wdiff draft-ietf-mmusic-sip-cc-00.txt draft-ietf-mmusic-sip-cc-01.txt

Internet Engineering Task Force MMUSIC WG
Internet Draft Schulzrinne/Rosenberg
draft-ietf-mmusic-sip-cc-00.txt
draft-ietf-mmusic-sip-cc-01.txt Columbia U./Bell Laboratories
March 13, 1998
June 17, 1999
Expires: August 1, 1998 December, 1999

SIP Call Control Services

STATUS OF THIS MEMO

This document is an Internet-Draft. Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.

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ABSTRACT can be accessed at
http://www.ietf.org/shadow.html.

Abstract

This document describes the org.ietf.sip.call a set of extensions to the Session Initiation Protocol (SIP). SIP which allow for
various call control services. Example services include blind
transfer, transfer with consultation, multi-party calls, bridged
conferences, and ad-hoc conferencing. The document
also describes how standard telephony services are supported in a
fully distributed manner, so that they can be
implemented in SIP. provided without a
central conference server. However, a SIP proxy can act as a
conference server to provide these services. For the various services
described here, we overview the requirements for the service, and
specify the protocol functions needed to support it. We then define a
basic set of SIP primitives which can be used to construct these
services, and others.

1 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 [1] and
indicate requirement levels for compliant SIP implementations.

2 Introduction

This document describes the org.ietf.sip.call extensions to the

The Session Initiation Protocol (SIP) [2]. When using the extensions
described here, [2] is a signaling protocol
used for the client MUST include initiation of multimedia sessions. SIP also defines
mechanisms for termination of sessions using the extension name in a
Require header.

3 Headers

type ACK BYE INV OPT REG
_________________________________________________________
Accept-Location R - - o o -
Also R - - o o -
Also r - - o o -
Call-Disposition R - o o - -
Replaces R - - o o -
Replaces r - - o o -
Requested-By R - - o o -
Requested-By r - - o o -

Table 1: Summary method. However,
SIP has less support for signaling of header fields. "o": optional, "m": mandatory, "-
": not applicable, "R': request header, "r": response header, "g":
general header, "*": needed if message body is not empty. A numeric
value in the "type" column indicates services that take place during
the status code call itself. These kinds of services can be broken into several
classes:

Session Control Services These services relate to the header field
is used with.

3.1 Accept-Location

The Accept-Location request header allows media session.
Examples include floor and chair control (which controls who can
send/receive data in the caller session), hold, and mute.

Call Control Services These services relate to provide
hints participant
management. These services are all built on the basic blocks of
adding and removing users from a call. Examples include transfer
(the simultaneous removal and addition of a member from a call),
multi-party calling, call bridging, and ad-hoc bridged
conferences.

This document describes extensions to SIP for providing call control
services. The services are supported in a fully distributed manner,
so that they can be provided without a central conference server.
However, a SIP proxy and redirect servers. It uses the same parameters can act as
the Location header (Section 3.4).

3.2 Also

The Also request and response header advises the recipient a conference server to issue
INVITE requests provide these
services. Our aim is to the addresses listed. Each provide a general set of these invitations
SHOULD contain tools which can be
used to construct, at a Requested-By header that contains the From field minimum, a core set of the message containing the Also field. The Also header MUST only services, but be processed
potentially useful as building blocks for future services. To
accomplish this goal, we begin by overviewing the calling or called user agent, not by any
intermediate proxy or redirect servers.

If a message contains both Also requirements for
each of the core services. This includes its basic functional
requirements and Replaces, its security requirements. Then, we overview the invitations
requested
technical issues in providing these services, and outline the Also header MUST be completed, successfully or not,
before basic
primitives that we have concluded are needed. The next section
formally defines these primitives through new headers and UA
behavior.

3 Services

We overview the terminations requested in services which we desire to support at a minimum. For
each, we define the Replaces header.

Also = "Also" ":" 1#( SIP-URL | URI ) [ comment ]
Example header:

Also: sip://jones@foo.com, sip://mueller@bar.edu

If A, B and C are end points, requirements for the following is service, with a typical scenario:

A -> B: INVITE B SIP/2.0
Call-ID: 19971214T123503.33@A
Also: C
From: A
B -> A: SIP/2.0 200 OK
Call-ID: 19971214T123503.33@A
From: A
B -> C: INVITE C SIP/2.0
Call-ID: 19971214T123503.33@A
From: B
Requested-By: A
C -> B: SIP/2.0 200 OK
Call-ID: 19971214T123503.33@A
From: B

If G particular
focus on security. Security is a group address with members X through Z, a group invitation
may proceed as follows:

A -> G: INVITE G SIP/2.0
From: A
Call-ID: 19971214T124523.00@A

G -> A: SIP/2.0 200 OK
From: A
Call-ID: 19971214T124523.00@A
Also: X, Y, Z

A -> X: INVITE X SIP/2.0
From: A
Call-ID: 19971214T124523.00@A
Requested-By: G

A -> Y: INVITE Y SIP/2.0
From: A
Call-ID: 19971214T124523.00@A
Requested-By: G

A -> Z: INVITE Z SIP/2.0
From: A
Call-ID: 19971214T124523.00@A
Requested-By: G

The Also header makes it possible to create full meshes
(generalized "three-way" calling) and supports the
resolution primary concern for many of group addresses. Unlike these
services. As such, the Location header,
which enumerates alternatives to following general principles apply:

o Parties involved in some service should be tried, the Also header
lists addresses able to be all invited.

3.3 Call-Disposition

The Call-Disposition request header field allows
cyryptographically verify the client to
indicate how identity of the server is to handle other parties in
the call. The following options
can be used singly or service

o Parties involved in some service should have a choice about
their participation in combination:

all: If the user part of service

o Parties involved in some service should know what the SIP request address identifies service
being invoked is

These three basic requirements are a group
rather than natural consequence of an individual, the " all" feature indicates
architecture where endpoints are assumed to a
proxy or redirect server be intelligent. Note,
however, that it should resolve just because the address protocol provides information and
gives choices, does not mean all implementations need to a
list take
advantage of group members this. Thin and return a 300 (Multiple Choices)
response. The list dumb endpoints can choose not to provide
information to the end users, and can choose not to provide choices
to them. This has the advantage of group members is contained enabling one common protocol for
smart and dumb endpoints alike.

3.1 Blind Transfer

In the blind transfer service, two parties are in an Also
header.

do-not-forward: The "do-not-forward" request prohibits proxies from
forwarding existing call.
One party, the transferring party , wishes to terminate the call with
the other party the transferred pary , and at the same time, transfer
them to another individual (e.g., party, the transferred-to party

----------------
-> | Transferred-to |
/ | party |
/ ----------------
/
/
-------------- ----------------
| Transferred |<------------------------| Transferring |
| party | | party |
-------------- ----------------

Figure 1: Transfer Service

Some of the requirements for this service include:

o The original call is
personal or terminates regardless of whether the caller
transfer succeeds or not.

o The transferring party does not want know whether the transfer
succeeds or not.

o The transferred party should be able to know that they are
being transferred

o The transferred party should be shunted able to a
secretary if know to whom they are
being transferred

o The transferred party should be able to decide whether to
accept or reject the line is busy.)

queue: transfer

o If the called transferred party is temporarily unreachable, e.g., because
it is in another call, rejects the caller can indicate that it wants to
have its call queued rather than rejected immediately. If transfer, the call is queued, with
the server returns "181 Queued" (see Section
4.1). A pending call be terminated by a SIP BYE request.

do-not-respond: transferring party still terminates

o The do-not-respond directive indicates transferred party should be able to verify that they are
being transferred by the callee transferring party

o The transferred-to party should know that it they are being
transferred-to

o The transferred-to party should not issue a response, informational or final.
This may be used to send invitations able to multicast groups.

Maybe know the combination identity
of do-not-respond and all could the transferring party

o The transferred-to party should be
used for group invitations to larger lists?

