeeds (17-20). flow happens, but from B to A, when B's reservation succeeds (17-20).
Since the preconditions have been met, both sides ring (21 and 22), Since the preconditions have been met, both sides ring (21 and 22),
and then both answer (23 and 25), completing the call. and then both answer (23 and 25), completing the call.
What is important about this flow is that the controller doesn't know What is important about this flow is that the controller doesn't know
anything about preconditions. It merely passes the SDP back and forth anything about preconditions. It merely passes the SDP back and forth
as needed. The trick is the usage of UPDATE and PRACK to pass the SDP as needed. The trick is the usage of UPDATE and PRACK to pass the SDP
when needed. That determination is made entirely based on the when needed. That determination is made entirely based on the offer/
offer/answer rules described in [6] and [7], and is independent of answer rules described in [6] and [7], and is independent of
preconditions. preconditions.
9 Example Call Flows 10. Example Call Flows
9.1 Click to Dial 10.1 Click to Dial
The first application of this capability we discuss is click to dial. The first application of this capability we discuss is click to dial.
In this service, a user is browsing the web page of an e-commerce In this service, a user is browsing the web page of an e-commerce
site, and would like to speak to a customer service representative. site, and would like to speak to a customer service representative.
They click on a link, and a call is placed to a customer service They click on a link, and a call is placed to a customer service
representative. When the representative picks up, the phone on the representative. When the representative picks up, the phone on the
user's desk rings. When they pick up, the customer service user's desk rings. When they pick up, the customer service
representative is there, ready to talk to the user. representative is there, ready to talk to the user.
The call flow for this service is given in Figure 12. It is identical
to that of Figure 4, with the exception that the service is triggered
through an http GET request when the user clicks on the link.
We note that this service can be provided through other mechanisms,
namely PINT [9]. However, there are numerous differences between the
way in which the service is provided by pint, and the way in which it
is provided here:
Customer Service Controller Users Phone Users Browser Customer Service Controller Users Phone Users Browser
| |(1) HTTP POST | | | |(1) HTTP POST | |
| |<--------------------------------------| | |<--------------------------------------|
| |(2) HTTP 200 OK | | | |(2) HTTP 200 OK | |
| |-------------------------------------->| | |-------------------------------------->|
|(3) INVITE offer1 | | | |(3) INVITE offer1 | | |
|no media | | | |no media | | |
|<------------------| | | |<------------------| | |
|(4) 200 answer1 | | | |(4) 200 answer1 | | |
|no media | | | |no media | | |
skipping to change at page 21, line 33 skipping to change at page 27, line 45
|<------------------| | | |<------------------| | |
|(9) 200 answer2' | | | |(9) 200 answer2' | | |
|------------------>| | | |------------------>| | |
| |(10) ACK answer2 | | | |(10) ACK answer2 | |
| |------------------>| | | |------------------>| |
|(11) ACK | | | |(11) ACK | | |
|<------------------| | | |<------------------| | |
|(12) RTP | | | |(12) RTP | | |
|.......................................| | |.......................................| |
Figure 12: Click to Dial Call Flow Figure 12
o The pint solution enables calls only between two PSTN The call flow for this service is given in Figure 12. It is identical
endpoints. The solution described here allows calls between to that of Figure 4, with the exception that the service is triggered
PSTN phones (through SIP enabled gateways) and native IP through an http GET request when the user clicks on the link.
phones.
o When used for calls between two PSTN phones, the solution here We note that this service can be provided through other mechanisms,
may result in a portion of the call being routed over the namely PINT [10]. However, there are numerous differences between the
Internet. In pint, the call is always routed only over the way in which the service is provided by pint, and the way in which it
PSTN. This may result in better quality calls with the pint is provided here:
solution, depending on the codec in use and QoS capabilities
of the network routing the Internet portion of the call.
o The PINT solution requires extensions to SIP (PINT is an o The pint solution enables calls only between two PSTN endpoints.
extension to SIP), whereas the solution described here is done The solution described here allows calls between PSTN phones
with baseline SIP. (through SIP enabled gateways) and native IP phones.
o The PINT solution allows the controller (acting as a PINT o When used for calls between two PSTN phones, the solution here may
client) to "step out" once the call is established. The result in a portion of the call being routed over the Internet. In
solution described here requires the controller to maintain pint, the call is always routed only over the PSTN. This may
call state for the entire duration of the call. result in better quality calls with the pint solution, depending
on the codec in use and QoS capabilities of the network routing
the Internet portion of the call.
