One document matched: draft-ietf-sip-manyfolks-resource-00.txt
SIP Working Group W. Marshall
Internet Draft AT&T
Document: <draft-ietf-sip-manyfolks-resource-00>
K. Ramakrishnan
TeraOptic Networks
E. Miller
G. Russell
CableLabs
B. Beser
Pacific Broadband
M. Mannette
K. Steinbrenner
3Com
D. Oran
F. Andreasen
M. Ramalho
Cisco
J. Pickens
Com21
P. Lalwaney
Nokia
J. Fellows
Motorola
D. Evans
D. R. Evans Consulting
K. Kelly
NetSpeak
A. Roach
Ericsson
J. Rosenberg
D. Willis
S. Donovan
dynamicsoft
H. Schulzrinne
Columbia University
November, 2000
Integration of Resource Management and SIP
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Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026[1].
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts. Internet-Drafts are draft documents valid for a maximum of
six months and may be updated, replaced, or obsoleted by other
documents at any time. It is inappropriate to use Internet- Drafts
as reference material or to cite them other than as "work in
progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
The distribution of this memo is unlimited. It is filed as <draft-
ietf-sip-manyfolks-resource-00.txt>, and expires May 31, 2001.
Please send comments to the authors.
1. Abstract
This document discusses how network QoS and security establishment
can be made a precondition to sessions initiated by the Session
Initiation Protocol (SIP), and described by SDP. These preconditions
require that the participant reserve network resources (or establish
a secure media channel) before continuing with the session. We do
not define new QoS reservation or security mechanisms; these pre-
conditions simply require a participant to use existing resource
reservation and security mechanisms before beginning the session.
This results in a multi-phase call-setup mechanism, with the
resource management protocol interleaved between two phases of call
signaling. The objective of such a mechanism is to enable deployment
of robust IP Telephony services, by ensuring that resources are made
available before the phone rings and the participants of the call
are "invited" to participate.
This document also proposes an extension to the Session Initiation
Protocol (SIP) to add a new COMET method, which is used to confirm
the completion of all pre-conditions by the session originator.
2. Conventions used in this document
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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[4].
3. Introduction
For an Internet Telephony service to be successfully used by a large
number of subscribers, it must offer few surprises to those
accustomed to the behavior of existing telephony services. One
expectation is that of connection quality, implying resources must
be set aside for each call.
A key contribution of the DCS architecture, as described in [5], is
a recognition of the need for coordination between call signaling,
which controls access to telephony specific services, and resource
management, which controls access to network-layer resources. This
coordination is designed to meet the user expectations and human
factors associated with telephony.
While customers may expect, during times of heavy load, to receive a
"fast busy" or an announcement saying "all circuits are busy now,"
the general expectation is that once the destination phone rings
that the connection can be made. Blocking a call after ringing the
destination is considered a "call defect" and is a very undesirable
exception condition.
This draft addresses both "QoS-Assured" and "QoS-Enabled" sessions.
A "QoS-Assured" session will complete only if all the required
resources are available and assigned to the session. A provider may
choose to block a call when adequate resources for the call are not
available. Public policy demands that the phone system provide
adequate quality at least in certain cases: e.g., for emergency
communications during times of disasters. Call blocking enables a
provider to meet such requirements.
A "QoS-Enabled" session allows the endpoints to complete the session
establishment either with or without the desired resources. Such
session will use dedicated resources if available, and use a best-
effort connection as an alternative if resources can not be
dedicated. In cases where resources are not available, the
originating and/or terminating User Agent might consult with the
customer to obtain guidance on whether the session should complete.
Coordination between call signaling and resource management is also
needed to prevent fraud and theft of service. The coordination
between call-signaling and QoS setup protocols ensures that users
are authenticated and authorized before receiving access to the
enhanced QoS associated with the telephony service.
This coordination, referred to in this draft as "preconditions,"
require that the participant reserve network resources (or establish
a secure media channel) before continuing with the session. We do
not define new QoS reservation or security mechanisms; these pre-
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conditions simply require a participant to use existing resource
reservation and security mechanisms before beginning the session.
In the case of SIP [2], this effectively means that the "phone won't
ring" until the preconditions are met. These preconditions are
described by new SDP parameters, defined in this document. The
parameters can mandate end-to-end QoS reservations based on RSVP [6]
or any other end-to-end reservation mechanism (such as YESSIR [7],
or PacketCable's Dynamic Quality of Service (D-QoS) [8]), and
security based on IPSEC [9]. The preconditions can be defined
independently for each media stream.
The QoS architecture of the Internet separates QoS signaling from
application level signaling. Application layer devices (such as web
proxies and SIP servers) are not well suited for participation in
network admission control or QoS management, as this is
fundamentally a network layer issue, independent of any particular
application. In addition, since application devices like SIP servers
are almost never on the "bearer path" (i.e., the network path the
RTP [10] takes), and since the RTP path and signaling paths can be
completely different (even traversing different autonomous systems),
these application servers are generally not capable of managing QoS
for the media. Keeping QoS out of application signaling also means
that there can be a single infrastructure for QoS across all
applications. This eliminates duplication of functionality, reducing
management and equipment costs. It also means that new applications,
with their own unique QoS requirements, can be easily supported.