Call-Disposition = "Call-Disposition" ":" 1#( "all" | "do-not-forward"
| "queue" | "do-not-respond" )

Example:

Call-Disposition: all, do-not-forward, queue

3.4 Location

This document adds extension parameters able to accept or reject
the Location header.

class: The class parameter indicates whether this terminal is found
in a residential or business setting. (A caller may defer a
personal other call if only a business line

3.2 Transfer and Hold

This service is available, for
example.)

description: a variation on blind transfer. The description field further describes, as text, the
terminal. It difference is expected that
the user interface will render
this text.

duplex: The duplex parameter lists whether transferring party does not leave the terminal can
simultaneously send and receive ("full"), alternate between
sending and receiving ("half"), can only receive ("receive-
only") or only send ("send-only"). Typically, a caller will
prefer a full-duplex terminal over a half-duplex terminal and
these over receive-only or send-only terminals.

features: The feature list enumerates additional features of this
address. The "permanent" flag indicates that this address call with the transferred
party. If the service is a
new, permanent address. When used for registration, successful, the server
SHOULD return a 301 status instead of 302.

language: The language parameter lists, transferred party is
involved in order of preference, two calls - one with the
languages spoken by transferring party, and one with
the person answering. This feature may be
used transferred to have a caller automatically select party. Many of the requirements are similar. The
requirements for the appropriate
attendant or customer service representative, without having to
declare its own language skills.

media: are:

o The media tag lists call between the media types supported by transferring and transferred parties
remains active, regardless of the terminal.
Media types can be status of the standard Internet media types ("audio",
"video", "text", "application"), optionally followed by a
subtype (e.g., "text/html"). In addition, new call
between the type
"application/email" is defined.

mobility: transferred and transferred-to parties.

o The mobility parameter indicates if transferring party does not know whether the terminal is fixed
or mobile. In some locales, this may affect voice quality transfer
succeeds or
charges.

priority: not.

o The priority tag indicates the minimum priority level this
terminal is transferred party should be able to know that they are
being transferred

o The transferred party should be used for. It can able to know to whom they are
being transferred

o The transferred party should be used for automatically
restricting the choice of terminals available able to decide whether to
accept or reject the user.

service: transfer

o If the transferred party rejects the transfer, the call with
the transferring party remains unchanged

o The service tag describes what service is transferred party should be able to verify that they are
being provided transferred by the terminal.

Attributes which transferring party

o The transferred-to party should know that they are unknown being
transferred-to

o The transferred-to party should be omitted. New tags for class-
tag and service-tag can be registered with IANA. The media tag uses
Internet media types, e.g., audio, video, application/x-wb. This is
meant for indicating general communication capability, sufficient for able to know the caller identity
of the transferring party

o The transferred-to party should be able to choose an appropriate address.

Location: sip://watson@worcester.bell-telephone.com ;q=0.7
;service=IP,voice-mail
;media=audio+video+application/x-wb ;duplex=full
Location: rtsp://tape.bell-telephone.com?watson482178 ;q=0.6
;service=IP,voice-mail
;media=audio+video ;duplex=full
Location: phone://1-415-555-1212 ;q=0.5
;service=ISDN;mobility=fixed;
language=en,es,iw
Location: phone://1-800-555-1212 ;q=0.2
;service=pager;mobility=mobile;
duplex=send-only;media=text; priority=urgent;
description="For emergencies only"
Location: mailto:watson@bell-telephone.com ;q=0.1
;media=application/email
Location: http://www.bell-telephone.com/~watson ;q=0.1
;service=text/html

A 301 accept or 302 response MAY contain additional information in human-
readable form, e.g., as Content-Type: text/html. It is up to the
server issuing reject
the Location header transferred call just like any other call

o The transferred-to party should not be able to ensure consistency between ascertain,
through signaling messages, that the
content of transferring party is
still communicating with the Location header transferred party. In other
words, blind transfer and the response entity.

3.5 Replaces

The Replaces request transfer and response header hold appear identical
to the transferred-to party.

3.3 Transfer with Consultation

This service is analogous similar to blind transfer. However, the Also
header (Section 3.2, except that it asks transferring
party first contacts the recipient to issue a
BYE transferred-to party to approve the addresses listed, with transfer
through multimedia communication. Pending approval, the same Call-ID as transferring
party then simultaneosly disconnects from the request
containing transferred-to and
transferred parties, and connects the Replaces header. transferred and transferred-to
parties. The special address "*" indicates transferring and transferred parties stay connected if,
for some reason, the transfer fails.

The requirements for this service are more complex. They include:

o The transferring party should not need to know ahead of time
that they will transfer the recipient call to the transferred-to party.
In other words, it should replace all existing legs not be neccesary to know ahead of
time that the consultation call
within (between the Call-ID. If a message contains both Also transferring and Replaces,
the invitations requested in
transferred-to parties) is for the Also header MUST purposes of a transfer.

o The transferred party should be completed,
successfully able to know that they are
being transferred

o The transferred party should be able to know to whom they are
being transferred

o The transferred party should be able to decide whether to
accept or not, before the terminations requested in reject the
Replaces header.

For example, with entities A, B and M (an MCU):

A -> B: INVITE B SIP/2.0
Call-ID: 19971214T123503.33@A
From: A

B -> A: SIP/2.0 200 OK
Call-ID: 19971214T123503.33@A
From: A

M -> B: INVITE B SIP/2.0
Call-ID: 19971214T123503.33@A
From: M
Replaces: *

B -> M: SIP/2.0 200 OK
Call-ID: 19971214T123503.33@A
From: M

B -> A: BYE A SIP/2.0
Call-ID: 19971214T123503.33@A
From: B
Requested-By: M

A -> B: SIP/2.0 200 OK
Call-ID: 19971214T123503.33@A
From: A

3.6 Requested-By transfer

o The Requested-By request header is only used in requests triggered transferred party should be able to verify that they are
being transferred by Also or Replaces. It contains the URI of the entity transferring party

o The transferred-to party should know that issued they are being
transferred-to

o The transferred-to party should be able to know the request containing identity
of the Also header. transferring party

o The URI is taken from transferred-to party should be able to know that the
From header
transferred party is being transferred as a result of the request. For example, if A sends an invitation to
B containing Also: C, B issues an invitation to C
consultation call in progress with Requested-
By: A.

4 Status Code Definitions

This feature set defines one additional status code.

4.1 181 Queued the transferring party.

o The called transferred-to party was temporarily unavailable, but the caller
indicated via a "Call-Disposition: Queue" directive (Section 3.3) should be able to
queue accept or reject
the transferred call rather than reject it. When just like any other call

o If the callee becomes
available, it will return transferred-to party accepts the appropriate final status response. The
reason phrase MAY give further details about transfer, the status of
transferring party should be able to know this

o If the call,
e.g., "5 calls queued; expected waiting time is 15 minutes". The
server may issue several 181 responses transferred-to party rejects the transfer, the
transferring party should be able to update know this

o The call between the caller about transferring and transferred-to party
terminates at the
status of same time as the queued call.

5 ISDN call between the
transferring and Intelligent Network Services

SIP may transferred party, should the transfer be used
successful

This service is harder to support implement. To be done in a number distributed
manner requires that information on the success of ISDN [4] and Intelligent
Network [5] telephony services, described below. Due to the
fundamental differences call between Internet-based telephony
transferred and
conferencing as compared transferred-to parties is communicated back to public switched telephone network
(PSTN)-based services, service definitions cannot be precisely the
same. Where large differences beyond addressing and location
transferring party.

3.4 Multi-party Conferencing

Multiparty conferencing allows multiple participants to
simultaneously exchange media so that each party hears media from
every other one. There are many flavors of
implementation exist, this service.

3.4.1 Loosely Coupled Multicast Conference

In this flavor, there is a very large conference, for which multicast
is indicated below. being used to distribute the media. The term address
implies any SIP address.

This section conference is for information large enough
so that there is not a direct signaling relationship between all
parties. Session participants simply join the multicast group, and illustration only. There are many
different ways
learn about each other through RTCP [3]. This kind of implementing services conference
model is often referred to as a loosely coupled conference

The main requirement is to be able to invite another participant to
join in SIP. Since SIP this conference. In fact, this kind of conference is readily
supported by baseline SIP, as it was the initial application for it.
The only
describes new requirement is that the behavior induced by messages, different means of
implementing services called party needs some way to
know that there will interoperate.

5.1 Call Redirection not be an actual SIP session - no BYE will ever
arrive (nor should one be sent). The INVITE delivers the session
invitation, and "Number"-Translation Services

Call transfer (TRA) enables thats it. Relying on session parameters for this is
undesirable, since it leads to a user dependency between SIP behavior and
the specific session type. Furthermore, it may not be possible to transfer
ascertain from the media session whether an established (i.e.,
active) actual SIP session is
needed.

3.4.2 Distributed Full Mesh

In this conference model, each participant has a SIP signaling
session open with each other participant. The media session may be
multi-unicast or multicast. To support these conferences, the
signaling must provide support for:

o Transitioning gracefully from a normal two-party call to a third party. SIP signals
conference without knowing apriori this via the Location
header in will happen

o Adding parties to the BYE method.

Call forwarding (CF) permits conference

o Leaving the called user to forward particular
pre-selected calls to another address. Unlike telephony, conference

The requirements for the
choice service are:

o Any member of calls an existing conference can add another party to be forwarded depends on program logic
contained in any of
the SIP servers and can thus be made
dependent on time-of-day, subject of call, media types, urgency
or caller identity, rather than conference.

o The new party should know they are being restricted asked to matching
list entries. This forwarding service encompasses:

Call forwarding busy/don't answer (CFB/CFNR, SCF-BY/DA) allows the
called user join an
existing conference.

o The new party should be able to forward particular pre-selected calls if the
called user is busy accept or does not answer within a set time.

