SIPPING J. Rosenberg
Internet-Draft dynamicsoft
Expires: December 29, 2003 J. Peterson
Neustar
H. Schulzrinne
Columbia University
G. Camarillo
Ericsson Advanced Signalling
Research Lab
June 30, 2003
Best Current Practices for Third Party Call Control in the Session
Initiation Protocol
draft-ietf-sipping-3pcc-04
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
Third party call control refers to the ability of one entity to
create a call in which communications is actually between other
parties. Third party call control is possible using the mechanisms
specified within the Session Initiation Protocol (SIP). However,
there are several possible approaches, each with different benefits
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and drawbacks. This document discusses best current practices for the
usage of SIP for third party call control.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . 5
4. 3pcc Call Establishment . . . . . . . . . . . . . . . . . . 6
4.1 Flow I . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.2 Flow II . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.3 Flow III . . . . . . . . . . . . . . . . . . . . . . . . . . 9
4.4 Flow IV . . . . . . . . . . . . . . . . . . . . . . . . . . 11
5. Recommendations . . . . . . . . . . . . . . . . . . . . . . 13
6. Error Handling . . . . . . . . . . . . . . . . . . . . . . . 14
7. Continued Processing . . . . . . . . . . . . . . . . . . . . 16
8. 3pcc and Early Media . . . . . . . . . . . . . . . . . . . . 18
9. Third Party Call Control and SDP Preconditions . . . . . . . 21
9.1 Controller Initiates . . . . . . . . . . . . . . . . . . . . 21
9.2 Party A Initiates . . . . . . . . . . . . . . . . . . . . . 23
10. Example Call Flows . . . . . . . . . . . . . . . . . . . . . 27
10.1 Click to Dial . . . . . . . . . . . . . . . . . . . . . . . 27
10.2 Mid-Call Announcement Capability . . . . . . . . . . . . . . 28
11. Implementation Recommendations . . . . . . . . . . . . . . . 31
12. Security Considerations . . . . . . . . . . . . . . . . . . 32
12.1 Identity . . . . . . . . . . . . . . . . . . . . . . . . . . 32
12.2 End-to-End Encryption and Integrity . . . . . . . . . . . . 32
13. IANA Considerations . . . . . . . . . . . . . . . . . . . . 34
14. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 35
Normative References . . . . . . . . . . . . . . . . . . . . 36
Informative References . . . . . . . . . . . . . . . . . . . 37
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 37
Intellectual Property and Copyright Statements . . . . . . . 39
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1. Introduction
In the traditional telephony context, third party call control allows
one entity (which we call the controller) to set up and manage a
communications relationship between two or more other parties. Third
party call control (referred to as 3pcc) is often used for operator
services (where an operator creates a call that connects two
participants together), and conferencing.
Similarly, many SIP services are possible through third party call
control. These include the traditional ones on the PSTN, but also new
ones such as click-to-dial. Click-to-dial allows a user to click on a
web page when they wish to speak to a customer service
representative. The web server then creates a call between the user
and a customer service representative. The call can be between two
phones, a phone and an IP host, or two IP hosts.
Third party call control is possible using only the mechanisms
specified within RFC 3261 [1]. Indeed, many different call flows are
possible, each of which will work with SIP compliant user agents.
However, there are benefits and drawbacks to each of these flows. The
usage of third party call control also becomes more complex when
aspects of the call utilize SIP extensions or optional features of
SIP. In particular, the usage of RFC 3312 [2] (used for coupling of
signaling to resource reservation) with third party call control is
non-trivial. Similarly, the usage of early media (where session data
is exchanged before the call is accepted) with third party call
control is not trivial.
This document serves as a best current practice for implementing
third party call control without usage of any extensions specifically
designed for that purpose. Section 4 presents the known call flows
that can be used to achieve third party call control, and provides
guidelines on their usage. Section 9 discusses the interactions of
RFC 3312 [2] with third party call control. Section 8 discusses the
interactions of early media with third party call control. Section 10
provides example applications that make usage of the flows
recommended here.