This loose coupling works very well for a wide range of
applications. For example, in an interactive game, one can establish
the game using an application signaling protocol, and then later on
use RSVP to reserve network resources. The separation is also
effective for applications which have no explicit signaling.
However, certain applications may require tighter coupling. In the
case of Internet telephony, the following is an important
requirement:
When A calls B, B's phone should not ring unless resources
have been reserved from A to B, and B to A.
This could be achieved without coupling if A knew B's address, port,
and codecs before the telephony signaling took place. However, since
telephony signaling is used largely to obtain this information in
the first place, the coupling cannot be avoided.
A similar model exists for security. Rather than inventing new
security mechanisms for each new application, common security tools
(such as IPSEC) can be used across all applications. As with QoS, a
means in application level protocols is needed to indicate that a
security association is needed for the application to execute.
To solve both of these problems, we propose an extension to SDP
which allows indication of pre-conditions for sessions. These
preconditions indicate that participation in the session should not
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proceed until the preconditions are met. The preconditions we define
are (1) success of end-to-end resource reservation, and (2) success
of end- to-end security establishment. We chose to implement these
extensions in SDP, rather than SIP [2] or SAP [11], since they are
fundamentally a media session issue. SIP is session agnostic;
information about codecs, ports, and RTP [10] are outside the scope
of SIP. Since it is the media sessions that the reservations and
security refer to, SDP is the appropriate venue for the extensions.
Furthermore, placement of the extensions in SDP rather than SIP or
SAP allows specification of preconditions for individual media
streams. For example, a multimedia lecture might require reservation
for the audio, but not the video (which is less important).
Our extensions are completely backward compatible. If a recipient
does not understand them, normal SIP or SAP processing will occur,
at no penalty of call setup latency.
3.1 Requirements
The basic motivation in this work is to meet and possibly exceed the
user expectations and human factors associated with telephony.
In this section, we briefly describe the application requirements
that led to the set of DCS signaling design principles. In its
basic implementation, DCS supports a residential telephone service
comparable to the local telephone services offered today. Some of
the requirements for resource management, in concert with call
signaling, are as follows:
The system must minimize call defects. These are situations where
either the call never completes, or an error occurs after the
destination is alerted. Requirements on call defects are typically
far more stringent than call blocking. Note that we expect the
provider and the endpoints to attempt to use lower bandwidth CODECs
as the first line of defense against limited network capacity, and
to avoid blocking calls.
The system must minimize the post-dial-delay, which is the time
between the user dialing the last digit and receiving positive
confirmation from the network. This delay must be short enough that
users do not perceive a difference with post-dial delay in the
circuit switched network or conclude that the network connectivity
no longer exists.
Call signaling needs to provide enough information to the resource
management protocol so as to enable resources to be allocated in the
network. This typically requires most if not all of the components
of a packet classifier (source IP, destination IP, source port,
destination port, protocol) to be available for resource allocation.
3.2 Overview
For acceptable interactive voice communication it is important to
achieve end-to-end QoS. The end-to-end QoS assurance implies
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achieving low packet delay and packet loss. End-to-end packet delay
must be small enough that it does not interfere with normal voice
conversations. The ITU recommends no greater than 300 ms roundtrip
delay for telephony service. Packet loss must be small enough to
not perceptibly impede voice quality or the performance of fax and
voice band modems.
If it is found that the network cannot guarantee end-to-end QoS
resources, there are two alternatives: either (1) allow call
signaling to proceed with high probability of excessive delay and
packet loss which could impair any interactive real-time
communication between the participants, or (2) block the call prior
to the called party being alerted. When calls are blocked because
of a lack of resources in a particular segment of the network, it is
highly desirable that such blocking occur before the called party
picks up.
We do expect the endpoints to attempt to use lower bandwidth CODECs,
thereby avoiding blocking calls, as the first line of defense
against limited network capacity.
The call signaling and resource reservation must be achieved in such
a way that the post-dial-delay must be minimized without increasing
the probability of call defects. This means that the number of
round-trip messages must be kept at an absolute minimum and messages
must be sent directly end-system to end-system if possible.
The general idea behind the extension is simple. We define two new
SDP attributes, "qos" and "security". The "qos" attribute indicates
whether end-to-end resource reservation is optional or mandatory,
and in which direction (send, recv, or sendrecv). When the attribute
indicates mandatory, this means that the participant who has
received the SDP does not proceed with participation in the session
until resource reservation has completed in the direction indicated.
In this case, "not proceeding" means that the participant behaves as
if they had not received the SDP at all. If the attribute indicates
that QoS for the stream is optional, then the participant proceeds
normally with the session, but should reserve network resources in
the direction indicated, if they are capable. Absence of the "qos"
attribute means the participant reserves resources for this stream,
and proceeds normally with the session. This behavior is the normal
behavior for SDP.
Resource reservation takes place using whatever protocols
participants must use, based on support by their service provider.
If the ISP's of the various participants are using differing
resource reservation protocols, translation is necessary, but this
is done within the network, without knowledge of the participants.
The direction attribute indicates in which direction reservations
should be reserved. If "send", it means reservations should be made
in the direction of media flow from the session originator to
participants. If "recv", it means reservations should be made in the
direction of media flow from participants to the session originator.
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In the case of "sendrecv", it means reservations should be made in
both directions. If the direction attribute is "sendrecv" but the
endpoints only support a single-direction resource reservation
protocol, then both the originator and participants cooperate to
ensure the agreed precondition is met.