Selective call forwarding (SCF) permits reject the user to have her incoming
calls addressed
invitation to another network destination, no matter what join the called existing call.

o If the new party status is, if rejects the calling address is included
in, or excluded from, a screening list. The user's originating
service is unaffected.

Destination call routing (DCR) allows customers invitation to specify the
routing of their incoming calls conference, no
other participant should have received any messages which
indicates they were ever asked to destinations according join the conference

o The new party should be able to

- time know, within the limits of day, day
synchronization of week, etc.;

- area state across participants, the current set
of participants in the call origination;

- network address of caller;

- service attributes;

- priority (e.g., from input of a PIN before they decide whether to
reject or password);

- charge rates applicable for accept the invitation.

o Each participant in the destination;

- proportional routing of traffic.

In SIP, destination call routing is implemented by user agents, proxy
and redirect servers should learn that implement custom a new party is
being added.

o Each participant in the call handling logic, with
parameters including, but not limited should be able to
cryptographically verify that the set listed above. Caller
preferences are expressed new party has been invited
by a specific participant.

o Each participant in the Accept-Location header, callee
preferences in call should be able to decide whether
to accept or reject the Location header.

Follow-me diversion (FMD) allows new participant.

o If any existing participant in the service subscriber to remotely
control call rejects the redirection (diversion) of calls from his primary
network address to other locations.

In SIP, finding new
participant, the current network-reachable location of a callee new participant is
left not added to the location service and is outside call at
all.

o The inviting party can learn the scope success or failure of this
specification. However, users may use the REGISTER method to
appraise their "home" SIP server
addition of their the new location.

Universal access number (UAN) allows a subscriber with several
network addresses to party.

o Each participant should be reached with a single, unique address.
The subscriber may specify which incoming calls are able to know whether the new party
was successfully added or not.

o Any participant should be routed
to which address. SIP offers this functionality through proxies
and redirection.

Universal personal telecommunications (UPT) is a mobility service
which enables subscribers able to be reached with a unique personal
telecommunication number (PTN) across multiple networks leave the conference at any
network access. The PTN will
time.

o Each participant should know within a short period of time
when some other participant has left

o A participant who leaves the conference should have its SIP
signaling relationship terminated with all other participants.

o It must be translated to an appropriate
destination address possible for routing based on the capabilities
subscribed two participants to by each service subscriber. A person may have
multiple PTNs, e.g., simultaneously add
a business and private PTN. In SIP, new party to the conference.

o It must be possible for a
host-independent address participant to add another to the
conference while some other participant leaves the conference.

o The existence of the form user@domain serves as conference does not depend on the
PTN, which is translated into one or more host-dependent
addresses.

5.2 Camp-on

Completion
presence of calls to busy subscriber (CCBS) allows a calling any single user
encountering a busy destination to be informed in the conference.

o The conference terminates when the busy
destination becomes free, without having last two parties terminate
their signaling relationships.

It is important to make note that this kind of conference does not require
the use of a new call attempt.
SIP supports services close to CCBS by allowing centralized conference controller.

3.4.3 Dial-in Bridge
Another conferencing application is the "dial-up bridge". In this
scenario, a callee to indicate media bridge is used, and this device also acts as a more opportune time to call back with
centralized signaling server. Users join the Retry-After header.
Also, calling conference by "dialing-
in", which means they try and called user agents can easily record initiate a SIP session with the URI of
outgoing and incoming calls, so that
conference bridge directly. Participants do not maintain a user can re-try or return
calls signaling
session with each other. Rather, each participant maintains a single mouse click or automatically. Call-Disposition:
queue allows a caller to wait until the line becomes available. This
service is also known as a "camp-on" service.

5.3 Call Screening

Originating call screening (OCS) controls
SIP session with the ability conference bridge.

The requirements for this kind of conference are:

o It should not be neccesary for a node participant to
originate calls. In know apriori
that they are contacting a fashion similar to closed user groups, dial-up bridge - it should take
place as a
firewall would have regular SIP call.

o Participants should be able to join the conference at any time
by dialing in.

o Participants should be used able to restrict invite another participant to
join the ability conference call.

o Participants should still be able to initiate
SIP invitations outside a designated part learn, through some
means, the identity of the network. In many
cases, gateways other participants in the call.

o Participants should be able to leave the PSTN will require appropriate authentication.

5.4 Directed Call Pickup

Directed call pickup allows conference at any
time.

o When a station user to answer calls directed participant leaves or joins, this information should be
propagated to a specific address from any all other address by providing conference participants through some
means besides tones or announcements in the address
of media stream.

o It must be possible for the line conference bridge to be answered. The rung station must permit pickup. If authenticate
the call has identity of participants.

3.4.4 Ad-hoc Bridge

This service is not been answered at so much another conferencing model, as a
transition mechanism between conferencing models. A conference starts
out as a fully distributed mesh. These conferences become unwieldy as
this number of participants approaches tens to hundreds. Someone in
the ringing station, regular call
pickup occurs. If conference then decides to transition the call has been answered already, an error is
generated.

5.5 Directed Call Pickup with Barge-In

Directed call pickup with barge-in establishes to a 3-way call if conference
bridge. The bring a conference bridge into the
call has been answered at call, and then
instruct each participant to drop their signaling relationships with
the original destination.

5.6 Outgoing Call Routing

User-defined routing (UDR) allows other participants in favor of a subscriber to specify how
outgoing calls, from single signaling relationship
with the subscriber's location, shall be routed. SIP
cannot specify routing preferences; this bridge. After the transition is presumed to be handled by
a policy-based routing protocol, source routing or complete, the conference
runs similar
mechanisms. to the dial-in bridge case. However, there are some
distinctions. In the SIP Accept-Location header may be used by
proxies and redirect servers to route calls according to caller
preferences.

5.7 End-System Services

Some telephony services dialup conference, any participant can be provided by the end system, without
involvement by SIP:

Abbreviated dialing allows users to reach local subscribers join in
without
specifying being invited if they know a conference code of some sort. In
the full address (domain or host name). For SIP, ad-hoc bridge case, participants must still be actively invited.

The requirements for this service are:

o The transition must be at the
user application completes behest of one of the address to be a fully qualified
domain name.

Call waiting (CW) allows
participants.

o Any participant can cause the called party transition to receive a notification
that another party take place.

o It is trying not necessary for the protocol to reach her while she detect and resolve
simultaneous transitions. It is busy
talking assumed that the persons in
the conference would coordinate this themselves.

o The conference should continue to another calling party.

For SIP-based telephony, be operable during the called party can maintain several call
presences at
transition

o Participants should be informed of the same time, limited by local resources. Thus, transition, but it is
up to must
be possible for the called party perception to decide whether be that there has been no
change.

o It should be possible for some participants to accept another call. The
separation of resource reservation and call control may lead to the
situation that the called party accepts the incoming call, but that
transition, and appear through the network or system resource allocation fails. This cannot bridge, and for others to
remain in full mesh.

o Participants should be
completely prevented, but if able to leave the likely resource bottleneck is conference at any
time, including the
local system, the user agent may transition period.

o Participants should be able to determine whether there
are sufficient resources available or roughly track its own resource
consumption.

Consultation calling (COC) allows a subscriber to place a call on
hold, in order invite others to initiate a new call for consultation. In
systems using SIP, consultation calling can be implemented as
two separate SIP calls, possibly with the temporary release of
reserved resources for
conference, even during the call being put transition period. The mechanism
for inviting them should not depend on hold.

Customized ringing (CRG) allows the subscriber to allocate a
distinctive ringing to fact that a list
transition is taking place.

3.4.5 Conference out of calling parties. Consultation

In a SIP-based
system, this feature is offered by the service, a user application, based
on caller identification ( A has a call in progress with B, and a
separate call in progress with C. These calls are unrelated, with
different Call-ID's. From header) provided by the SIP
INVITE request.

Malicious this double call identification (MCI) allows scenario, the conference
out of consultation service subscriber to
control allows the logging (making a record) of calls received that are
of to be merged, resulting
in a malicious nature. In SIP, by default, all calls identify
the calling party and single, full-mesh conference, as described above.

The requirements for this service are:

o Only participant A can invoke the SIP servers service

o It must not be neccesary for A to know that have forwarded he will merge the
call. In addition,
two calls may be authenticated using standard
HTTP methods before any or transport-layer security. either of them is made

o It must not be neccesary for A callee may decide
only to accept have been the initiator of
the calls that are authenticated.

Multiway calling (MWC) allows the user being merged

o It must be possible to establish multiple,
simultaneous merge an arbitrary number of calls with other parties. For a SIP-based end
system, the considerations for consultation calling apply.