9.2 Mid-Call Announcement Capability o The PINT solution requires extensions to SIP (PINT is an extension
to SIP), whereas the solution described here is done with baseline
SIP.
o The PINT solution allows the controller (acting as a PINT client)
to "step out" once the call is established. The solution described
here requires the controller to maintain call state for the entire
duration of the call.
10.2 Mid-Call Announcement Capability
The third party call control mechanism described here can also be The third party call control mechanism described here can also be
used to enable mid-call announcements. Consider a service for pre- used to enable mid-call announcements. Consider a service for
paid calling cards. Once the pre-paid call is established, the system pre-paid calling cards. Once the pre-paid call is established, the
needs to set a timer to fire when they run out of minutes. When this system needs to set a timer to fire when they run out of minutes.
timer fires, we would like the user to hear an announcement which When this timer fires, we would like the user to hear an announcement
tells them to enter a credit card to continue. Once they enter the which tells them to enter a credit card to continue. Once they enter
credit card info, more money is added to the pre-paid card, and the the credit card info, more money is added to the pre-paid card, and
user is reconnected to the destination party. the user is reconnected to the destination party.
We consider here the usage of third party call control just for We consider here the usage of third party call control just for
playing the mid-call dialog to collect the credit card information. playing the mid-call dialog to collect the credit card information.
We assume the call is set up so that the controller is in the call as
a B2BUA. When the timer fires, we wish to connect the caller to a
media server. The flow for this is shown in Figure 13. When the
timer expires, the controller places the called party with a
connection address of 0.0.0.0 (1). This effectively "disconnects" the
called party. The controller then sends an INVITE without SDP to the
the pre-paid caller (4). The offer returned from the caller (5) is
used in an INVITE to the media server which will be collecting digits
(6). This is an instantiation of Flow I. This flow can only be used
here because the media server is an automata, and will answer the
INVITE immediately. If the controller was connecting the pre-paid
user with another end user, Flow III would need to be used. The media
server returns an immediate 200 OK (7) with an answer, which is
passed to the caller in an ACK (8). The result is that the media
server and the pre-paid caller have their media streams connected.
The media server plays an announcement, and prompts the user to enter
a credit card number. After collecting the number, the card number is
validated. The media server then passes the card number to the
controller (using some means outside the scope of this
specification), and then hangs up the call (11).
After hanging up with the media server, the controller reconnects the
user to the original called party. To do this, the controller sends
an INVITE without SDP to the called party (13). The 200 OK (14)
contains an offer, offer3. The controller modifies the SDP (as is
Pre-Paid User Controller Called Party Media Server Pre-Paid User Controller Called Party Media Server
| |(1) INV SDP c=0 | | | |(1) INV SDP c=invld| |
| |------------------>| | | |------------------>| |
| |(2) 200 answer1 | | | |(2) 200 answer1 | |
| |<------------------| | | |<------------------| |
| |(3) ACK | | | |(3) ACK | |
| |------------------>| | | |------------------>| |
|(4) INV no SDP | | | |(4) INV no SDP | | |
|<------------------| | | |<------------------| | |
|(5) 200 offer2 | | | |(5) 200 offer2 | | |
|------------------>| | | |------------------>| | |
| |(6) INV offer2 | | | |(6) INV offer2 | |
skipping to change at page 23, line 44 skipping to change at page 29, line 36
|<------------------| | | |<------------------| | |
|(16) 200 answer3' | | | |(16) 200 answer3' | | |
|------------------>| | | |------------------>| | |
| |(17) ACK answer3' | | | |(17) ACK answer3' | |
| |------------------>| | | |------------------>| |
|(18) ACK | | | |(18) ACK | | |
|<------------------| | | |<------------------| | |
|(19) RTP | | | |(19) RTP | | |
|.......................................| | |.......................................| |
Figure 13: Mid-Call Announcement Figure 13
We assume the call is set up so that the controller is in the call as
a B2BUA. When the timer fires, we wish to connect the caller to a
media server. The flow for this is shown in Figure 13. When the timer
expires, the controller places the called party with a connection
address of rtp.invalid (1). This effectively ``disconnects'' the
called party. The controller then sends an INVITE without SDP to the
the pre-paid caller (4). The offer returned from the caller (5) is
used in an INVITE to the media server which will be collecting digits
(6). This is an instantiation of Flow I. This flow can only be used
here because the media server is an automata, and will answer the
INVITE immediately. If the controller was connecting the pre-paid
user with another end user, Flow III would need to be used. The media
server returns an immediate 200 OK (7) with an answer, which is
passed to the caller in an ACK (8). The result is that the media
server and the pre-paid caller have their media streams connected.