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2. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [3] and
indicate requirement levels for compliant implementations.
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3. Definitions
The following terms are used throughout this document:
3pcc: Third Party Call Control, which refers to the general ability
to manipulate calls between other parties.
Controller: A controller is a SIP User Agent that wishes to create a
session between two other user agents.
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4. 3pcc Call Establishment
The primary primitive operation of third party call control is the
establishment of a session between participants A and B.
Establishment of this session is orchestrated by a third party,
referred to as the controller.
This section documents three call flows that the controller can
utilize in order to provide this primitive operation.
4.1 Flow I
A Controller B
|(1) INVITE no SDP | |
|<------------------| |
|(2) 200 offer1 | |
|------------------>| |
| |(3) INVITE offer1 |
| |------------------>|
| |(4) 200 OK answer1 |
| |<------------------|
| |(5) ACK |
| |------------------>|
|(6) ACK answer1 | |
|<------------------| |
|(7) RTP | |
|.......................................|
Figure 1
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
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
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
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
an ACK sent to it (6). Because the offer was generated by A, and the
answer generated by B, the actual media session is between A and B.
Therefore, media flows between them (7).
This flow is simple, requires no manipulation of the SDP by the
controller, and works for any media types supported by both
endpoints. However, it has a serious timeout problem. User B may not
answer the call immediately. The result is that the controller cannot
send the ACK to A right away. This causes A to retransmit the 200 OK
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response periodically. As specified in RFC 3261 Section 13.3.1.4, the
200 OK will be retransmitted for 64*T1 seconds. If an ACK does not
arrive by then, the call is considered to have failed. This limits
the applicability of this flow to scenarios where the controller
knows that B will answer the INVITE immediately.
4.2 Flow II
A Controller B
|(1) INVITE bh sdp1 | |
|<------------------| |
|(2) 200 sdp2 | |
|------------------>| |
| |(3) INVITE sdp2 |
| |------------------>|
|(4) ACK | |
|<------------------| |
| |(5) 200 OK sdp3 |
| |<------------------|
| |(6) ACK |
| |------------------>|
|(7) INVITE sdp3 | |
|<------------------| |
|(8) 200 OK sdp2 | |
|------------------>| |
|(9) ACK | |
|<------------------| |
|(10) RTP | |
|.......................................|
Figure 2
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,
containing an offer (sdp1) with a single audio media line, one codec,
a random port number (but not zero), and a connection address that is
somewhere within the .invalid DNS top level domain (TLD) [8] (for
example, rtp.invalid). The use of a connection address in the invalid
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, with a valid
address in the connection line.
A user agent should respond to an offer containing a .invalid
domain just as it would any other offer. The answer contains its
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own address in the connection line. However, due to the presence
of a purposefully invalid domain name, media would not be sent to
the offerer.
The controller sends an ACK (4). It then generates a second INVITE
(3). 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 2), it was an answer, but in the INVITE, it is an offer.
Fortunatly, all valid answers are valid initial offers. This INVITE
causes B's phone to ring. When it answers, it generates a 200 OK (5)
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
ACKed. It therefore does not suffer from the timeout and message
inefficiency problems of flow 1. However, it too has troubles. First
off, it requires that the controller know the media types to be used
for the call (since it must generate a ``blackhole'' SDP, which
requires media lines). Secondly, the first INVITE to A (1) contains
media with a rtp.invalid connection address to indicate that neither
media nor RTCP packets should be sent. This mechanism for indicating
that neither media or RTCP should be sent is different the one
documented in RFC 3264 [4], which uses 0.0.0.0 for this purpose.
However, according to RFC 3330 [11], 0.0.0.0 is a special case
address which refers to hosts on this network. This is not the same
as the desired semantic, which is much closer to the meaning of the
.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
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
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
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
having very smart UAs which can return the same SDP whenever
possible, or really smart controllers that can analyze the SDP to
determine if a re-INVITE is really needed. However, we wish to keep
this mechanism simple, and avoid SDP awareness in the controller. As
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a result, this flow is not really workable. It is therefore NOT
RECOMMENDED.