Terminating call screening (TCS) allows

o The participants being merged must be informed that the subscriber
merging is taking place

o A participant must be able to specify
that incoming calls either reject the merge, in which case
they are disconnected with all parties

o A participant must be restricted or allowed, according able to a screening list and/or by time verify that A was the party that
initiated the merge.

4 Discussion of day or other parameters.

5.8 Billing Features

Billing features such as account card dialing , automatic alternative
billing , credit card calling (CCC) , reverse charging , freephone
(FPH) , premium rate (PRM) and split charging are supported through
authentication. However, mechanisms for indicating billing
preferences and capabilities have not yet been specified for SIP.

Advice Implementation Options

This section discusses some of charge allows the user paying for a call to be informed of
usage-based charging information. Charges incurred by reserving
resources technical issues in the network are probably best indicated by designing a
protocol
closely affiliated with mechanism to support the reservation protocol. Advice of charge
when using Internet-to-PSTN gateways through SIP appears feasible,
but above requirements.

4.1 Transfer

For the discussion which follows, we assume the transferring party is for further study. Desirable facilities include indication of
charges at call setup time, during
A, the call transferred party is B, and at the end transferred-to party is C.

The nature of the
call
Closed user groups (CUGs) transfer service is that restrict members to communicate only
within the group can transferred party (B)
must be implemented using firewalls informed about the transfer and SIP proxies.

5.9 User-to-User Signaling

User-to-user signaling accept it before C (the
transferred-to party) is supported within SIP through contacted. This implies that the addition
of headers, with predefined header fields such as Subject or
Organization.

5.10 Operator Services

There are a number of services which involve three parties, for
example, a secretary dialing messaging
flow for boss, an auto-dialer handing the service must consist of a call message from A to a telemarketer, attended call transfer, operator services such as
person-to-person calls B, and full-mesh "multicast".

Operator services can be implemented in a number of ways, combining
Also, Replaces with either INVITE or BYE. then
B to C.

The callee's end system
does not have message from A to be cognizant of B must simultaneously disconnect A and B, and
alert B about the fact that this an operator
service. The services make use transfer. This is most readily accomplished by
including some kind of header in the Call-ID rules that stipulate BYE message which indicates that
B should initiate a new INVITE for an existing Call-ID does not alert call to C. This header is the user,
but Also header, which
is added silently.

Figure 1 shows described in greater detail in section 5.1. It contains the example
address of an autodialer connecting to a
prospective customer and, once participant, along with a signed token. This token is
the callee picks up, transfering signature over the
call to sender of the message (the From field), the
address in the Also header, and the telemarketer. (This goes Call-ID. Since C needs to show know
that SIP he is morally
neutral.)

telemarketer
T ------------. n
^ : ----> nth step; INVITE (Also:)
| : ....> nth step; BYE (Also:)
2(C) | : 4 3(
| : (
| V 5 V
.................>
A C
----------------->
auto dialer 1 customer

Figure 1: Telemarketer example

5.11 Multipoint Control Unit (MCU) Services
In the language being contacted as a result of IN services, SIP supports:

Conferencing (CON) allows a transfer, the user INVITE from
B to communicate simultaneously with
multiple parties, which may also communicate among themselves.
SIP can initiate IP multicast conferences with any number C must contain some kind of
participants, conferences where media are mixed by a conference
bridge (multipoint control unit or MCU) and, header indicating that it was A who
asked for exceptional
applications the transfer. This header needs to contain A's name along
with a small number of participants, fully-meshed
conferences, where each participant sends and receives data the authorization token from the Also header. This token allows
C to
all other participants. verify that A requested the transfer to C for this particular
call. This header is the Requested-By header, described in more greater
detail in
Sections 5.11 and 5.12.

Conference calling add-on allows a user to add and drop participants
once section 5.3.

Therefore, the conference basic transfer messaging flow is active. Participants in the SIP session
accomplish this by sending INVITE and simple. A sends a BYE requests
to the
parties B, containing an Also header listing C. The BYE causes A and B to
be added and dropped. disconnected. User B is alerted about the transfer. If A wants accepted, B to drop out of a
conference, it
sends a BYE request with " Replaces: *".

Conference calling meet-me (MMC) allows the user an INVITE to set up C, including a
conference or multi-party call, indicating the date, time,
conference duration, conference media and other parameters. The
conference session description included Requested-By header in the SIP invitation
may indicate INVITE.

4.2 Full mesh conferences

We assume the conference starts as a time standard two party call in the future. For multicast conferences,
participants do not have to connect using SIP at the actual time SIP.
One of the conference; instead, they simply subscribe parties wishes to add a third to the
multicast addresses listed in the announcement. For MCU-based
conferences, the session description may contain conference. Based on
the address of requirements, the MCU new party needs to first be called at the time of asked if they wish
to join the conference.

Some conferences use This implies that messaging begins with the
inviting party (party A) sending a multipoint control unit (MCU) message to mix, convert
and replicate media streams. While this solution has scaling
problems, it is widely deployed in traditional telephony and ISDN
conferencing settings, as so-called conference bridges. In a MCU-
based conference, the conference initiator or any authorized member
invites a new participant and indicates the address
(party B). This message must contain a list of the MCU in other
participants. If the
Also header. The invitee then contacts invitation is acceptable to B, B can begin to
join the MCU using conference. To join the same session
description conference, a signaling relationship
must be established between B and requests to all other participants. This can be added to
done by having existing participants contact B, or B contacting
existing participants. Since B has the call, just like a normal
two-party call.

Parties inviting others to a conference do not have to know that list of participants in the
conference media is managed by an MCU. The inviting party A treats
initial INVITE from A, the MCU M like another most efficient approach is to have B
contact each participant and includes it directly.

Thus, in the simplest scenario, A (who is in a call with C), sends an
INVITE to B. This INVITE contains an Also list.
The newly invited participant header, indicating C. B invites the MCU, which in turn
sends an INVITE to C, containing a Replaces Requested-By header with naming A. C
accepts, and then B sends a 200 OK to A. Now, there is a signaling
relationship between all participants. (See example parties. Adding additional parties is done
in Section
3.5). Figure 2 shows a similar fashion.

On the transition from surface, this simple mechanism appears sufficient. However, it
is not. Consider the following problematic cases (assume A,B, and C
are already in a fully-meshed conference
(see below) to an MCU-based conference):

o While A is adding D, B adds E. Since A did not tell D about E
(as it didn't know about E), D may not know of E's existence.
This results in a partially connected conference.

MCU MCU -----------,
1 ^ 2 |
Also:B / Replace:A |
/ |
/ 3 V |
A........B A.<xxxxx.B |
: : : : : ^ : ^ |
: : : : : x : x 4| Replace: A,B
: :: : : 6 x: x 5 |
: :: : : :x x |
: : : : : : x x |
: : : : : : x x |
D........C D........C <------'

----> INVITE
xxxx>

o While A is adding D, B sends a BYE

Figure 2: Transition from fully-meshed to MCU-based conference

Operator-assisted dial-out: The operator calls each participant, and
simply indicates the MCU in the Also list. The Call-ID and/or the address used group. If this BYE
is sent by B before the operator serves as the identifier INVITE from D arrives at B, B should
respond to the
MCU. For example:

O -> A: INVITE A SIP/2.0
From: O
Also: conference176@M

A -> M: INVITE conference176@M
Requested-By: O

Meet-me: The leader and with an error. As far as B is concerned,
the INVITE has failed, and it responds with an error to A.
What should A do now? It cannot tell whether the add party
failed because someone left the group, or because someone
refused to add that party. In one case, the add should be
tried again, and in the other, it should not. Even worse,
should B accept the call from D, a partially disconnected
conference will occur.

o What happens if a transfer takes place at the same time as an
add party?

o A participant leaves the conference, but fails to send a BYE
to all the other participants dial into their (either on purpose or by
accident). The result is a partially disconnected conference.

The problems can all be categorized as difficulties in synchronizing
a distributed database. The database, in this case, is the set of
participants. This database is replicated at each participant. The
database is dynamic, with each participant owning the entry in the
database corresponding to itself. As changes occur, everyone must be
quickly synchronized to achieve a consistent view of the conference
participants.

4.2.1 Approach I: Caretaker

In this approach, the party (A) that invites another (D) to the
conference is its caretaker. When A adds D, it informs D of the other
participants it knows about. D then sends an INVITE to each of these
in turn, establishing a signaling relationship. Should the
participant list (at A) change during the time D is being addded
(until a 200 OK arrives from D), A makes note of these changes, and
then propagates them to D.

The difficulty with this approach is there is no easy way for A to
know when it can cease being caretaker for D. Lets say A invited D,
and told it to contact B and C, which it did. After receiving the 200
OK from D, A receives an INVITE from E, a new party added by B. Now,
does A need to inform D about E? If B had invited E after knowing
about D, A does not have to inform E, but if B invited E before
knowing about D, A does have to inform D.