In the case of security, the same attributes are defined -
optional/mandatory, and send/recv/sendrecv. Their meaning is
identical to the one above, except that a security association
should be established in the given direction. The details of the
security association are not signaled by SDP; these depend on the
Security Policy Database of the participant.
Either party can include a "confirm" attribute in the SDP. When the
"Confirm" attribute is present, the recipient sends a COMET message
to the sender, with SDP attached, telling the status of each
precondition as "success" or "failure." If the "confirm" attribute
is present in the SDP sent by the session originator to the
participant (e.g. in the SIP INVITE message), then the participant
sends the COMET message to the originator. If the "confirm"
attribute is present in the SDP sent by the recipient to the
originator (e.g. in a SIP response message), then the originator
sends the COMET message to the participant.
When the "Confirm" attribute is present in both the SDP sent by the
session originator to the participant (e.g. in the SIP INVITE
message), and in the SDP sent by the recipient back to the
originator (e.g. in a SIP response message), the session originator
would wait for the COMET message with the success/failure
notification before responding with a COMET message, and responds
instead with a CANCEL if a mandatory precondition is not met, or if
a sufficient combination of optional preconditions are not met. The
recipient does not wait for the COMET message from the originator
before sending its COMET message.
The "confirm" attribute is typically used if the direction attribute
is "sendrecv" and the originator or participant only supports a
single-direction resource reservation protocol. In such a case, the
originator (or participant) would reserve resources for one
direction of media flow, and send a confirmation with a direction
attribute of "send." The participant (or originator) would reserve
resources for the other direction. On receipt of the COMET message,
they would know that both directions have been reserved, and the
precondition is met.
4. SDP Extension
The formatting of the qos attribute in the Session Description
Protocol (SDP)[3] is described by the following BNF:
qos-attribute = "a=qos:" strength-tag SP direction-tag
[SP confirmation-tag]
strength-tag = ("mandatory" | "optional" | "success" |
"failure")
direction-tag = ("send" | "recv" | "sendrecv")
confirmation-tag = "confirm"
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and the security attribute:
security-attribute = "a=secure:" SP strength-tag SP direction-tag
[SP confirmation-tag]
4.1 SDP Example
The following example shows an SDP description carried in a SIP
INVITE message from A to B:
v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
m=audio 49170 RTP/AVP 0
a=qos:mandatory recv confirm
m=video 51372 RTP/AVP 31
a=secure:mandatory sendrecv
m=application 32416 udp wb
a=orient:portrait
a=qos:optional sendrecv
a=secure:optional sendrecv
This SDP indicates that B should not continue its involvement in the
session until resources for the audio are reserved from B to A, and
a bi-directional security association is established for the video.
B can join the sessions once these preconditions are met, but should
reserve resources and establish a bidirectional security association
for the whiteboard.
4.2 SDP Allowable Combinations
If the recipient of the SDP (e.g. the UAS) is capable and willing to
honor the precondition(s), it returns a provisional response
containing SDP, along with the qos/security attributes, for each
such stream. This SDP MUST be a subset of the preconditions
indicated in the INVITE.
Table 1 illustrates the allowed values for the direction tag in the
response. Each row represents a value of the direction in the SIP
INVITE, and each column the value in the response. An entry of N/A
means that this combination is not allowed. A value of A->B (B->A)
implies that the precondition is for resources reserved (or security
established) from A to B (B to A). A value of A<->B means that the
precondition is for resource reservation or security establishment
in both directions. The value in the response is the one used by
both parties.
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B: response
A: request send recv sendrecv none
send N/A A->B N/A --
recv B->A N/A N/A --
sendrecv A<-B B<-A A<->B --
none -- -- -- --
Table 1: Allowed values of coupling
Table 2 illustrates the allowed values for the strength tag in the
request and response. A "Y" means the combination is allowed, and a
"N" means it is not. The value in the response is the one used by
both parties.
B: response
A: request mandatory optional none
mandatory Y Y Y
optional N Y Y
none N N Y
Table 2: Allowed values of strength parameter
Table 3 illustrates the allowed values for the direction tag in a
confirmation message (COMET) sent from the originator to a
participant, based on the value of the direction tag negotiated in
the initial request and response. A "Y" means the combination is
allowed, and a "N" means it is not and SHOULD be ignored by the
participant.
A: confirmation
B: reponse send recv sendrecv
A->B Y N N
A<-B N Y N
A<->B Y Y Y
Table 3: Allowed values of direction in confirmation from A
Table 4 illustrates the allowed values for the direction tag in a
confirmation message (COMET) sent from the participant to the
originator, based on the value of the direction tag negotiated in
the initial request and response. A "Y" means the combination is
allowed, and a "N" means it is not and SHOULD be ignored by the
originator.
B: confirmation
B: reponse send recv sendrecv
A->B N Y N
A<-B Y N N
A<->B Y Y Y
Table 4: Allowed values of direction in confirmation from B
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5. SIP Extension: The COMET Method
The COMET method is used for communicating successful completion of
preconditions from the calling to called user agents.
The signaling path for the COMET method is the signaling path
established as a result of the call setup. This can be either
direct signaling between the calling and called user agents or a
signaling path involving SIP proxy servers that were involved in the
call setup and added themselves to the Record-Route header on the
initial INVITE message.