4.3 Flow III
A Controller B
|(1) INVITE no SDP | |
|<---------------------| |
|(2) 200 offer1 | |
|--------------------->| |
|(3) ACK answer1 (bh) | |
|<---------------------| |
| |(4) INVITE no SDP |
| |--------------------->|
| |(5) 200 OK offer2 |
| |<---------------------|
|(6) INVITE offer2' | |
|<---------------------| |
|(7) 200 answer2' | |
|--------------------->| |
| |(8) ACK answer2 |
| |--------------------->|
|(9) ACK | |
|<---------------------| |
|(10) RTP | |
|.............................................|
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
(which is good, since it means that the controller doesn't need to
assume anything about the media composition of the session). A's
phone rings. When A answers, a 200 OK is generated (2) containing its
offer, offer1. The controller generates an immediate ACK containing
an answer (3). This answer is a ``black hole'' SDP, with its
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
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
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
example, if offer1 contained an audio and a video line, in that
order, but offer2 contained just an audio line, the controller would
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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
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,
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),
and then media can flow.
This flow has many benefits. First, it will usually operate without
any spurious retransmissions or timeouts (although this may still
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
the participants.
There are some drawbacks. The controller does need to perform SDP
manipulations. Specifically, it must take some SDP, and generate
another SDP which has the same media composition, but has connection
addresses within the .invalid TLD. This is needed for message 3.
Secondly, it may need to reorder and trim on SDP X, so that its media
lines match up with those in some other SDP, Y. Thirdly, the offer
from B (offer2) may have no codecs or media streams in common with
the offer from A (offer 1). The controller will need to detect this
condition, and terminate the call. Fourth, it requires usage of the
.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).
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4.4 Flow IV
A Controller B
|(1) INVITE offer1 | |
|no media | |
|<---------------------| |
|(2) 200 answer1 | |
|no media | |
|--------------------->| |
|(3) ACK | |
|<---------------------| |
| |(4) INVITE no SDP |
| |--------------------->|
| |(5) 200 OK offer2 |
| |<---------------------|
|(6) INVITE offer2' | |
|<---------------------| |
|(7) 200 answer2' | |
|--------------------->| |
| |(8) ACK answer2 |
| |--------------------->|
|(9) ACK | |
|<---------------------| |
|(10) RTP | |
|.............................................|
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
offer1 (validify requires that the version increments by one, and
that the other parameters remain unchanged).
There are some limitations associated with this flow. First, user A
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
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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
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
have already been alerted, resulting in user annoyance and possibly
resulting in billing events.
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5. Recommendations
Flow I (Figure 1) represents the simplest and the most efficient
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
call immediately. This is the case for devices such as media servers,
conferencing servers, and messaging servers, for example. Since we
expect a great deal of third party call control to be to automata,
special caseing this scenario is reasonable.
For calls to unknown entities, or to entities known to represent
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
additional benefits over Flow IV. However, Flow II SHOULD NOT be
used, because of the potential for infinite ping-ponging of
re-INVITEs.
Several of these flows use a ``black hole'' connection address, which
is any domain in the .invalid domain. A UA MUST be prepared to
receive offers or answers with connection lines containing addresses
within this domain. A UA receiving such a domain proceeds with normal
RFC 3264 offer/answer processing. However, it will not send media to
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.
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6. Error Handling
There are numerous error cases which merit discussion.
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.
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
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
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
inform A of the precise reason for the failure. The information is
important from a user interface perspective. For example, if A was
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
local busy signal so that A knows that B is busy.
A Controller B
|(1) INVITE offer1 | |
|no media | |
|<---------------------| |
|(2) 200 answer1 | |
|no media | |
|--------------------->| |
|(3) ACK | |
|<---------------------| |
| |(4) INVITE no SDP |
| |--------------------->|
| |(5) 180 |
| |<---------------------|
|(6) INVITE offer2 | |
|--------------------->| |
|(7) 491 | |
|<---------------------| |
|(8) ACK | |
|--------------------->| |
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
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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
the 491), and this may occur before the exchange with B is completed.