Furthermore, should the caretaker itself leave the conference, the
mechanism ceases to work. As a result, we don not believe this
approach is viable.

4.2.2 Approach II: Flooding

We make the following important observation:

synchronization of the set of participants in a fully
meshed multiparty conference is similar to the problem of
database synchronization in link state routing protocols,
like OSPF.

Based on this, we can develop mechanisms for SIP based on the same
synchronization, flooding, and adjacency notions in OSPF. We further
observe that this approach has already been used as the basis for
existing conferencing mechanisms [4].

To solve the first problem above, we introduce additional semantics
and behavior into the Also header. When A invites D into the
conference, the INVITE includes an Also header listing B and C. This
prompts D to send an INVITE to both B and C. In OSPF terminology,
this effectively establishes an adjacency between D and B, and D and
C. These INVITEs contain Also headers as well, listing the set of
participants the D believes is in the call.

When B and C receive this INVITE, they compare the set of
participants in the Also header with the set of participants they
believe are in the call. Note that this is effectively the same
operation as database synchronization in OSPF. The result is three
sets for each pair (assume B below):

S1: S1 is BD - the intersection of the set of participants B and D
both believe to be in the conference.

S2: S2 is B - BD - the set of participants B believes to be in the
conference, but D is not aware of

S3: S3 is D - BD - the set of participants D has been asked to
contact, which are not known to B

First consider S2. There are only two ways this inconsistency can
happen. The first way is that B has learned of a new participant
before A issued the add party to D. The second is that A has learned
the party has left the call before the INVITE from D arrives at B.
Unfortunately, the scheduled desired behavior is different in each case. If B
is correct, and a new party has joined, B should return the address
of the party in the 200 OK to the INVITE from D. This would prompt D,
in turn, to add those parties. On the other hand, if B is wrong, and
the party has left the conference, B should say nothing in the 200 OK
about this participant.

To enable these differing cases, we can add two additional pieces of
information to the addresses in the Also header. These are the
participant state (either active or inactive), and the version
number. When a participant receives a BYE from another, they mark
that participant as inactive, and hold onto the state for a short
duration (time TBD). This member is included in Also headers as other
participants, but they are marked as inactive. Based on this, in the
case above, B can ascertain the right behavior.

The version number satisfies a different need. What happens if the
participant that left, comes back because they are re-INVITEd? In
this case, some of the participants will think this participant is
inactive, and others will consider them active. To determine which
piece of state is correct, the version number increments each time
the state changes. The version with the highest value is always the
most recent. (TBD: who sets this? Can't always be the originator).
This is identical to the use of sequence numbers in LSA's in OSPF.

Consider now the set S3. When B receives the INVITE, this represents
the set of users D claims is in the conference, but B does not know
about. Since B keeps a cache of users who have left the conference, B
can be sure these are new participants that it has not learned of
yet. B should then send an assigned INVITE to these users to establish
signaling relationships with them. As with other INVITEs' the Also
field contains B's perspective on the set of conference identifier participants.
This is effectively the same process as flooding of new LSA's in
OSPF.

TBD: How is requested-by handled in these various cases?

We believe the flooding approach to be robust and
security code. well-proven from
many other protocols.

4.3 Dial-up Bridges

Dial up bridges are easily supported. We model them as virtual users.
When a user wants to join a dial-up conference, they send an INVITE
to the conference bridge. The bridge answers the call, and
establishes a point to point signaling relationship with the new
participant. The bridge performs the mixing locally, and sends the
mixed stream to each participant separately. As far as each
participant is concerned, they have a single signaling relationship
with one other entity - the conference server.

Fortunately, this does not prohibit each party from learning the
identity of the others in the call. The bridge is effectively an RTP
mixer. As such, it can use contributing sources (CSRC) in the RTP and
RTCP packets to identify the other participants in the call.

A -> M: user leaves the conference by hanging up with the bridge, as they
would hang up with any other user in a normal two party call.

An important issue is how conferences are identified. In the
telephone network, there is usual a dial-in number and a passcode
that the participant must know. In SIP, there are many more
possibilities:

o The conference is identified by a single URL -
sip:conference332@conferences.com, for example. A user sends
an INVITE conference189@M
From: to this address. The bridge identifies the
conference by looking at the URI in the Request-URI.

o There is a single URI for each bridge -
sip:bridge3@conferences.com. The specific conference is
identified by a passcode sent as the password in the URI:
sip:bridge3:9987097@conferences.com.

o The conference is identified by a single URL, as in the first
case. However, participants must also have a passcode. When
the server receives an INVITE for this URI, it responds with a
401 demanding digest authorization. The shared secret used for
authenticating the caller is the passcode.

o The conference is identified by a single URL, as in the first
case. The server is programmed with the public keys of those
participants allowed to join. When a participant tries to join
the conference by sending an INVITE to its address, the server
uses PGP authentication to verify the user is one of those
permitted. This allows for tight, per user controls on
conference participation.

Some have suggested identifying the conference by Call-ID. We do not
believe this is the right approach. The Call-ID represents a SIP
signaling relationship shared among two or more users. Since, in the
conference bridge case, each user has a separate signaling
relationship with the bridge, using a common Call-ID is not
appropriate.

Note that, based on this description, dial-in conferences are readily
supported in baseline SIP without any extensions. However, the
situation is more complex when a participant wishes to add another to
the conference.

We believe it is essential that the act of adding a party to a
bridged conference is no different than the act of adding a party to
a fully meshed one. Consider a bridged conference with participants
A, B, and C. Each has a signaling relationship with the bridge, X. A

5.12 Fully-Meshed Conferences
For very small
wishes to bring D into the conference. Using the same mechanisms as
for fully meshed conferences, such A sends an INVITE to D, with the Also
header indicating X. D then sends an INVITE to X, which accepts. The
result is that D has a signaling relationship with the bridge, but is
still maintaining its signaling relationship with A.

To resolve this, the bridge needs to step up and instruct D to
effectively abandon its signaling relationship with A (and vice a
versa). This does not mean the bridge wants A to send an BYE to D.
Rather, the bridge wants another one call leg to subsume another. For
D, this means that the D-X call leg should subsume the D-A call leg.
To accomplish this, the bridge sends an INVITE to D with a header
called Replaces. Replaces indicates that the call leg the INVITE
arrived on is subsuming the one identified in the header. The
Replaces contains the address of A. The request must also be
authenticated, since the Replaces header presents a powerful DOS
attack. Users should accept an INVITE with a Replaces header only
after either requesting confirmation from the user, or if the request
is signed by an authorized bridging service.

4.4 Conference out of Consultation

In this service, A is in a call with B, and separately, A is in a
call with C. These are two separate calls, and thus have identical
Call-IDs. Transitioning to a full mesh multiparty conference is
relatively straightforward. A can simply send an INVITE to B, with an
Also listing C. As far as adding B is concerned, the process is a third normal add
party.

The only difference is that the Call-ID is different in both calls.
Thus, the INVITE to C from B would not appear to be for the same
call. To resolve this, A must effectively change the Call-ID with B,
and then perform an add party. The change in Call-ID is accomplished
by having A send an INVITE to B (using the Call-ID from the A-C
call), with a Replaces header containing the A-B Call-ID and A's
address. The Replaces header has the same semantic here as in the
bridged conference case above. The call leg identified in the
Replaces header is subsumed by the call-leg of the INVITE.

Once this transition has taken place, A can send an INVITE to B,
containing Also:C, as discussed above.

If the calls being connected are multi-party calls, the situation is
more complex. (TBD: does this mechanism work for bridging two full
mesh calls?)

4.5 Ad-hoc conference bridging

To support an ad-hoc conference bridge, the following operations must
take place:

o One of the parties in the call must contact a bridge,
informing it of the set of participants

o The bridge must contact those participants, and cause them to
replace their signaling relationship with the other parties
with the relationship with the bridge

To support the first, the initiator sends a message to the bridge,
containing the list of participants. We use an INVITE method for
this, and the participants are listed in the Also headers. It is not
clear if this is the right approach. The semantics of INVITE with
Also are not the same here. The bridge is not being asked to join the
call, rather, its being asked to take over the the signaling and
media connectivity for the call. For this reason, it might be
appropriate to define a new method to indicate this, or perhaps a new
header or parameter to Also.

Once the bridge has been contacted with the list of participants, it
must send an INVITE to each (using the same Call-ID as the current
call) to establish a relationship with them. This call leg must
eventually replace the call legs the user has with all the other
users. However, the user should not subsume a call leg with some
other user until the bridge has succesfully contacted that other
user.

For this to work, the initial INVITE with each user is treated as a
normal add-party. The Also list contains those users the bridge knows
about (initially, those the initiator told the bridge about). As far
as the contacted user is concerned, a normal add party is taking
place. The response is (under normal cases) a 200 OK containing those
additional parties the contacted user knows about. This way, if a
user was in the process of an add party while someone else
transitioned to a bridge, the bridge can learn about the new party.
Should the user add parties after being contacted by the bridge, the
user will tell the new party about the bridge. This allows the bridge
to learn about all users that come (and go) during the transition
period.