The media server plays an announcement, and prompts the user to enter
a credit card number. After collecting the number, the card number is
validated. The media server then passes the card number to the
controller (using some means outside the scope of this
specification), and then hangs up the call (11).
After hanging up with the media server, the controller reconnects the
user to the original called party. To do this, the controller sends
an INVITE without SDP to the called party (13). The 200 OK (14)
contains an offer, offer3. The controller modifies the SDP (as is
done in Flow III), and passes the offer in an INVITE to the pre-paid done in Flow III), and passes the offer in an INVITE to the pre-paid
user (15). The pre-paid user generates an answer in a 200 OK (16) user (15). The pre-paid user generates an answer in a 200 OK (16)
which the controller passes to user B in the ACK (17). At this point, which the controller passes to user B in the ACK (17). At this point,
the caller and called party are reconnected. the caller and called party are reconnected.
10 Implementation Recommendations 11. Implementation Recommendations
Most of the work involved in supporting third party call control is Most of the work involved in supporting third party call control is
within the controller. A standard SIP UA should be controllable using within the controller. A standard SIP UA should be controllable using
the mechanisms described here. However, third party call control the mechanisms described here. However, third party call control
relies on a few features that might not be implemented. As such, we relies on a few features that might not be implemented. As such, we
RECOMMEND that implementors of user agent servers to support the RECOMMEND that implementors of user agent servers to support the
following: following:
o Offers and answers that contain a connection line with an address
within the .invalid TLD.
o Re-invites that change the port to which media should be sent o Re-invites that change the port to which media should be sent
o Re-invites that change the connection address o Re-invites that change the connection address
o Re-invites that add a media stream o Re-invites that add a media stream
o Re-invites that remove a media stream (setting its port to o Re-invites that remove a media stream (setting its port to zero)
zero)
o Re-invites that add a codec amongst the set in a media stream o Re-invites that add a codec amongst the set in a media stream
o SDP Connection address of zero o SDP Connection address of zero
o Initial invites with a connection address of zero o Initial invites with a connection address of zero
o Initial invites with no SDP o Initial invites with no SDP
o Initial invites with SDP but no media lines o Initial invites with SDP but no media lines
o Re-invites with no SDP o Re-invites with no SDP
o The UPDATE method [7] o The UPDATE method [7]
o Reliability of provisional responses [6] o Reliability of provisional responses [6]
o Integration of resource management and SIP [2]. o Integration of resource management and SIP [2].
11 Security Considerations 12. Security Considerations
11.1 Identity 12.1 Identity
The principal security consideration with third party call control is The principal security consideration with third party call control is
identity. When the controller initiates the call, what identity does identity. When the controller initiates the call, what identity does
it place in the From field of the INVITE? The controller could it place in the From field of the INVITE? The controller could
indicate that the call is from itself (From: indicate that the call is from itself (From:
sip:controller@example.com), but the call is really from some user, sip:controller@example.com), but the call is really from some user,
and is just facilitated by the controller. This impacts on how the and is just facilitated by the controller. This impacts on how the
call is authenticated by the end users. call is authenticated by the end users.
There are many cases, and the right one depends on the application of There are many cases, and the right one depends on the application of
3pcc. In one common scenario, the controller is acting on behalf of 3pcc. In one common scenario, the controller is acting on behalf of
one of the participants in the call. A typical example is click-to- one of the participants in the call. A typical example is
dial, where the controller and the customer service representative click-to-dial, where the controller and the customer service
are run by the same administrative domain. Indeed, for the purposes representative are run by the same administrative domain. Indeed, for
of identification, the controller can legitimately claim to be the the purposes of identification, the controller can legitimately claim
customer service representative. In this scenario, it would be to be the customer service representative. In this scenario, it would
appropriate for the INVITE to the end user to contain a From field be appropriate for the INVITE to the end user to contain a From field
identifying the customer service rep, and authenticate the request identifying the customer service rep, and authenticate the request
using S/MIME signed by the key of the customer service rep (which is using S/MIME signed by the key of the customer service rep (which is
held by the controller). held by the controller)
This requires the controller to actually have credentials with which This requires the controller to actually have credentials with which
it can authenticate itself as the customer support representative. In it can authenticate itself as the customer support representative. In
many other cases, the controller is representing one of the many other cases, the controller is representing one of the
participants, but does not possess their credentials. Unfortunately, participants, but does not possess their credentials. Unfortunately,
there are currently no standardized mechanisms that allow a user to there are currently no standardized mechanisms that allow a user to
delegate credentials to the controller in a way that limits their delegate credentials to the controller in a way that limits their
usage to specific third party call control operations. In the absence usage to specific third party call control operations. In the absence
of such a mechanisms, the best that can be done is to use the display of such a mechanisms, the best that can be done is to use the display
name in the From field to indicate the identity of the user on who's name in the From field to indicate the identity of the user on who's
behalf the call is being made. It is RECOMMENDED that the display behalf the call is being made. It is RECOMMENDED that the display
name be set to "<user> on behalf of <controller>", where user and name be set to ``[controller] on behalf of [user]'', where user and
controller are textual identities of the user and controller, controller are textual identities of the user and controller,
respectively. In this case, the URI in the From field would identify respectively. In this case, the URI in the From field would identify
the controller. the controller.