In that case, the controller would respond with another 491.
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7. Continued Processing
Once the calls are established, both participants believe they are in
a single point-to-point call. However, they are exchanging media
directly with each other, rather than with the controller. The
controller is involved in two dialogs, yet sees no media.
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
the participants, it can create a new BYE and hang up with the other
participant. This is shown in Figure 6.
A Controller B
|(1) BYE | |
|------------------>| |
|(2) 200 OK | |
|<------------------| |
| |(3) BYE |
| |------------------>|
| |(4) 200 OK |
| |<------------------|
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
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
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
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
IV.
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A Controller B C
|(1) BYE | | |
|--------------->| | |
|(2) 200 OK | | |
|<---------------| | |
| |(3) INV no media| |
| |-------------------------------->|
| |(4) 200 no media| |
| |<--------------------------------|
| |(5) ACK | |
| |-------------------------------->|
| |(6) INV no SDP | |
| |--------------->| |
| |(7) 200 offer3 | |
| |<---------------| |
| |(8) INV offer3' | |
| |-------------------------------->|
| |(9) 200 answer3'| |
| |<--------------------------------|
| |(10) ACK | |
| |-------------------------------->|
| |(11) ACK answer3| |
| |--------------->| |
| | |(12) RTP |
| | |................|
Figure 7
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).
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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.
A Controller B
| | |
|(1) INVITE offer1 | |
|no media | |
|<---------------------| |
| | |
|<ring> | |
| | |
|<answer> | |
| | |
|(2) 200 answer1 | |
|no media | |
|--------------------->| |
|(3) ACK | |
|<---------------------| |
| |(4) INVITE no SDP |
| |--------------------->|
| | |<ring>
| |(5) 183 offer2 |
| |<---------------------|
|(6) INVITE offer2' | |
|<---------------------| |
|(7) 200 answer2' | |
|--------------------->| |
|(8) ACK | |
|<---------------------| |
| |(9) PRACK answer2 |
| |--------------------->|
| |(10) 200 PRACK |
| |<---------------------|
|(11) RTP | |
|.............................................|
| | |<answer>
| |(12) 200 OK |
| |<---------------------|
| |(13) ACK |
| |--------------------->|
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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.
A Controller B
| | |
|(1) INVITE offer1 | |
|no media | |
|<---------------------| |
| | |
|ring | |
| | |
|(2) 183 answer1 | |
|no media | |
|--------------------->| |
|(3) PRACK | |
|<---------------------| |
|(4) 200 PRACK | |
|--------------------->| |
| |(5) INVITE no SDP |
| |--------------------->|
| | |ring
| | |
| | |answer
| | |
| |(6) 200 OK offer2 |
| |<---------------------|
|(7) UPDATE offer2' | |
|<---------------------| |
|answer | |
| | |
|(8) 200 answer2' | |
|--------------------->| |
| |(9) ACK answer2 |
| |--------------------->|
|(10) RTP | |
|.............................................|
|(11) 200 OK | |
|--------------------->| |
|(12) ACK | |
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|<---------------------| |
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
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
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
call flow. User A is likely a PSTN gateway, and has generated a
provisional response because of early media from the PSTN side. The
PSTN will deliver this media even though the gateway does not have
anywhere to send it, since the initial offer from the controller had
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
will be lost.
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9. Third Party Call Control and SDP Preconditions
A SIP extension has been specified that allows for the coupling of
signaling and resource reservation [2]. This specification relies on
exchanges of session descriptions before completion of the call
setup. These flows are initiated when certain SDP parameters are
passed in the initial INVITE. As a result, the interaction of this
mechanism with third party call control is not obvious, and worth
detailing.