The precondition information is communicated in the message body,
which MUST contain an SDP. For every agreed precondition, the
strength-tag MUST indicate "success" or "failure".
If the initial request contained Record-Route headers, the
provisional response MUST contain a copy of those headers, as if the
response were a 200 OK to the initial request. Since provisional
responses MAY contain Record-Route and Contact headers, the COMET
request MUST contain Route headers if the Record-Route headers were
present in the provisional response. The Route header is constructed
as specified in [2]. The Route header that is constructed from some
provisional response MUST NOT be placed in any other request except
for the COMET for that provisional response.
A UAC MUST NOT insert a Route header into a COMET request if no
Record-Route header was present in the response.
If the initial request was sent with credentials, the COMET request
SHOULD contain those credentials as well.
The Call-ID in the COMET MUST match that of the provisional
response. The CSeq in this request MUST be larger than the last
request sent by this UAC for this call leg. The To, From, and Via
headers MUST be present, and MUST be constructed as they would be
for a re-INVITE or BYE as specified in [2]. In particular, if the
provisional response contained a tag in the To field, this tag MUST
be mirrored in the To field of the COMET.
Once the COMET request is created, it is sent by the UAC. It is sent
as would any other non-INVITE request for a call. In particular,
when sent over UDP, the COMET request is retransmitted with an
exponentially increasing interval, starting at 500 milliseconds and
increasing to 4 seconds. Note that a UAC SHOULD NOT retransmit the
COMET request when it receives a retransmission of the provisional
response being acknowledged, although doing so does not create a
protocol error. As with any other non-INVITE request, the UAC
continues to retransmit the COMET request until it receives a final
response.
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A COMET request MAY be cancelled. However, whilst allowed for
purposes of generality, usage of CANCEL with COMET is NOT
RECOMMENDED.
5.1 Header Field Support for COMET Method
Tables 3 and 4 are extensions of tables 4 and 5 in the SIP
specification[2]. Refer to Section 6 of [2] for a description of
the content of the tables.
5.2 Responses to the COMET Request Method
If a server receives a COMET request it MUST send a final response.
A 200 OK response MUST be sent by a UAS for a COMET request if the
COMET request was successfully received for an existing call.
Beyond that, no additional operations are required.
A 481 Call Leg/Transaction Does Not Exist message MUST be sent by a
UAS if the COMET request does not match any existing call leg.
Header Where COMET
------ ----- ----
Accept R o
Accept-Encoding R o
Accept-Language R o
Allow 200 -
Allow 405 o
Authorization R o
Call-ID gc m
Contact R o
Contact 1xx -
Contact 2xx -
Contact 3xx -
Contact 485 -
Content-Encoding e o
Content-Length e o
Content-Type e *
CSeq gc m
Date g o
Encryption g o
Expires g o
From gc m
Hide R o
Max-Forwards R o
Organization g o
Table 3 Summary of header fields, A-0
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Header Where COMET
------ ----- ----
Priority R o
Proxy-Authenticate 407 o
Proxy-Authorization R o
Proxy-Require R o
Require R o
Retry-After R -
Retry-After 404,480,486 o
Retry-After 503 o
Retry-After 600,603 o
Response-Key R o
Record-Route R o
Record-Route 2xx o
Route R o
Server r o
Subject R o
Timestamp g o
To gc(1) m
Unsupported 420 o
User-Agent g o
Via gc(2) m
Warning r o
WWW-Authenticate 401 o
Table 4 Summary of header fields, P-Z
Other request failure (4xx), Server Failure (5xx) and Global Failure
(6xx) responses MAY be sent for the COMET Request.
5.3 Message Body Inclusion
The COMET request MUST contain a message body.
5.4 Behavior of SIP User Agents
Unless stated otherwise, the protocol rules for the COMET request
governing the usage of tags, Route and Record-Route, retransmission
and reliability, CSeq incrementing and message formatting follow
those in [2] as defined for the BYE request.
A COMET request MAY be cancelled. A UAS receiving a CANCEL for an
COMET request SHOULD respond to the COMET with a "487 Request
Cancelled" response if a final response has not been sent to the
COMET and then behave as if the request were never received.
5.5 Behavior of SIP Proxy and Redirect Servers
5.5.1 Proxy Server
Unless stated otherwise, the protocol rules for the COMET request at
a proxy are identical to those for a BYE request as specified in
[2].
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5.5.2 Forking Proxy Server
Unless stated otherwise, the protocol rules for the COMET request at
a proxy are identical to those for a BYE request as specified in
[2].
5.5.3 Redirection Server
Unless stated otherwise, the protocol rules for the COMET request at
a proxy are identical to those for a BYE request as specified in
[2].
6. SIP Extension: The 580-Precondition-Failure Response
An additional error response is defined by this draft, which is
returned by a UAS if it is unable to perform the mandatory
preconditions for the session.
6.1 Status Code and Reason Phrase
The following is to be added to Figure 8, Server error status codes
Server-Error = "580" ;Precondition-Failure
6.2 Status Code Definition
The following is to be added to a new section 7.5.7.
7.5.7 580 Precondition Failure
The server was unable to establish the qos or security association
mandated by the SDP precondition.
