SIP WG R. Mahy
Internet-Draft Cisco Systems, Inc.
Expires: April 1, 2003 D. Petrie
Pingtel
Oct 2002
The Session Inititation Protocol (SIP) "Join" Header
draft-ietf-sip-join-00.txt
Status of this Memo
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Copyright Notice
Copyright (C) The Internet Society (2002). All Rights Reserved.
Abstract
This document defines a new header for use with SIP multi-party
applications and call control. The Join header is used to logically
join an existing SIP dialog with a new SIP dialog. This primitive
can be used to enable a variety of features, for example: "Barge-In",
answering-machine-style "Message Screening" and "Call Center
Monitoring". Note that definition of these example features is non-
normative.
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Table of Contents
1. Conventions . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Applicability of RFC2804 ("Raven") . . . . . . . . . . . . . 4
4. User Agent Server Behavior: Receiving a Join Header . . . . 5
5. User Agent Client Behavior: Sending a Join header . . . . . 7
6. Proxy behavior . . . . . . . . . . . . . . . . . . . . . . . 7
7. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
7.1 The Join Header . . . . . . . . . . . . . . . . . . . . . . 8
7.2 New option tag for Require and Supported headers . . . . . . 9
8. Usage Examples . . . . . . . . . . . . . . . . . . . . . . . 9
8.1 Join accepted and transitioned to central mixer . . . . . . 10
8.2 Join rejected . . . . . . . . . . . . . . . . . . . . . . . 11
9. Security Considerations . . . . . . . . . . . . . . . . . . 11
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . 11
10.1 Registration of "Join" SIP header . . . . . . . . . . . . . 12
10.2 Registration of "join" SIP Option-tag . . . . . . . . . . . 12
11. Changes . . . . . . . . . . . . . . . . . . . . . . . . . . 12
11.1 Changes Since draft-mahy-join-and-fork-01 . . . . . . . . . 12
11.2 Changes Since -00 . . . . . . . . . . . . . . . . . . . . . 12
12. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . 13
Normative References . . . . . . . . . . . . . . . . . . . . 13
Informational References . . . . . . . . . . . . . . . . . . 13
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 14
Full Copyright Statement . . . . . . . . . . . . . . . . . . 15
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1. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC-2119 [2].
This document refers frequently to the terms "confirmed dialog" and
"early dialog". These are defined in Section 12 of SIP [1].
2. Overview
This document describes a SIP [1] extension header field as part of
the SIP multiparty applications architecture framework [7]. The Join
header is used to logically join an existing SIP dialog with a new
SIP dialog. This is especially useful in peer-to-peer call control
environments.
One use of the "Join" header is to insert a new participant into a
multimedia conversation (which may a two-party call or a conference).
While this functionality is already available using 3rd party call
control [12] style call control, the 3pcc model requires a central
point of control which may not be desirable in many environments. As
such, a method of performing these same call control primitives in a
distributed, peer-to-peer fashion is very desirable.
Use of an explicit Join header is needed in some cases instead of
addressing an INVITE to a conference URI for the following reasons:
o A conference may not exist--the new invitation may be trying to
join an ordinary two-party call.
o The party joining may not know if the dialog it wants to join is
part of a conference.
o The party joining may not know the conference URI.
The Join header enables services such as barge-in, real-time message
screening, and call center monitoring in a distributed peer-to-peer
way. This list of services is not exhaustive.
For example, the Boss has an established 2-party conversation with a
Customer, and using some out-of-band mechanism (ex:voice, gestures,
or email) asks an Assistant to join the conversation. The Assistant
sends an INVITE with a Join header to the Boss with the dialog
information for the established dialog. The Assistant obtained this
information from some other mechanism, for example a web-page, an
instant message, or from the SIP session dialog package [8].
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Assitant Boss Customer
| callid: 4@A | callid: 7@c |
| | |
| |<============>|
| | |
|INVITE------>| |
|Join: 7@c | |
| |reINVITE----->|
|<----200-----|<----200------|
|-----ACK---->|<----ACK------|
| | |
| .. begins mixing .. |
| | |
|<===========>|<============>|
|<::::::::::::::::::::::::::>|
Note that this operation effectively creates a new conference. The
Boss needs to cause a new conference to start (and consequently
create or obtain a new conference URI). In our example, the Boss
mixes all media locally, so it needs to generate a new conference
URI, return the conference URI as the Contact to the Join INVITE, and
reINVITE the Customer with the conference URI as the new Contact.