9.1 Controller Initiates
In one usage scenario, the controller wishes to make use of
preconditions in order to avoid the call failure scenarios documented
in Section 4.4. Specifically, the controller can use preconditions in
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
information on the media composition of the call before they decide
to accept it.
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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
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.
The controller sends an INVITE to user A containing no SDP, but with
a Require header indicating that preconditions are required. This
specific scenario (an INVITE without an offer, but with a Require
header indicating preconditions) is not described in [2]. It is
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
preconditions it supports as optional. Of course, the user is not
alerted at this time. The controller takes this offer and passes it
to user B (3). User B does not support preconditions, or does, but is
not interested in them. Therefore, when it answers the call, the 200
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OK contains an answer without any preconditions listed (4). This
answer is passed to user A in the PRACK (6). At this point, user A
knows that there are no preconditions actually in use for the call,
and therefore, it can alert the user. When the call is answered, user
A sends a 200 OK to the controller (8) and the call is complete.
In the event that the offer generated by user A was not acceptable to
user B (because of non-overlapping codecs or media, for example),
user B would immediately reject the INVITE (message 3). The
controller would then CANCEL the request to user A. In this
situation, neither user A nor user B would have been alerted,
achieving the desired effect. It is interesting to note that this
property is achieved using preconditions even though it doesn't
matter what specific types of preconditions are supported by user A.
It is also entirely possible that user B does actually desire
preconditions. In that case, it might generate a 1xx of its own with
an answer containing preconditions. That answer would still be passed
to user A, and both parties would proceed with whatever measures are
necessary to meet the preconditions. Neither user would be alerted
until the preconditions were met.
9.2 Party A Initiates
In Section 9.1, the controller requested the use of preconditions to
achieve a specific goal. It is also possible that the controller
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
case is shown in Figure 11.
A Controller B
|(1) INVITE offer1 | |
|no media | |
|<---------------------| |
|(2) 183 answer1 | |
|no media | |
|--------------------->| |
|(3) PRACK | |
|<---------------------| |
|(4) 200 OK | |
|--------------------->| |
| |(5) INVITE no SDP |
| |--------------------->|
| |(6) 183 offer2 |
| |des=sendrecv |
| |conf=recv |
| |cur=none |
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| |<---------------------|
|(7) UPDATE offer2' | |
|des=sendrecv | |
|conf=recv | |
|cur=none | |
|<---------------------| |
|(8) 200 UPDATE | |
|answer2' | |
|des=sendrecv | |
|conf=recv | |
|cur=none | |
|--------------------->| |
| |(9) PRACK answer2 |
| |des=sendrecv |
| |conf=recv |
| |cur=none |
| |--------------------->|
| |(10) 200 PRACK |
| |<---------------------|
|(11) reservation | |
|-------------------------------------------->|
|(12) reservation | |
|<--------------------------------------------|
|(13) UPDATE offer3 | |
|des=sendrecv | |
|conf=recv | |
|cur=recv | |
|--------------------->| |
| |(14) UPDATE offer3' |
| |des=sendrecv |
| |conf=recv |
| |cur=recv |
| |--------------------->|
| |(15) 200 UPDATE |
| |answer3' |
| |des=sendrecv |
| |conf=recv |
| |cur=send |
| |<---------------------|
|(16) 200 UPDATE | |
|answer3 | |
|des=sendrecv | |
|conf=recv | |
|cur=send | |
|<---------------------| |
| | |<ring>
| |(17) UPDATE offer4 |
| |des=sendrecv |
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| |conf=recv |
| |cur=sendrecv |
| |<---------------------|
|(18) UPDATE offer4' | |
|des=sendrecv | |
|conf=recv | |
|cur=sendrecv | |
|<---------------------| |
|<ring> | |
|(19) 200 UPDATE | |
|answer4' | |
|des=sendrecv | |
|conf=recv | |
|cur=sendrecv | |
|--------------------->| |
| |(20) 200 UPDATE |
| |answer4 |
| |des=sendrecv |
| |conf=recv |
| |cur=sendrecv |
| |--------------------->|
|(21) 180 INVITE | |
|--------------------->| |
| |(22) 180 INVITE |
| |<---------------------|
|<answer> | |
|(23) 200 INVITE | |
|--------------------->| |
|(24) ACK | |
|<---------------------| |
| | |<answer>
| |(25) 200 INVITE |
| |<---------------------|
| |(26) ACK |
| |--------------------->|
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).