7. SIP Extension: Content-Disposition type
An additional type value is defined by this draft, which is returned
by a UAS in a provisional response indicating preconditions for the
session.
The following is to be changed in section 6.15.
Disposition-type = "render" | "session" | "icon" | "alert" |
"precondition" | disp-extension-token
The value "precondition" indicates the body part describes QoS
and/or security preconditions that SHOULD be established prior to
the start of the session.
8. SIP Usage Rules
8.1 Overview
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The session originator (UAC) prepares an SDP message body for the
INVITE describing the desired QoS and security preconditions for
each media flow, and the desired directions. The token value "send"
means the direction of media from originator (whichever entity
created the SDP) to recipient (whichever entity received the SDP in
a SIP message), and "recv" is from recipient to originator. In an
INVITE, the UAC is the originator, and the UAS is the recipient. The
roles are reversed in the response.
The recipient of the INVITE (UAS) returns a 18x provisional response
containing a Content-Disposition of "precondition," and SDP with the
qos/security attribute for each stream having a precondition. The
UAS would typically include a confirmation request. This SDP is a
subset of the preconditions indicated in the INVITE. Unlike normal
SIP processing, the UAS MUST NOT alert the called user at this
point. The UAS now attempts to reserve the qos resources and
establish the security associations.
The 18x provisional response is received by the UAC. If the 18x
contained SDP with mandatory qos/security parameters, the UAC does
not let the session proceed until the mandatory preconditions are
met. The UAC attempts to reserve the needed resources and establish
the security associations.
If either party requests a confirmation, a COMET message is sent to
that party. The COMET message contains the success/failure
indication for each precondition. For a precondition with a
direction value of "sendrecv," the COMET indicates whether the
sender is able to confirm both directions or only one direction.
Upon receipt of the COMET message, the UAC/UAS continues normal SIP
call handling, by (for a UAS) alerting the user and sending e.g. a
180-Ringing or 200-OK response. The UAC SIP transaction completes
normally.
Note that this extension requires usage of reliable provisional
responses [12]. This is because the 18x contains SDP with
information required for the session originator to initiate
reservations from it towards the participant.
8.2 Behavior of Originator (UAC)
The session originator (UAC) MAY include QoS and security
preconditions (including the desired direction) for each media flow
in the SDP sent with the INVITE. The token value "send" means the
direction of media from originator (whichever entity created the
SDP) to recipient (whichever entity received the SDP in a SIP
message), and "recv" is from recipient to originator. If
preconditions are included in the INVITE request, the UAC MUST
indicate support for reliable provisional responses [12].
If the UAC receives a 18x provisional response without a Content-
Disposition of "precondition," or without SDP, or with SDP but
without any qos/security preconditions in any stream, UAC treats it
as an indication that the UAS is unable or unwilling to perform the
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preconditions requested. As such, the UAC SHOULD proceed with normal
call setup procedures.
If the 18x provisional response contained a Content-Disposition of
"precondition" and contained SDP with mandatory qos/security
parameters, the UAC SHOULD NOT let the session proceed until the
mandatory preconditions are met.
If the 18x provisional response with preconditions requested an
acknowledgement (using the methods of [12]), the UAC MUST include an
updated SDP in the PRACK if the UAC modified the original SDP based
on the response from the UAS. Such a modification MAY be due to
negotiation of compatible CODECs, or MAY be due to negotiation of
mandatory preconditions. If the provisional response received from
the UAS is a 180-Ringing, and the direction value of a mandatory
precondition is "sendrecv," and the UAC uses a one-way QoS mechanism
(such as RSVP), the updated SDP in the PRACK SHOULD request a
confirmation from the UAS so that the bi-directional precondition
can be satisfied before performing the requested alerting function.
Upon receipt of the 18x provisional response with preconditions, the
UAC MUST initiate the qos reservations and establish the security
associations to the best of its capabilities.
If the UAC had requested confirmation in the initial SDP, it MAY
wait for the COMET message from the UAS containing the
success/failure status of each precondition. The UAC MAY set a
local timer to limit the time waiting for preconditions to complete.
The UAC SHOULD merge the success/failure status in the COMET message
with its local status. For example, if the COMET message indicated
success in the "send" direction, and the UAC was also able to meet
the precondition in the "send" direction, they combine to meet the
precondition in the "sendrecv" direction.
If any of the mandatory preconditions cannot be met, and a
confirmation was not requested by the UAS, the UAC MUST send a
CANCEL and terminate the session. If any of the optional
preconditions cannot be met, the UAC MAY consult with the
originating customer for guidance on whether to complete the
session.
When all the preconditions have either been met or have failed, and
the SDP received from the UAS included a confirmation request, the
UAC MUST send a COMET message to the UAS with SDP, where each
precondition is updated to indicate success/failure and the
appropriate direction tag based on local operations performed
combined with the received COMET message, if any.
The session now completes normally, as per [2].
8.3 Behavior of Destination (UAS)
On receipt of an INVITE request containing preconditions, the UAS
MUST generate a 18x provisional response containing a subset of the
preconditions supported by the UAS. In the response, the token
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value "send" means the direction of the media from the UAS to the
UAC, and "recv" is from the UAC to the UAS. This is reversed from
the SDP in the initial INVITE. The 18x provisional response MUST
include a Content-Disposition header with parameter "precondition."