3. Applicability of RFC2804 ("Raven")
This primitive can be used to create services which are used for
monitoring purposes, however these services do not meet the
definition of a wiretap according to RFC2804 [9]. The definition
from RFC2804 is included here:
Wiretapping is what occurs when information passed across the
Internet from one party to one or more other parties is delivered
to a third party:
1. Without the sending party knowing about the third party
2. Without any of the recipient parties knowing about the delivery
to the third party
3. When the normal expectation of the sender is that the
transmitted information will only be seen by the recipient
parties or parties obliged to keep the information in
confidence
4. When the third party acts deliberately to target the
transmission of the first party, either because he is of
interest, or because the second party's reception is of
interest.
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Specifically, item 2 of this definition does not apply to this
extension, as one party is always aware of a Join request and can
even decline such requests. In addition, in many applications of
this primitive, some or all of the other items may not apply. For
example, in many call centers which handle financial transactions,
all conversations are recorded with the full knowledge and
expectation of all parties involved.
4. User Agent Server Behavior: Receiving a Join Header
The Join header contains information used to match an existing SIP
dialog (call-id, to-tag, and from-tag). Upon receiving an INVITE
with a Join header, the UA attempts to match this information with a
confirmed or early dialog. The to-tag and from-tag are matched as if
they were present in an incoming request. In other words the to-tag
is compared to the local tag, and the from-tag is compared to the
remote tag.
If more than one Join header field is present in an INVITE, or if a
Join header field is present in a request other than INVITE, the UAS
MUST reject the request with a 400 Bad Request response.
The Join header has specific call control semantics. If both a Join
header field and another header field with contradictory semantics
(for example a Replaces [5] header field) are present in a request,
the request MUST be rejected with a 400 "Bad Request" response.
If the Join header field matches more than one dialog, the UA MUST
act as if no match is found.
If no match is found, but the Request-URI in the INVITE corresponds
to a conference URI, the UAS MUST ignore the Join header and continue
processing the INVITE as if the Join header did not exist. This
allows User Agents which receive an INVITE with Join to redirect the
request to a conference.
Otherwise if no match is found, the UAS rejects the INVITE and
returns a 481 Call/Transaction Does Not Exist response. Likewise, if
the Join header field matches a dialog which was not created with an
INVITE, the UAS MUST reject the request with an appropriate response
(ex: 400, 481, or 501).
If the Join header field matches a dialog which has already
terminated, the UA SHOULD decline the request with a 603 Declined
response.
If the Join header field matches a active dialog, the UA SHOULD
verify that the initiator of the new INVITE is authorized to join the
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matched dialog. If the initiator of the new INVITE has authenticated
successfully as equivalent to the user who is being joined, then the
join is authorized. The UA MAY also maintain a list of authorized
entities who are allowed to join any dialog with certain
characteristics (for example, all dialogs placed in the call center
context of the UA). In addition, the UA MAY use other authorization
mechanisms defined for this purpose in standards track extensions.
For example, an extension could define a mechanism for transitively
asserting authorization of a join.
If authorization is successful, the UA attempts to accept the new
INVITE, and assign any mixing or conferencing resources necessary to
complete the join. If the UA cannot accept the new INVITE (for
example: it cannot establish required QoS or keying, or it has
incompatible media), the UA MUST return an appropriate error response
and MUST leave the matched dialog unchanged.
A User Agent that accepts a Join header needs to setup dialogs or
conferences such that the requesting UAC is logically added to the
conversation space associated with the matched dialog. Any dialogs
which are already logically associated with the matched dialog in the
same conversation space are included as well. All the participants
in a conversation space should have access to all the media/content
sent in the context of that conversation space. That a participant
does not negotiate a specific type of media does not mean that it is
not otherwise a full participant. For a detailed description of
various conferencing mechanisms that could be used to handle a Join,
please consult the SIP conferencing framework [10].
If the UAS has sufficient resources to locally handle the Join
request, the UAS SHOULD accept the Join request and perform the
appropriate media mixing or combining. The UAS MAY rearrange
appropriate dialogs instead as described below, based on some local
policy.
If the UAS does not have sufficient resources locally to handle the
request, or does not wish to use these local resources, but is aware
of other resources which could be used to satisfy the request (ex: a
centralized mixer), the UA SHOULD create a conference using this
resource (ex: INVITE the centralized mixer to obtain a conference
URI), redirect the requestor to this resource, and request other
participants in the same conversation space to use this resource.
The UA MAY use any appropriate mechanism to transition participants
to the new resource (ex: 3xx repsonse, 3rd-party call control
reinvitiations, REFER requests, or reinvitations to a multicast
group). The UA SHOULD only use mechanisms which are expected to be
acceptable to the other participants. For example, the UA SHOULD NOT
attempt to transition the participants to a multicast group unless
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the UA can reasonably expect that all the particpants can support
multicast.