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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
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
cannot use a re-INVITE. User A sees that its peer is capable of
supporting preconditions. Since it desires preconditions for the
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
response (9). Now, both sides perform resource reservation. User A
succeeds first, and passes an updated session description in an
UPDATE request (13). The controller simply passes this to A (after
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
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),
and then both answer (23 and 25), completing the call.
What is important about this flow is that the controller doesn't know
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
when needed. That determination is made entirely based on the offer/
answer rules described in [6] and [7], and is independent of
preconditions.
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10. Example Call Flows
10.1 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
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
representative. When the representative picks up, the phone on the
user's desk rings. When they pick up, the customer service
representative is there, ready to talk to the user.
Customer Service Controller Users Phone Users Browser
| |(1) HTTP POST | |
| |<--------------------------------------|
| |(2) HTTP 200 OK | |
| |-------------------------------------->|
|(3) INVITE offer1 | | |
|no media | | |
|<------------------| | |
|(4) 200 answer1 | | |
|no media | | |
|------------------>| | |
|(5) ACK | | |
|<------------------| | |
| |(6) INVITE no SDP | |
| |------------------>| |
| |(7) 200 OK offer2 | |
| |<------------------| |
|(8) INVITE offer2' | | |
|<------------------| | |
|(9) 200 answer2' | | |
|------------------>| | |
| |(10) ACK answer2 | |
| |------------------>| |
|(11) ACK | | |
|<------------------| | |
|(12) RTP | | |
|.......................................| |
Figure 12
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,
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namely PINT [10]. 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:
o The pint solution enables calls only between two PSTN endpoints.
The solution described here allows calls between PSTN phones
(through SIP enabled gateways) and native IP phones.
o When used for calls between two PSTN phones, the solution here may
result in a portion of the call being routed over the Internet. In
pint, the call is always routed only over the PSTN. This may
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.
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
used to enable mid-call announcements. Consider a service for
pre-paid calling cards. Once the pre-paid call is established, the
system needs to set a timer to fire when they run out of minutes.
When this timer fires, we would like the user to hear an announcement
which tells them to enter a credit card to continue. Once they enter
the credit card info, more money is added to the pre-paid card, and
the user is reconnected to the destination party.
We consider here the usage of third party call control just for
playing the mid-call dialog to collect the credit card information.
Pre-Paid User Controller Called Party Media Server
| |(1) INV SDP c=invld| |
| |------------------>| |
| |(2) 200 answer1 | |
| |<------------------| |
| |(3) ACK | |
| |------------------>| |
|(4) INV no SDP | | |
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|<------------------| | |
|(5) 200 offer2 | | |
|------------------>| | |
| |(6) INV offer2 | |
| |-------------------------------------->|
| |(7) 200 answer2 | |
| |<--------------------------------------|
|(8) ACK answer2 | | |
|<------------------| | |
| |(9) ACK | |
| |-------------------------------------->|
|(10) RTP | | |
|...........................................................|
| |(11) BYE | |
| |-------------------------------------->|
| |(12) 200 OK | |
| |<--------------------------------------|
| |(13) INV no SDP | |
| |------------------>| |
| |(14) 200 offer3 | |
| |<------------------| |
|(15) INV offer3' | | |
|<------------------| | |
|(16) 200 answer3' | | |
|------------------>| | |
| |(17) ACK answer3' | |
| |------------------>| |
|(18) ACK | | |
|<------------------| | |
|(19) RTP | | |
|.......................................| |
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
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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
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,
the caller and called party are reconnected.