Once the bridge has completed contacting all participants in the
party, it attempts to subsume the various call legs into its own call
leg. To do this, it sends another INVITE to each participant, listing
those call legs which must be subsumed. In the case where a
participant has added another user after the response to the bridges
initial INVITE was sent, but before the the "subsuming INVITE"
arrives, things still work. The new party will be informed about the
bridge, contact the bridge, and the bridge accepts. The bridge can
then send another INVITE to each user subsuming this particular new
call leg.

4.6 Transfers to Multiparty Conferences

This situation is more complex than normal transfers. We first
consider the case of a full mesh signaling relatioship. Assume A, B,
and C are already in a call. A wishes to transfer both B and C to Z.

Extending the mechanism for a single party call is the ideal choice.
In this case, A would ask B to contact Z, and A would ask C to
contact Z. Everything works fine so long as (1) both B and C perform
the transfer (i.e., both contact Z), and (2) Z accepts both B and C's
invitations. However, if these assumptions fail to hold, the
resulting transfer will only partially complete. For example, it is
possible that only A gets transferred to Z.

Whether this behavior is acceptable or not is a two- good question. We
believe that since the blind transfer mechanism has no guarantees on
success (the transferring party disconnects in either case), this
behavior is acceptable.

Another issue that arises for multiparty conference transfers is a
flooding effect at the transferred-to party. If a large number of
participants where transferred, Z would receive, in rapid succession,
an INVITE from each. To facilitate a usable application, Z should not
really prompt the user about accepting each of these parties. Rather,
it should accept them all if it accepts the first. So, we therefore
have the rule: if a user accepts a transfer, it must accept all other
parties which have been transferred.

The specific mechanism is the same for multiparty conferences. A
sends a BYE to B and C containing an Also header listing Z. B and C
send a 200 OK to the BYE, and then send an INVITE to Z. This INVITE
contains a Requested-By header listing A. When Z gets the first of
these, it alerts the user and accepts the call. (TBD: should these
triggered INVITEs contain Also's? Probably. But, in this case, Z is
going to get the first INVITE with lots of Also's. Many of these (but
perhaps not all) will eventually contact Z directly. So, should Z
send an INVITE to those in the Also headers it doesn't know about
already? Perhaps it should wait a while to see who contacts it first.
As an alternative, the BYE from A can contain Z's address, PLUS those
it send the BYE to. As a result, the INVITE from B or C to Z would
only contain those users in the Also which Z did not list in the BYE.
What is the right approach here?)

5 Header Syntax

This section defines the syntax for the three new headers defined
here - Also, Replaces, and Requested-By.

5.1 Also

The Also header is used to list other participants in a call. It is a
request and response header, and contains a list of SIP URI's, along
with some special parameters.

Also = ``Also'' ``:'' 1#Also-Values
Also-Values = name-addr [parameters]
parameters = 1*parameter
parameter = ``;'' (status-param | version-param | crypto-param)
status-param = ``status'' ``='' (``active'' | ``inactive'')
version-param = ``version'' ``='' 1*3digit
crypto-param = ``token'' ``='' token

The crypto-param is a token which is copied into the Requested-By
header for requests that are "triggered" as a result of an Also
header. The token is a signature over the URI of the entity
generating the Also header, the address in the Also header itself,
and the Call-ID. See section 6.1 for details on its computation.

5.2 Replaces

The Replaces header is used to indicate that the call leg identified
in the header is to be subsumed by the one initiated by this INVITE.
It is a request header only, valid only in INVITE messages. The
syntax is:

Replaces = ``Replaces'' ``:'' 1#Replaces-Values
Replaces-Values= SIP-URI [call-id-param]
call-id-param = ``;'' ``call-id'' ``='' token

When the call-id parameter is not present, it is presumed to be the
same as the Call-ID of the INVITE itself.

5.3 Requested-By

The Requested-By header is a request header only. It identifies the
participant who asked the UAC to send the request. The syntax is:

Requested-By = ``Requested-By'' ``:'' name-addr [req-params]
req-params = ``;'' token-param
token-param = ``token'' ``='' token

6 Also and Requested-By Header Semantics

This section overviews the detailed semantics associated with the
Also and Requested-By headers.

6.1 Sending an Untriggered INVITE

When a UAC sends an INVITE containing an Also header, without having
been asked by some other UAC to do so, it is called an untriggered
INVITE. Untriggered INVITEs are sent when a user wishes to add
another user to a call, multicast or to perform a transfer and hold service.
Other uses may exist.

An untriggered INVITE MUST NOT contain a Requested-By header. This
header is used to determine whether an INVITE is triggered or not.

When a UAC sends an untriggered INVITE containing an Also header, it
implies that the UAC wishes the recipient to send an INVITE to those
parties listed in the Also headers. If sent to a party not always already in
the call, the INVITE effects an add party operation. If sent to a
party already in a call, it affects a transfer and hold operation. To
ensure fully connected conferences, it is RECOMMENDED that a UAC
include a URI for each participant it is aware of.

Each element in the Also list should additionally contain a status
and a version parameter. If the UAC believes the participant is no
longer in the call, the status parameter is set to inactive,
otherwise its active. The version parameter contains the version of
the status for each participant that the UAC is currently aware of.

The Also header SHOULD contain a token parameter for each URI listed.
This parameter is computed in the following fashion:

1. Initialize a string to the value of the Call-ID.

2. Append the URI from the Also, not including any
displaynames, but otherwise including all URI parameters.
Also append the Also parameters status and version.

3. Append the URI that will be appropriate included in the From field of
the INVITE.

4. Append the URI that will be included in the To field of the
INVITE.

5. Compute a signature over this field, using a XXX hash and
encryption using XXX.

6. The signature is then base64 encoded. The result is the
token.

The response to the INVITE is a non-200 value if the UAS failed to
establish a call leg with all the participants listed in the Also
fields, or available. if the UAS was unwilling or unable to execute the request.

A 200 OK response means that the UAS successfully established the
call with those participants which have not already left the call. In
other words, if A sends an untriggered INVITE to B, containing C in
the Also header, B will send an INVITE to C. If C has left the call
(a fact which A did not know yet), C will respond with a specific
error code indicating this. In this fashion, B will know that it may
still respond with a 200 OK to A should all other call legs become
established. Furthermore, if other participants have joined the call
since A sent the INVITE to B, B may have established call legs to
them as well. The triggered INVITE will fail if B fails to establish
a call leg with those participants, even if they are not listed in
the Also header.

Thus, a UAC SHOULD NOT treat a 200 OK to an untriggered INVITE as an
indication that a call leg was established with all (and only) the
participants listed in the Also header.

6.2 Receiving an Untriggered INVITE

A UAS can determine whether or not an INVITE was triggered or
untriggered based on the presence of the Requested-By header.
Presence of this header means that the INVITE was triggered, and its
absence implies untriggered.

If the UAS receiving the INVITE is not currently in the call
identified by the Call-ID, the UAS is being invited to join an
existing call as a new member. The UAS SHOULD alert the user and ask
for confirmation.

If the UAS receiving the INVITE is currently in a call identified by
the Call-ID, the UAS is being transferred and held. The UAS SHOULD
alert the user and ask for confirmation.

6.2.1 New Call

The UAS SHOULD send a 100 Trying response. If the transfer or add
party request is not acceptable to the user, a 6XX response SHOULD be
sent to the UAC. If the transfer/add-party is acceptable, the UAS
MUST NOT respond definitively at this point.

Instead, the UA formulates an INVITE for each participant listed in
the Also header. Each INVITE MUST also contain a Requested-By header.
This header is formed by attaching the URI in the From field in the
INVITE to the Requested-By header. The token from the element in the
Also field is copied to the token parameter in the Requested-By
header. The URI for the Also field is copied into the To field of the
INVITE. The remaining fields are initialized as they would be for any
other INVITE sent by this UA. The INVITE's generated by the UA are
called triggered INVITEs.

The UA also formulates an internal participant list. This list
contains a set of URIs for each user, and for each, a version and
status parameter. This list is initialized to the set contained in
the Also header in the INVITE. This list is also placed into the Also
headers of each triggered INVITE. The token in the Also field is
generated as described in section 6.1. Note that this is NOT the same
token received for this Also element in the untriggered INVITE. It is
regenerated with the UA as the originator.

Each triggered INVITE is then sent. The INVITEs MAY be sent in
parallel, or MAY be sent sequentially, or MAY be sent in any
groupings deemed appropriate. However, for sake of low latencies,
sending the triggered INVITEs all at once is RECOMMENDED.