If the UAS is incapable of satisfying the Join request, it MUST
return a 488 "Not Acceptable Here" response.
5. User Agent Client Behavior: Sending a Join header
A User Agent that wishes to add a new dialog of its own to a single
existing early or confirmed dialog and any associated dialogs or
conferences, MAY send the target User Agent an INVITE request
containing a Join header field. The UAC places the Call-ID, to-tag,
and from-tag information for the target dialog in a single Join
header field and sends the new INVITE to the target.
If the User Agent receives a 300-class response, and acts on this
response by sending an INVITE to a Contact in the response, this
redirected INVITE MUST contain the same Join header which was present
in the original request. Although this is unusual, this allows
INVITE requests with a Join header to be redirected before reaching
the target UAS.
Note that use of the Join mechanism does not provide a way to match
multiple dialogs, nor does it provide a way to match an entire call,
an entire transaction, or to follow a chain of proxy forking logic.
For example, if Alice replaces Cathy in an early dialog with Bob, but
he does not answer, Alice's replacement request will not match other
dialogs to which Bob's UA redirects, nor other branches to which his
proxy forwards.
6. Proxy behavior
Proxy Servers do not require any new behavior to support this
extension. They simply pass the Join header field transparently as
described in the SIP specification.
Note that it is possible for a proxy (especially when forking based
on some application layer logic, such as caller screening or time-of-
day routing) to forward an INVITE request containing a Join header
field to a completely orthogonal set of Contacts than the original
request it was intended to replace. In this case, the INVITE request
with the Join header field will fail.
7. Syntax
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7.1 The Join Header
The Join header field indicates that a new dialog (created by the
INVITE in which the Join header field in contained) should be joined
with a dialog identified by the header field, and any associated
dialogs or conferences. It is a request header only, and defined
only for INVITE requests. The Join header field MAY be encrypted as
part of end-to-end encryption. Only a single Join header field value
may be present in a SIP request
This document adds the following entry to Table 3 of [1]. Additions
to this table are also provided for extension methods defined at the
time of publication of this document. This is provided as a courtesy
to the reader and is not normative in any way. SUBSCRIBE and NOTIFY,
REFER, INFO, UPDATE, and PRACK are defined respectively in [14], [4],
[15], [16], and [17].
Header field where proxy ACK BYE CAN INV OPT REG
------------ ----- ----- --- --- --- --- --- ---
Join R - - - o - -
SUB NOT REF INF UPD PRA
--- --- --- --- --- ---
Join R - - - - - -
The following syntax specification uses the augmented Backus-Naur
Form (BNF) as described in RFC-2234 [3].
Join = "Join" HCOLON callid *(SEMI join-param)
join-param = to-tag / from-tag / generic-param
to-tag = "to-tag" EQUAL token
from-tag = "from-tag" EQUAL token
A Join header MUST contain exactly one to-tag and exactly one from-
tag, as they are required for unique dialog matching. For
compatibility with dialogs initiated by RFC2543 [6] compliant UAs, a
tag of zero matches both tags of zero and null tags.
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Examples:
Join: 98732@sip.example.com
;from-tag=r33th4x0r
;to-tag=ff87ff
Join: 12adf2f34456gs5;to-tag=12345;from-tag=54321
Join: 87134@192.0.2.23;to-tag=24796;from-tag=0
7.2 New option tag for Require and Supported headers
This specification defines a new Require/Supported header option tag
"join". UAs which support the Join header MUST include the "join"
option tag in a Supported header field. UAs that want explicit
failure notification if Join is not supported MAY include the "join"
option in a Require header field.
Example:
Require: join, 100rel
8. Usage Examples
The following non-normative examples are not intended to enumerate
all the possibilities for the usage of this extension, but rather to
provide examples or ideas only. For more examples, please see
service-examples [13].
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8.1 Join accepted and transitioned to central mixer
A B C mixer
| callid: 4@A | callid: 7@c | |
| | | |
| |<============>| |
| | | |
|INVITE------>| | |
|Join: 7@c |--INVITE-------------------->|
| |<----200---------------------|
| |-----ACK-------------------->|
|<----300-----| |
|INVITE------------------------------------>|
|<--200-------------------------------------|
|---ACK------------------------------------>|
| |--REFER------>| |
| |<---200-------|--INVITE----->|
| | |<----200------|
| |<--NOTIFY-----|-----ACK----->|
| |------200---->| |
| |---BYE------->| |
| |<--200--------| |
| | | |
|<=========================================>| mixes the
| |<===========================>| three sessions
| | |<============>| together
The conversation space now looks identical to the locally mixed
example in the Introduction. Details of how the Join are implemented
are transparent to A. B could have used 3rd party call control
instead to move the necessary sessions.