If the "confirm" attribute is present in the SDP received from the
UAC, or if the direction tag of a mandatory QoS precondition is
"sendrecv" and the UAS only supports a one-way QoS reservation
scheme (e.g. RSVP), then the UAS SHOULD include a "confirm"
attribute.
If the INVITE request did not contain any preconditions, but did
indicate support for reliable provisional responses[12], the UAS MAY
include preconditions in a 18x provisional response to the INVITE.
The 18x provisional response MUST include a Content-Disposition
header with the parameter "precondition." The 18x provisional
response MUST request an acknowledgement using the mechanism of
[12]. If the PRACK includes an SDP without any preconditions, the
UAS MAY complete the session without the preconditions, or MAY
reject the INVITE request.
The UAS SHOULD request an acknowledgement to the 18x provisional
response, using the mechanism of [12]. The UAS SHOULD wait for the
PRACK message before initiating resource reservation to allow the
resource reservation to reflect 3-way SDP negotiation, or other
information available only through receipt of the PRACK.
If the INVITE request or PRACK message contained mandatory
preconditions, or requested a confirmation of preconditions, the UAS
MUST NOT alert the called user.
The UAS now attempts to reserve the qos resources and establish the
security associations. The UAS MAY set a local timer to limit the
time waiting for preconditions to complete.
If the UAS is unable to perform any mandatory precondition, and
neither the UAC nor UAS requested a confirmation, the UAS MUST send
a 580-Precondition-Failure response to the UAC. If the UAS is
unable to perform any optional precondition, it MAY consult with the
customer to obtain guidance regarding completion of the session.
When processing of all preconditions is complete, if a precondition
in the initial INVITE specified a confirmation request, the UAS MUST
send a COMET message to the UAC containing SDP, along with the
qos/security result of success/failure for each precondition. If
the direction tag of the precondition was "sendrecv" but the UAS was
only able to ensure "send" or "recv," the direction tag in the COMET
MUST only indicate what the UAS ensures. The Request-URI, call-leg
identification, and other headers of this COMET message are to be
constructed identically to a BYE.
If the UAS had requested confirmation of a precondition in the
response SDP, it SHOULD wait for the COMET message from the
originator containing the success/failure indication of each
precondition from the originator's point of view. The
success/failure indications in the COMET message from the UAC SHOULD
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be combined with the local status to determine the overall
success/failure of the precondition. For example, if the COMET
message indicated success in the "send" direction, and the UAS was
also able to meet the precondition in the "send" direction, they
combine to meet the precondition in the "sendrecv" direction. If
that combination indicates a failure for a mandatory precondition,
the UAS MUST send a 580-Precondition-Failure response to the UAC.
Once the preconditions are met, the UAS alerts the user, and the SIP
transaction proceeds normally.
The UAS MAY send additional 18x provisional responses with Content-
Disposition of "precondition," and the procedures described above
are repeated sequentially for each.
9. Examples
9.1 Basic (Single-Media) Call Flow
Figure 1 presents a high-level overview of a basic end-point to end-
point (UAC-UAS) call flow. This example is appropriate for a
single-media session with a mandatory quality-of-service "sendrecv"
precondition, where both the UAC and UAS can only perform a single-
direction ("send") resource reservation.
The session originator (UAC) prepares an SDP message body for the
INVITE describing the desired QoS and security preconditions for
each media flow, and the desired direction "sendrecv." This SDP is
included in the INVITE message sent through the proxies, and
includes an entry "a=qos:mandatory sendrecv."
The recipient of the INVITE (UAS), being willing to perform the
requested precondition, returns a 183-Session-Progress provisional
response containing SDP, along with the qos/security attribute for
each stream having a precondition. Since the "sendrecv" direction
tag required a cooperative effort of the UAC and UAS, the UAS
requests a confirmation when the preconditions are met, with the SDP
entry "a=qos:mandatory sendrecv confirm." The UAS now attempts to
reserve the qos resources and establish the security associations.
The 183-Session-Progress provisional response is sent using the
reliability mechanism of [12]. UAC sends the appropriate PRACK and
UAS responds with a 200-OK to the PRACK.
The 183-Session-Progress is received by the UAC, and the UAC
requests the resources needed in its "send" direction, and
establishes the security associations. Once the preconditions are
met, the UAC sends a COMET message as requested by the confirmation
token. This COMET message contains an entry "a=qos:success send"
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Originating (UAC) Terminating (UAS)
| SIP-Proxy(s) |
| INVITE | |
|---------------------->|---------------------->|
| | |
| 183 w/SDP | 183 w/SDP |
|<----------------------|<----------------------|
| |
| PRACK |
|---------------------------------------------->|
| 200 OK (of PRACK) |
|<----------------------------------------------|
| Reservation Reservation |
===========> <===========
| |
| |
| COMET |
|---------------------------------------------->|
| 200 OK (of COMET) |
|<----------------------------------------------|
|
|
| SIP-Proxy(s) User Alerted
| | |
| 180 Ringing | 180 Ringing |
|<----------------------|<----------------------|
| |
| PRACK |
|---------------------------------------------->|
| 200 OK (of PRACK) |
|<----------------------------------------------|
| |
| User Picks-Up
| SIP-Proxy(s) the phone
| | |
| 200 OK | 200 OK |
|<----------------------|<----------------------|
| | |
| |
| ACK |
|---------------------------------------------->|
Figure 1: Basic Call FLow
The UAS successfully performs its resource reservation, in its
"send" direction, and waits for the COMET message from the UAC.