In other situations, there is no real relationship between the In other situations, there is no real relationship between the
controller and the participants in the call. In these situations, controller and the participants in the call. In these situations,
ideally the controller would have a means to assert that the call is ideally the controller would have a means to assert that the call is
from a particular identity (which could be one of the participants, from a particular identity (which could be one of the participants,
or even a third party, depending on the application), and to validate or even a third party, depending on the application), and to validate
that assertion with a signature using the key of the controller. that assertion with a signature using the key of the controller.
11.2 End-to-End Encryption and Integrity 12.2 End-to-End Encryption and Integrity
With third party call control, the controller is actually one of the With third party call control, the controller is actually one of the
participants as far as the SIP dialog is concerened. Therefore, participants as far as the SIP dialog is concerened. Therefore,
encryption and integrity of the SIP messages, as provided by S/MIME, encryption and integrity of the SIP messages, as provided by S/MIME,
will occur between participants and the controller, rather than will occur between participants and the controller, rather than
directly between participants. directly between participants.
However, end-to-end integrity, authenticity and confidentiality of However, end-to-end integrity, authenticity and confidentiality of
the media sessions can be guaranteed through a controller. End-to-end the media sessions can be guaranteed through a controller. End-to-end
media security is based on the exchange of keying material within media security is based on the exchange of keying material within SDP
SDP. For example, protocols such as MIKEY [10] can be used within [12]. The proper operation of these mechanisms with third party call
SDP. The proper operation of these mechanisms with third party call
control depends on the controller behaving properly. So long as it is control depends on the controller behaving properly. So long as it is
not attempting to explicitly disable these mechanisms, the protocols not attempting to explicitly disable these mechanisms, the protocols
will properly operate end-to-end, resulting in a secure media session will properly operate end-to-end, resulting in a secure media session
that even the controller cannot eavesdrop or modify. Since third that even the controller cannot eavesdrop or modify. Since third
party call control is based on a model of trust between the users and party call control is based on a model of trust between the users and
the controller, it is reasonable to assume it is operating in a the controller, it is reasonable to assume it is operating in a
well-behaved manner. well-behaved manner.
12 IANA Considerations 13. IANA Considerations
There are no IANA considerations associated with this specification. There are no IANA considerations associated with this specification.
13 Acknowledgements 14. Acknowledgements
The authors would like to thank Paul Kyzivat, Rohan Mahy, Eric The authors would like to thank Paul Kyzivat, Rohan Mahy, Eric
Rescorla, Allison Mankin and Sriram Parameswar for their comments. Rescorla, Allison Mankin and Sriram Parameswar for their comments.
14 Authors Addresses Normative References
[1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[2] Camarillo, G., Marshall, W. and J. Rosenberg, "Integration of
Resource Management and Session Initiation Protocol (SIP)", RFC
3312, October 2002.
[3] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[4] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[5] Schulzrinne, H., Oran, D. and G. Camarillo, "The Reason Header
Field for the Session Initiation Protocol (SIP)", RFC 3326,
December 2002.
[6] jdrosen@dynamicsoft.com and schulzrinne@cs.columbia.edu,
"Reliability of Provisional Responses in Session Initiation
Protocol (SIP)", RFC 3262, June 2002.
[7] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
Method", RFC 3311, October 2002.
[8] Eastlake, D. and A. Panitz, "Reserved Top Level DNS Names", BCP
32, RFC 2606, June 1999.
Informative References
[9] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", RFC
1889, January 1996.
[10] Petrack, S. and L. Conroy, "The PINT Service Protocol:
Extensions to SIP and SDP for IP Access to Telephone Call
Services", RFC 2848, June 2000.