[ B , C ] --> [ A , B , C ]
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8.2 Join rejected
A B C
| callid: 4@A | callid: 7@c |
| | |
| |<============>|
| | |
|INVITE------>| |
|Join: 7@c | |
| | |
|<----486-----| |
|-----ACK---->| |
| | |
In this example B is Busy (does not want to be disturbed), and
therefore does not wish to add A. B could also decline the request
with a 603 response.
9. Security Considerations
The extension specified in this document significantly changes the
relative security of SIP devices. Currently in SIP, even if an
eavesdropper learns the Call-ID, To, and From headers of a dialog,
they cannot easily modify or destroy that dialog if Digest
authentication or end-to-end message integrity are used.
This extension can be used to insert or monitor potentially sensitive
content in a multimedia conversation. As such, invitations with the
Join header SHOULD only be accepted if the peer requesting
replacement has been properly authenticated using a standard SIP
mechanism, and authorized to by joined with the target dialog.
Some mechanisms for obtaining the dialog information needed by the
Join header (Call-ID, to-tag, and from-tag) include URIs on a web
page, subscriptions to an appropriate event package, and notifcations
after a REFER request. Use of end-to-end security mechanisms to
encrypt this information is also RECOMMENDED.
This extension was designed to take advantage of future signature or
authorization schemes defined by the SIP Working Group. In general,
call control features would benefit considerably from such work.
10. IANA Considerations
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10.1 Registration of "Join" SIP header
Name of Header: Join
Short form: none
Normative description: section 7.1 of this document
10.2 Registration of "join" SIP Option-tag
Name of option: join
Description: Support for the SIP Join header
SIP headers defined: Join
Normative description: This document
11. Changes
11.1 Changes Since draft-mahy-join-and-fork-01
o Added discussion about handling of 300-class responses to an
INVITE with Join
o Fixed several typos
o Updated references
o Resubmitted as a Working Group item
11.2 Changes Since -00
o Realigned the text to mirror the outline of Replaces
o Removed the fork header
o Added a section to explain how this is not a "Raven" wiretap
mechanism
o Reorganized motivational overview material
o Added authorization language in UAS behavior section
o Updated and Added references
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12. Acknowledgments
Thanks to Robert Sparks, Alan Johnston, and Ben Campbell and many
other members of the SIP WG for their continued support of the cause
of distributed call control in SIP.
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] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 2234, November 1997.
Informational References
[4] Sparks, R., "The SIP Refer Method", draft-ietf-sip-refer-06
(work in progress), July 2002.
[5] Dean, R., Biggs, B. and R. Mahy, "The Session Inititation
Protocol (SIP) 'Replaces' Header", draft-ietf-sip-replaces-02
(work in progress), May 2002.
[6] Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
"SIP: Session Initiation Protocol", RFC 2543, March 1999.
[7] Mahy, R., "A Multi-party Application Framework for SIP", draft-
ietf-sipping-cc-framework-01 (work in progress), July 2002.
[8] Rosenberg, J. and H. Schulzrinne, "A Session Initiation
Protocol (SIP) Event Package for Dialog State", draft-ietf-
sipping-dialog-package-00 (work in progress), June 2002.
[9] IAB and IESG, "IETF Policy on Wiretapping", RFC 2804, May 2000.
[10] Rosenberg, J., "SIP Conferencing Framework", draft-rosenberg-
sipping-conferencing-framework-00.txt (work in progress), Oct
2002.
[11] Sparks, R. and A. Johnston, "SIP Call Control - Transfer",
draft-ietf-sipping-cc-transfer-00.txt (work in progress), Oct
2002.
[12] Rosenberg, J., Schulzrinne, H., Camarillo, G. and J. Peterson,
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"Best Current Practices for Third Party Call Control in the
Session Initiation Protocol", draft-ietf-sipping-3pcc-02 (work
in progress), June 2002.
[13] Johnston, A., "SIP Service Examples", draft-ietf-sipping-
service-examples-02 (work in progress), July 2002.
[14] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[15] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.
[16] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
Method", RFC 3311, October 2002.
[17] jdrosen@dynamicsoft.com and schulzrinne@cs.columbia.edu,
"Reliability of Provisional Responses in Session Initiation
Protocol (SIP)", RFC 3262, June 2002.
Authors' Addresses
Rohan Mahy
Cisco Systems, Inc.
101 Cooper Street
Santa Cruz, CA 95060
USA
EMail: rohan@cisco.com
Dan Petrie
Pingtel
400 West Cummings Park, Suite 2200
Woburn, MA 01801
USA
EMail: dpetrie@pingtel.com
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Full Copyright Statement
Copyright (C) The Internet Society (2002). All Rights Reserved.
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