On receipt of the COMET message, the UAS combines the "send"
confirmation in the SDP with its "send" success, and knows all
preconditions have been met. The UAS then continues with session
establishment. At this point it alerts the user, and sends a 180-
Ringing provisional response. This provisional response is also
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sent using the reliability mechanism of [12], resulting in a PRACK
message and 200-OK of the PRACK.
When the destination party answers, the normal SIP 200-OK final
response is sent through the proxies to the originator, and the
originator responds with an ACK message.
Either party can terminate the call. An endpoint that detects an
"on-hook" (request to terminate the call) releases the QoS resources
held for the connection, and sends a SIP BYE message to the remote
endpoint, which is acknowledged with a 200-OK.
9.2 Advanced (Multiple-Media) Call Flow
Figure 2 presents a high-level overview of an advanced end-point to
end-point (UAC-UAS) call flow. This example is appropriate for a
multiple-media session with some combination of mandatory and
optional quality-of-service precondition. For example, the
originator may suggest five media streams, and be willing to
establish the session if any three of them are satisfied.
The use of reliable provisional responses is assumed, but not shown
in this figure.
The session originator (UAC) prepares an SDP message body for the
INVITE describing the desired QoS and security preconditions for
each media flow, and the desired directions. UAC also requests
confirmation of the preconditions. The UAS receiving the INVITE
message responds with 183-Session-Progress, as in the previous
example.
When the UAS has completed the resource reservations and security
session establishment, it sends a confirmation to the UAC in the
form of a COMET message, with each precondition marked in the SDP as
either success or failure. Note that if UAS was not satisfied with
the combination of successful preconditions, it could instead have
responded with 580-Precondition-Failure, and ended the INVITE
transaction.
If the UAC has satisfied its preconditions, and is satisfied with
the preconditions achieved by the UAS, it responds with the COMET
message. The COMET message contains the SDP with the
success/failure results of each precondition attempted by UAC. If
UAC is not satisfied with the combination of successful
preconditions, it could instead have sent a CANCEL message.
On receipt of the COMET message, UAS examines the combination of
satisfied preconditions reported by UAC, and makes a final decision
whether to proceed with the session. If sufficient preconditions
are not satisfied, the UAS responds with 580-Precondition-Failure.
Otherwise, the session proceeds as in the previous example.
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Originating (UAC) Terminating (UAS)
| SIP-Proxy(s) |
| INVITE | |
|---------------------->|---------------------->|
| | |
| 183 w/SDP | 183 w/SDP |
|<----------------------|<----------------------|
| |
| Reservation Reservation |
===========> <===========
| |
| COMET |
|<----------------------------------------------|
| 200 OK (of COMET) |
|---------------------------------------------->|
| |
| COMET |
|---------------------------------------------->|
| 200 OK (of COMET) |
|<----------------------------------------------|
|
|
| SIP-Proxy(s) User Alerted
| | |
| 180 Ringing | 180 Ringing |
|<----------------------|<----------------------|
| |
| |
| User Picks-Up
| SIP-Proxy(s) the phone
| | |
| 200 OK | 200 OK |
|<----------------------|<----------------------|
| | |
| |
| ACK |
|---------------------------------------------->|
Figure 2: Call Flow with negotiation of optional preconditions
10. Special considerations with Forking Proxies
If a proxy along the signaling path between the UAC and UAS forks
the INVITE request, it results in two or more UASs simultaneously
sending provisional responses with preconditions. The procedures
above result in the UAC handling each independently, reserving
resources and responding with COMET messages as required.
This results in multiple resource reservations from the UAC, only
one of which will be utilized for the final session. While
functionally correct, this has the unfortunate side-effect of
increasing the call blocking probability.
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Customized resource allocation protocols may be used by the UAC to
reduce this duplicate allocation and prevent excess blocking of
calls. For one such example, see [8].
11. Advantages of the Proposed Approach
The use of two-phase call signaling makes it possible for SIP to
meet user expectations that come from the telephony services.
The reservation of resources before the user is alerted makes sure
that the network resources are assured before the destination end-
point is informed about the call.
The number of messages and the total delay introduced by these
messages is kept to a minimum without sacrificing compatibility with
SIP servers that do not implement preconditions.
12. Security Considerations
If the contents of the SDP contained in the 183-Session-Progress are
private then end-to-end encryption of the message body can be used
to prevent unauthorized access to the content.
The security considerations given in the SIP specification apply to
the COMET method. No additional security considerations specific to
the COMET method are necessary.
13. Notice Regarding Intellectual Property Rights
AT&T may seek patent or other intellectual property protection for
some or all of the technologies disclosed in the document. If any
standards arising from this disclosure are or become protected by
one or more patents assigned to AT&T, AT&T intends to disclose those
patents and license them on reasonable and non-discriminatory terms.
Future revisions of this draft may contain additional information
regarding specific intellectual property protection sought or
received.
3COM may seek patent or other intellectual property protection for
some or all of the technologies disclosed in the document. If any
standards arising from this disclosure are or become protected by
one or more patents assigned to 3COM, 3COM intends to disclose those
patents and license them on reasonable and non-discriminatory terms.