[11] IANA, "Special-Use IPv4 Addresses", RFC 3330, September 2002.
[12] Baugher, M. and D. Wing, "SDP Security Descriptions for Media
Streams", draft-ietf-mmusic-sdescriptions-00 (work in
progress), February 2003.
Authors' Addresses
Jonathan Rosenberg Jonathan Rosenberg
dynamicsoft dynamicsoft
72 Eagle Rock Avenue 600 Lanidex Plaza
First Floor Parsippany, NJ 07054
East Hanover, NJ 07936
US US
email: jdrosen@dynamicsoft.com
Phone: +1 973 952-5000
EMail: jdrosen@dynamicsoft.com
URI: http://www.jdrosen.net
Jon Peterson Jon Peterson
NeuStar, Inc. Neustar
1800 Sutter St 1800 Sutter Street
Suite 570 Suite 570
Concord, CA 94520 Concord, CA 94520
US US
EMail: jon.peterson@neustar.biz
Phone: +1 925 363-8720
EMail: jon.peterson@neustar.biz
URI: http://www.neustar.biz
Henning Schulzrinne Henning Schulzrinne
Columbia University Columbia University
M/S 0401 M/S 0401
1214 Amsterdam Ave. 1214 Amsterdam Ave.
New York, NY 10027-7003 New York, NY 10027
US US
email: schulzrinne@cs.columbia.edu
EMail: schulzrinne@cs.columbia.edu
URI: http://www.cs.columbia.edu/~hgs
Gonzalo Camarillo Gonzalo Camarillo
Ericsson Ericsson Advanced Signalling Research Lab
Advanced Signalling Research Lab.
FIN-02420 Jorvas FIN-02420 Jorvas
Finland Finland
Phone: +358 9 299 3371
Fax: +358 9 299 3052
Email: Gonzalo.Camarillo@ericsson.com
15 Normative References
[1] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: session
initiation protocol," RFC 3261, Internet Engineering Task Force, June
2002.
[2] "Integration of resource management and session initiation
protocol (SIP)," RFC 3312, Internet Engineering Task Force, Oct.
2002.
[3] S. Bradner, "Key words for use in rfcs to indicate requirement
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[4] J. Rosenberg and H. Schulzrinne, "An offer/answer model with
session description protocol (SDP)," RFC 3264, Internet Engineering
Task Force, June 2002.
[5] H. Schulzrinne, D. Oran, and G. Camarillo, "The reason header
field for the session initiation protocol (SIP)," RFC 3326, Internet
Engineering Task Force, Dec. 2002.
[6] J. Rosenberg and H. Schulzrinne, "Reliability of provisional
responses in session initiation protocol (SIP)," RFC 3262, Internet
Engineering Task Force, June 2002.
[7] J. Rosenberg, "The session initiation protocol (SIP) UPDATE
method," RFC 3311, Internet Engineering Task Force, Oct. 2002.
16 Informative References
[8] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
transport protocol for real-time applications," RFC 1889, Internet
Engineering Task Force, Jan. 1996.
[9] S. Petrack and L. Conroy, "The PINT service protocol: Extensions
to SIP and SDP for IP access to telephone call services," RFC 2848,
Internet Engineering Task Force, June 2000.
[10] J. Arkko et al. , "MIKEY: multimedia Internet keying," internet EMail: Gonzalo.Camarillo@ericsson.com
draft, Internet Engineering Task Force, Feb. 2003. Work in progress.
Intellectual Property Statement Intellectual Property Statement
The IETF takes no position regarding the validity or scope of any The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described in pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it might or might not be available; neither does it represent that it
has made any effort to identify any such rights. Information on the has made any effort to identify any such rights. Information on the
IETF's procedures with respect to rights in standards-track and IETF's procedures with respect to rights in standards-track and
skipping to change at page 28, line 33 skipping to change at page 39, line 29
be obtained from the IETF Secretariat. be obtained from the IETF Secretariat.
The IETF invites any interested party to bring to its attention any The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice rights which may cover technology that may be required to practice
this standard. Please address the information to the IETF Executive this standard. Please address the information to the IETF Executive
Director. Director.
Full Copyright Statement 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
the copyright notice or references to the Internet Society or other the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than followed, or as required to translate it into languages other than
English. English.
The limited permissions granted above are perpetual and will not be The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns. revoked by the Internet Society or its successors or assignees.
This document and the information contained herein is provided on an This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
 End of changes. 

This html diff was produced by rfcdiff 1.23, available from http://www.levkowetz.com/ietf/tools/rfcdiff/