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11. Implementation Recommendations
Most of the work involved in supporting third party call control is
within the controller. A standard SIP UA should be controllable using
the mechanisms described here. However, third party call control
relies on a few features that might not be implemented. As such, we
RECOMMEND that implementors of user agent servers to support the
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 connection address
o Re-invites that add a media stream
o Re-invites that remove a media stream (setting its port to zero)
o Re-invites that add a codec amongst the set in a media stream
o SDP Connection address of zero
o Initial invites with a connection address of zero
o Initial invites with no SDP
o Initial invites with SDP but no media lines
o Re-invites with no SDP
o The UPDATE method [7]
o Reliability of provisional responses [6]
o Integration of resource management and SIP [2].
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12. Security Considerations
12.1 Identity
The principal security consideration with third party call control is
identity. When the controller initiates the call, what identity does
it place in the From field of the INVITE? The controller could
indicate that the call is from itself (From:
sip:controller@example.com), but the call is really from some user,
and is just facilitated by the controller. This impacts on how the
call is authenticated by the end users.
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
one of the participants in the call. A typical example is
click-to-dial, where the controller and the customer service
representative are run by the same administrative domain. Indeed, for
the purposes of identification, the controller can legitimately claim
to be the customer service representative. In this scenario, it would
be appropriate for the INVITE to the end user to contain a From field
identifying the customer service rep, and authenticate the request
using S/MIME signed by the key of the customer service rep (which is
held by the controller)
This requires the controller to actually have credentials with which
it can authenticate itself as the customer support representative. In
many other cases, the controller is representing one of the
participants, but does not possess their credentials. Unfortunately,
there are currently no standardized mechanisms that allow a user to
delegate credentials to the controller in a way that limits their
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
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
name be set to ``[controller] on behalf of [user]'', where user and
controller are textual identities of the user and controller,
respectively. In this case, the URI in the From field would identify
the controller.
In other situations, there is no real relationship between the
controller and the participants in the call. In these situations,
ideally the controller would have a means to assert that the call is
from a particular identity (which could be one of the participants,
or even a third party, depending on the application), and to validate
that assertion with a signature using the key of the controller.
12.2 End-to-End Encryption and Integrity
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With third party call control, the controller is actually one of the
participants as far as the SIP dialog is concerened. Therefore,
encryption and integrity of the SIP messages, as provided by S/MIME,
will occur between participants and the controller, rather than
directly between participants.
However, end-to-end integrity, authenticity and confidentiality of
the media sessions can be guaranteed through a controller. End-to-end
media security is based on the exchange of keying material within SDP
[12]. The proper operation of these mechanisms with third party call
control depends on the controller behaving properly. So long as it is
not attempting to explicitly disable these mechanisms, the protocols
will properly operate end-to-end, resulting in a secure media session
that even the controller cannot eavesdrop or modify. Since third
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
well-behaved manner.
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13. IANA Considerations
There are no IANA considerations associated with this specification.
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14. Acknowledgements
The authors would like to thank Paul Kyzivat, Rohan Mahy, Eric
Rescorla, Allison Mankin and Sriram Parameswar for their comments.
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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.
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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
dynamicsoft
600 Lanidex Plaza
Parsippany, NJ 07054
US
Phone: +1 973 952-5000
EMail: jdrosen@dynamicsoft.com
URI: http://www.jdrosen.net
Jon Peterson
Neustar
1800 Sutter Street
Suite 570
Concord, CA 94520
US
Phone: +1 925 363-8720
EMail: jon.peterson@neustar.biz
URI: http://www.neustar.biz
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Henning Schulzrinne
Columbia University
M/S 0401
1214 Amsterdam Ave.
New York, NY 10027
US
EMail: schulzrinne@cs.columbia.edu
URI: http://www.cs.columbia.edu/~hgs
Gonzalo Camarillo
Ericsson Advanced Signalling Research Lab
FIN-02420 Jorvas
Finland
EMail: Gonzalo.Camarillo@ericsson.com
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Rosenberg, et al. Expires December 29, 2003 [Page 40]