Future revisions of this draft may contain additional information
regarding specific intellectual property protection sought or
received.
14. References
1. Bradner, S., "The Internet Standards Process -- Revision 3", BCP
9, RFC 2026, October 1996.
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2. M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
Session Initiation Protocol," RFC 2543, March 1999.
3. M. Handley and V. Jacobson, "SDP: Session Description Protocol,"
RFC 2327, April 1998.
4. Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997
5. DCS Group, "Architectural Considerations for Providing Carrier
Class Telephony Services Utilizing SIP-based Distributed Call
Control Mechanisms", <draft-dcsgroup-sip-arch-03.txt>, November
2000, Work in Progress.
6. R. Braden, Ed., L. Zhang, S. Berson, S. Herzog, and S. Jamin,
"Resource ReSerVation protocol (RSVP) -- version 1 functional
specification," RFC 2205, September, 1997.
7. P. P. Pan and H. Schulzrinne, "YESSIR: A simple reservation
mechanism for the Internet," in Proc. International Workshop on
Network and Operating System Support for Digital Audio and Video
(NOSSDAV), (Cambridge, England), July 1998. Also IBM Research
Technical Report TC20967. Available at
http://www.cs.columbia.edu/~hgs/papers/Pan98_YESSIR.ps.gz.
8. PacketCable, Dynamic Quality of Service Specification, pkt-sp-
dqos-i01-991201, December 1, 1999. Available at
http://www.packetcable.com/specs/pkt-sp-dqos-i01-991202.pdf.
9. S. Kent and R. Atkinson, "Security architecture for the internet
protocol," RFC 2401, November 1998.
10. H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
a Transport Protocol for Real-Time Applications," RFC 1889,
January 1996.
11. M. Handley, C. Perkins, and E. Whelan, "Session Announcement
Protocol," Internet Draft, <draft-ietf-mmusic-sap-v2-06.txt>,
March 2000, Work in Progress.
12. "Reliability of Provisional Responses in SIP," Internet Draft
<draft-ietf-sip-100rel-00.txt>, January 2000, Work in Progress.
15. Acknowledgments
The Distributed Call Signaling work in the PacketCable project is
the work of a large number of people, representing many different
companies. The authors would like to recognize and thank the
following for their assistance: John Wheeler, Motorola; David
Boardman, Daniel Paul, Arris Interactive; Bill Blum, Jay Strater,
Jeff Ollis, Clive Holborow, General Instruments; Doug Newlin, Guido
SIP Working Group Expiration 5/31/01 22
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Schuster, Ikhlaq Sidhu, 3Com; Jiri Matousek, Bay Networks; Farzi
Khazai, Nortel; John Chapman, Bill Guckel, Cisco; Chuck Kalmanek,
Doug Nortz, John Lawser, James Cheng, Tung-Hai Hsiao, Partho Mishra,
AT&T; Telcordia Technologies; and Lucent Cable Communications.
16. Author's Addresses
Bill Marshall
AT&T
Florham Park, NJ 07932
Email: wtm@research.att.com
K. K. Ramakrishnan
TeraOptic Networks
Sunnyvale, CA
Email: kk@teraoptic.com
Ed Miller
CableLabs
Louisville, CO 80027
Email: E.Miller@Cablelabs.com
Glenn Russell
CableLabs
Louisville, CO 80027
Email: G.Russell@Cablelabs.com
Burcak Beser
Pacific Broadband Communications
San Jose, CA
Email: Burcak@pacband.com
Mike Mannette
3Com
Rolling Meadows, IL 60008
Email: Michael_Mannette@3com.com
Kurt Steinbrenner
3Com
Rolling Meadows, IL 60008
Email: Kurt_Steinbrenner@3com.com
Dave Oran
Cisco
Acton, MA 01720
Email: oran@cisco.com
Flemming Andreasen
Cisco
Edison, NJ
Email: fandreas@cisco.com
Michael Ramalho
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Cisco
Wall Township, NJ
Email: mramalho@cisco.com
John Pickens
Com21
San Jose, CA
Email: jpickens@com21.com
Poornima Lalwaney
Nokia
San Diego, CA 92121
Email: poornima.lalwaney@nokia.com
Jon Fellows
Motorola
San Diego, CA 92121
Email: jfellows@gi.com
Doc Evans
D. R. Evans Consulting
Boulder, CO 80303
Email: n7dr@arrl.net
Keith Kelly
NetSpeak
Boca Raton, FL 33587
Email: keith@netspeak.com
Adam Roach
Ericsson
Richardson, TX 75081
Email: adam.roach@ericsson.com
Jonathan Rosenberg
dynamicsoft
West Orange, NJ 07052
Email: jdrosen@dynamicsoft.com
Dean Willis
dynamicsoft
West Orange, NJ 07052
Email: dwillis@dynamicsoft.com
Steve Donovan
dynamicsoft
West Orange, NJ 07052
Email: sdonovan@dynamicsoft.com
Henning Schulzrinne
Columbia University
New York, NY
Email: schulzrinne@cs.columbia.edu
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Full Copyright Statement
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others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published
and 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
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the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
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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 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE."
Expiration Date This memo is filed as <draft-ietf-sip-manyfolks-
resource-00.txt>, and expires May 31, 2001.
SIP Working Group Expiration 5/31/01 25
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