If the UA receives a response to any of these INVITEs that is not 200
or 6XX (Not in Call), the UA determines that it was not successfully
added to the call. It MUST send a BYE to those participants which:

o responded to a triggered INVITE with a 200

o have not yet responded

o sent it a triggered INVITE for the same call

The latter case occurs when inviting another party in the call (who has
received an INVITE from the UA) adds a new participant, party as well. This new
party is informed of the caller asks UA, and sends it a triggered INVITE.

The UA MUST then respond to the original untriggered INVITE with an
error code (TBD: what code?).

If a response to a triggered INVITE is a 200, this response may
contain additional Also headers. These headers contain additional
participants that the recipient of the triggered INVITE knew about,
but the UA did not. The 200 may also contain updated status on
participants the UA knew about.

The UA uses this list to update its own list of participants. New
users learned about from the 200 OK are added to the list. Users
listed in the 200 OK, which are known to the UA, but whose version
number in the 200 OK is higher, are updated.

If the resulting update generates new active members, the UA MUST
generate additional triggered INVITEs for them. The generation of
these triggered INVITEs is identical to the above process, with an
important difference. The URI in the Requested-By field is copied
from the To field in the 200 OK. 200 responses to these triggered
INVITEs may cause further triggered invites.

If the resulting update causes members to move from active to
inactive, the UA should not send them a triggered INVITE if it has
not already done so.

If the response to a triggered INVITE is a 6xx (not in call), the UA
changes the status of that member to call inactive, and increments the
version number (TBD: should this be an increment? Perhaps the 6xx
should contain the new version number).

Once responses have been received to all triggered INVITEs, all of
which were either 200 or 6xx, the UA responds to the original INVITE
(TBD: should this contain an Also list?). The UA is now in the call.

6.2.2 Existing Call

When a UA receives an INVITE containing an Also field, but no
Requested-By field, the INVITE is to transfer/hold the UA.

If the originator of the INVITE is not already in the call, the
INVITE is ignored. A 200 OK response is sent, however. (Transfers can
only take place from parties already in the call). Those users in the
Also header, who are already in the call, are ignored. If there are
no remaining users from the Also list, the INVITE is ignored.

The UA then generates triggered INVITEs to the remaining members. UA's in the
Also list. The behavior from this point forward is identical to
processing triggered INVITE responses in the previous section.

6.3 Receiving a Triggered INVITE

When a UA receives an INVITE containing a Requested-By header, it has
received a triggered INVITE. If the INVITE is for a new call, the UA
has just been transferred-to. If the INVITE is for an existing call,
the UA is being informed of a new party in this call.

6.3.1 New Call

The UA has just been transferred-to. The Requested-By header contains
the address of the transferring party. The UA SHOULD verify that the
token in the Requested-By header is valid. This will verify that the
transferring party is, in fact the one listed, and that this party
did, in fact, transfer the user listed in the From field. If the
token is not verified, the UAS SHOULD respond with a 4xx code, and
SHOULD NOT alert the user.

If the token is verified, the UA SHOULD alert the user, and ask for
confirmation. If the user rejects the transfer-to, the UAS SHOULD
send a 6xx response.

In either case, the UA MUST remember that it rejected the transfer
for this Call-ID. Subsequent triggered INVITEs for the same call MUST
be responded to with the same error response code. The UA MUST cache
its rejection of this transfer (identified by the Call-ID and URI of
the transferring party) for at least XX minutes. (TBD - what happens
if a very old INVITE arrives after the cache expires, and the user
accepts this time - we get a partial disconnect). The UA SHOULD alert
the user if it receives a triggered INVITE with a different user
listed in the Requested-By header, and MAY respond differently to
this transfer.

If the INVITE is acceptable, the UA sends a 200 OK. Processing of
subsequent triggered INVITEs (one will likely come from each
participant in the call) follows the rules below for an existing
call.

6.3.2 Existing Call

When a UA receives a triggered INVITE for an existing call, the
INVITE is an attempt to inform the participant of new members for
that call.

The UA SHOULD first verify the token. It does so by computing the
hash of the Call-ID, To address, Requested-By address, and From
address. This is compared to the decrypted value of the token using
the public key of the user listed in the Requested-By. If the two
match, the token is verified.

If the token is not verified, the INVITE is rejected with a 4xx
response. If the token is verified, the UA checks to see if the user
listed in the Requested-By is an active call participant. If they are
not, the INVITE is rejected with a 4xx response (TBD: is there a case
where the UA might not know about this participant yet?). If the user
is a participant, the INVITE is accepted. The user SHOULD NOT be
alerted.

The list of users in the Also header is then examined. If this list
contains users already known to the UA, the local list of
participants is updated if the version number is higher. If the list
contains users not known to the UA, they are added to the local list
of participants.

The UA then computes a difference set between its updated list and
the list in the Also header. This set includes any users in its local
list and not in the Also list. The set also includes users in both
lists, but whose version is higher in the local list. This set is
included in the Also header in the 200 OK to the INVITE. The token in
the 200 OK is generated as described in 6.1.

The UA then computes a second difference set between its updated list
and the list in the Also header. This set includes any users in the
Also list not in its local list. The set also includes users in both
lists, but whose version is higher than in the local list. The active
users from this set are then sent triggered INVITEs. The Requested-By
and Also fields in these triggered INVITEs are computed as described
above. The inactive users in this set are then sent triggered BYE's.
The Requested-By and Also fields in the triggered BYEs are computed
in the same fashion as triggered INVITEs, except a triggered BYE
contains no Also fields.

6.4 Sending an untriggered BYE

A UA MAY send a BYE, containing Also headers, at any time. This BYE
simulataneously terminates a call leg with the recipient, and causes
the recipient to attempt to set up a call leg to the parties listed
in the Also header. Unlike the INVITE, the BYE response is sent
immediately, without first adding the various parties. Sending an
untriggered BYE is equivalent to a blind transfer.

The Also headers in the untriggered BYE MUST contain tokens. These
tokens are generated in the same way described in section 6.1.

6.5 Receiving an untriggered BYE

If the BYE corresponds to an existing call leg, the UA sends a 200 OK
to the BYE. If it does not, it sends a 481.

The UA then generates triggered INVITEs to all participants listed in
the Also field. Generation of the triggered INVITEs, and processing
of their responses, is done in the same.

6 Acknowledgements

Parameters same fashion as described in
section 6.1. The difference is, of course, that no additional
response is sent to the terminal negotiation mechanism BYE.

6.6 Receiving a triggered BYE

If the BYE doesn't correspond to an existing call leg, the UA sends a
481. The UA then validates the token in the Location Requested-By header. If
it is validated, a 200 OK is sent to the BYE, and the call-leg is
torn down.

7 Replaces header semantics

The Replaces header is used to allow one call leg to subsume another.
The new call leg is identified by the combination of To, From, and
Call-ID in the INVITE carrying the Replaces header. Replaces is a
request header only, and MUST appear only in INVITEs. A UAS receiving
a Replaces header in a non-INVITE request MUST respond with a 4xx
status code.

The request containing a Replaces header SHOULD be authenticated.

The Replaces header contains a list of call-legs, identified by the
URI of the remote party and a Call-ID. If any of these are not valid
call-legs as known to the UAS, the INVITE MUST be responded to with a
4xx status code. Otherwise, the UAS "abandons" each call leg listed -
acting as if it had never been established. No BYE is sent. A 200 OK
is then sent to the client.

If a BYE additionally contains Also headers, the UAS MUST first
generate the triggered INVITEs implied by the Also headers. Only if
all triggered INVITEs succeed should the UAS act on the Replaces
header.

8 Example Call Flows

This section illustrates some example call flows. Messages are of the
form:

INV B Also:C,D
BYE A Also:Y

Where INV implies an INVITE request, and BYE a BYE request. The
letter after the method is the Request URI. Also:C,D implies that
URI's C and D were influenced in the Also header.

8.1 Basic Transfer

Figure 2 exemplifies the basic transfer in a two party call. A first
sets up a call to B, and then transfers B to C.

8.2 Basic Add Party

Figure 3 exemplifies the basic add party. A and B are already in a
call. A adds C to the call.

8.3 Add Party during Add Party

In this example (Figure 4), A and B are in a call. A adds another
party, C, while B adds a different party, D. In the example, B adds D
before learning about C.

A B C
INV
----------------->

200 OK
<----------------

ACK
----------------->

BYE B Also:C
------------------>

200 OK
<------------------ INV C ReqBy:A
--------------------->

200 OK
<---------------------

ACK
--------------------->

Figure 2: Transfer Message Flow

In the example, C acts on the untriggered INVITE, and sends a
triggered INVITE to B. B responds with a 200 OK, also alerting it to
D's new presence in the call. D, acting on its untriggered INVITE,
sends a triggered INVITE to A, and learns about C. Now, both C and D
know about each other. In the example, C sends the INVITE to D first.
It is possible in other cases for D to send the INVITE to C first, or
for both INVITEs cross each other on the wire (in this case, both
sides back off with a 500 and a Retry-After, so that eventually one
invitation reaches the other side without an invite in transit in the
other direction).

Having received an INVITE from C, D doesn't bother to INVITE C. Both
D and C then OK their respective INVITEs.

8.4 Party leaves during add party

In this example (Figure 5), a three party call is in place between A,

A B C

INV C Also:B
---------------------------------->
INV B Also:C ReqBy:A
<-------------------
200 OK
-------------------->
ACK
<--------------------
200 OK
<-----------------------------------
ACK
----------------------------------->

Figure 3: Add Party Message Flow

B and C. A adds another user, D, and shortly thereafer, C leaves the
call.

Since D learns from B that C has left the call, D does not bother to
contact C, and responds right away to the add party. The result is
now a three party call with A,B, and D.

9 A note on multicast media

Another useful service, which we have not discussed so far, is to
transition a conference from distributing media through multi-unicast
to distribution through multicast. In fact, this is not a SIP issue
at all. However, we discuss it here for completeness.

Assume a call between A, B, and C. Media is being distributed through
multi-unicast. At some point, A decides its appropriate to transition
to multicast. It should send a re-INVITE to B and C, containing an
updated SDP with a multicast group (allocated by Scott Petrack's CMA design.

7 A by some means,
perhaps MADCAP [5]. If the transition to multicast is acceptable,
both B and C respond with a 200 OK. No SDP is needed in the response,
as per [2].

If B and C decide to switch to multicast, it is in their interest
(but not required) to send a re-INVITE to the other participants they

A B C D

INV C Also:B
---------------------------------->
INV D Also:A
--------------------------------->
INV B Also:A ReqBy:A
<----------------
200 OK Also:D
----------------->
INV A Also:A,B
<--------------------------------------------------
200 OK Also:C
--------------------------------------------------->
INV D Also:A,B ReqBy:B
------------------>
200 OK
<------------------
ACK
------------------->
200 OK
<-----------------------------------
ACK
----------------------------------->
200 OK
<-----------------
ACK
------------------>

Figure 4: Add Party During Add Party Message Flow

know about, containing the SDP describing the multicast session. The
result is that some or all of the sessions on the call-legs
transition to multicast. If not all have transitioned, the user may
still need to send some packets unicast.

There is no capability for determining the codec parameters for the
multicast session based on the intersection of the capabilities of
each participant. The model for multicast media distribution in a
tightly coupled conference is identical to that for loosely coupled
sessions. The initiator of the multicast session chooses a codec, and
that is what is used. Note, however, that in the case where the

A B C D

INV D Also:B,C
-------------------------------------------------->
BYE B
<-----------------
BYE A
<--------------------------------
200 OK
----------------->
200 OK
-------------------------------->
INV B Also:A,B,C ReqBy:A
<-----------------------------------
200 OK Also:C;status=inactive
----------------------------------->
ACK
<-----------------------------------
200 OK
<--------------------------------------------------
ACK
-------------------------------------------------->

Figure 5: Party Leaves During Add Message Flow

sessions start as multi-unicast, the originator will know the
capabilities of all the other parties, and thus can intelligently
choose the codecs for the session.

10 Security Considerations

Security issues are addressed throughout this document.

The call control mechanisms have serious security issues. An INVITE
with an Also cause the recipient to add or drop other parties,
possibly without user interaction. This implies that authorization of
the requests is critical.

11 Open Issues

There are many, many open issues:

1. How to do this with shared secrets rather than public keys?

2. If the transferred-to party in a transfer accepts some, but
not all (or rejects some, but not all) of the INVITEs for
it, we end up with a partially disconnected conference.

3. Should we use a session timer to refresh things and
periodically re-flood the participant list, in an attempt
to keep things synchronized?

4. The version/status concept is still very vague. Does it
work? Is it needed?

5. Conference out of consultation for multi-party calls - not
clear the Replaces mechanism works here.

12 Acknowledgements

The authors would like to especially thank Jonathan Lennox for his
many insightful comments and contributions to this work.

13 Bibliography

[1] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," RFC Request for Comments (Best Current Practice) 2119, Internet
Engineering Task Force, Mar. 1997.

[2] M. Handley, H. Schulzrinne, and E. Schooler, and J. Rosenberg, "SIP: Session
session initiation protocol," Internet Draft, Request for Comments (Proposed
Standard) 2543, Internet Engineering Task Force, Nov. 1997. Work in progress. Mar. 1999.

[3] M. Handley H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "SDP: Session description "RTP: a
transport protocol for real-time applications," Request for Comments
(Proposed Standard) 1889, Internet Engineering Task Force, Jan. 1996.

[4] C. Elliott, "A 'sticky' conference control protocol," vol. 5, pp.
97--119, 1994.

[5] S. Hanna, B. Patel, and M. Shah, "Multicast address dynamic
client allocation protocol (MADCAP)," Internet Draft, Internet
Engineering Task Force, Mar. 1997. May 1999. Work in progress.

[4] International Telecommunication Union, "Integrated services
digital network (ISDN) service capabilities -- definition

Full Copyright Statement

This document and translations of
supplementary services," Recommendation I.250, Telecommunication
Standardization Sector of ITU, Geneva, Switzerland, 1993.

[5] International Telecommunication Union, "General recommendations it may be copied and furnished to
others, and derivative works that comment on telephone switching or otherwise explain it
or assist in its implementation may be prepared, copied, published
and signaling -- intelligent network:
Introduction distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to intelligent network capability set 1," Recommendation
Q.1211, Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, Mar. 1993. the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.

The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
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Table of Contents

1 Terminology ......................................... 1
2 Introduction ........................................ 1 2
3 Headers ............................................. Services ............................................ 2
3.1 Accept-Location .................................... 2 Blind Transfer ...................................... 3
3.2 Also ............................................... 2
3.3 Call-Disposition Transfer and Hold ................................... 4
3.4 Location ...........................................
3.3 Transfer with Consultation .......................... 5
3.5 Replaces ...........................................
3.4 Multi-party Conferencing ............................ 6
3.4.1 Loosely Coupled Multicast Conference ................ 6
3.4.2 Distributed Full Mesh ............................... 7
3.6 Requested-By .......................................
3.4.3 Dial-in Bridge ...................................... 8
3.4.4 Ad-hoc Bridge ....................................... 9
3.4.5 Conference out of Consultation ...................... 10
4 Status Code Definitions ............................. 8 Discussion of Implementation Options ................ 11
4.1 181 Queued .......................................... 8 Transfer ............................................ 11
4.2 Full mesh conferences ............................... 12
4.2.1 Approach I: Caretaker ............................... 13
4.2.2 Approach II: Flooding ............................... 13
4.3 Dial-up Bridges ..................................... 15
4.4 Conference out of Consultation ...................... 17
4.5 Ad-hoc conference bridging .......................... 17
4.6 Transfers to Multiparty Conferences ................. 18
5 ISDN and Intelligent Network Services ............... 8 Header Syntax ....................................... 19
5.1 Call Redirection and "Number"-Translation Services
................................................................ 9 Also ................................................ 19
5.2 Camp-on ............................................. 10 Replaces ............................................ 20
5.3 Requested-By ........................................ 20
6 Also and Requested-By Header Semantics .............. 20
6.1 Sending an Untriggered INVITE ....................... 20
6.2 Receiving an Untriggered INVITE ..................... 22
6.2.1 New Call Screening ...................................... 10
5.4 Directed ............................................ 22
6.2.2 Existing Call Pickup ................................ 11
5.5 Directed ....................................... 24
6.3 Receiving a Triggered INVITE ........................ 24
6.3.1 New Call Pickup with Barge-In .................. 11
5.6 Outgoing ............................................ 24
6.3.2 Existing Call Routing ............................... 11
5.7 End-System Services ................................. ....................................... 25
6.4 Sending an untriggered BYE .......................... 26
6.5 Receiving an untriggered BYE ........................ 26
6.6 Receiving a triggered BYE ........................... 26
7 Replaces header semantics ........................... 26
8 Example Call Flows .................................. 27
8.1 Basic Transfer ...................................... 27
8.2 Basic Add Party ..................................... 27
8.3 Add Party during Add Party .......................... 27
8.4 Party leaves during add party ....................... 28
9 A note on multicast media ........................... 29
10 Security Considerations ............................. 31
11
5.8 Billing Features .................................... Open Issues ......................................... 31
12
5.9 User-to-User Signaling .............................. 13
5.10 Operator Services ................................... 13
5.11 Multipoint Control Unit (MCU) Services .............. 13
5.12 Fully-Meshed Conferences ............................ 15
6 Acknowledgements .................................... 16
7 32
13 Bibliography ........................................ 16 32