| Rfc | 4542 |
|---|
| Title | Implementing an Emergency Telecommunications Service (ETS) for
Real-Time Services in the Internet Protocol Suite |
|---|
| Author | F. Baker, J.
Polk |
|---|
| Date | May 2006 |
|---|
| Format: | TXT, HTML |
|---|
| Updated by | RFC5865 |
|---|
| Status: | INFORMATIONAL |
|---|
|
Network Working Group F. Baker
Request for Comments: 4542 J. Polk
Category: Informational Cisco Systems
May 2006
Implementing an Emergency Telecommunications Service (ETS) for
Real-Time Services in the Internet Protocol Suite
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
RFCs 3689 and 3690 detail requirements for an Emergency
Telecommunications Service (ETS), of which an Internet Emergency
Preparedness Service (IEPS) would be a part. Some of these types of
services require call preemption; others require call queuing or
other mechanisms. IEPS requires a Call Admission Control (CAC)
procedure and a Per Hop Behavior (PHB) for the data that meet the
needs of this architecture. Such a CAC procedure and PHB is
appropriate to any service that might use H.323 or SIP to set up
real-time sessions. The key requirement is to guarantee an elevated
probability of call completion to an authorized user in time of
crisis.
This document primarily discusses supporting ETS in the context of
the US Government and NATO, because it focuses on the Multi-Level
Precedence and Preemption (MLPP) and Government Emergency
Telecommunication Service (GETS) standards. The architectures
described here are applicable beyond these organizations.
Table of Contents
1. Overview of the Internet Emergency Preference Service
Problem and Proposed Solutions ..................................3
1.1. Emergency Telecommunications Services ......................3
1.1.1. Multi-Level Preemption and Precedence ...............4
1.1.2. Government Emergency Telecommunications Service .....6
1.2. Definition of Call Admission ...............................6
1.3. Assumptions about the Network ..............................7
1.4. Assumptions about Application Behavior .....................7
1.5. Desired Characteristics in an Internet Environment .........9
1.6. The Use of Bandwidth as a Solution for QoS ................10
2. Solution Proposal ..............................................11
2.1. Call Admission/Preemption Procedure .......................12
2.2. Voice Handling Characteristics ............................15
2.3. Bandwidth Admission Procedure .............................17
2.3.1. RSVP Admission Using Policy for Both
Unicast and Multicast Sessions .....................17
2.3.2. RSVP Scaling Issues ................................19
2.3.3. RSVP Operation in Backbones and Virtual
Private Networks (VPNs) ............................19
2.3.4. Interaction with the Differentiated
Services Architecture ..............................21
2.3.5. Admission Policy ...................................21
2.4. Authentication and Authorization of Calls Placed ..........23
2.5. Defined User Interface ....................................23
3. Security Considerations ........................................24
4. Acknowledgements ...............................................24
5. References .....................................................25
5.1. Normative References ......................................25
5.2. Informative References ....................................27
Appendix A. 2-Call Preemption Example using RSVP .................29
1. Overview of the Internet Emergency Preference Service Problem and
Proposed Solutions
[RFC3689] and [RFC3690] detail requirements for an Emergency
Telecommunications Service (ETS), of which an Internet Emergency
Preference Service (IEPS) would be a part. Some of these types of
services require call preemption; others require call queuing or
other mechanisms. The key requirement is to guarantee an elevated
probability of call completion to an authorized user in time of
crisis.
IEPS requires a Call Admission Control procedure and a Per Hop
Behavior for the data that meet the needs of this architecture. Such
a CAC procedure and PHB is appropriate to any service that might use
H.323 or SIP to set up real-time sessions. These obviously include
but are not limited to Voice and Video applications, although at this
writing the community is mostly thinking about Voice on IP, and many
of the examples in the document are taken from that environment.
In a network where a call permitted initially is not denied or
rejected at a later time, capacity admission procedures performed
only at the time of call setup may be sufficient. However, in a
network where session status can be reviewed by the network and
preempted or denied due to changes in routing (when the new routes
lack capacity to carry calls switched to them) or changes in offered
load (where higher precedence calls supersede existing calls),
maintaining a continuing model of the status of the various calls is
required.
1.1. Emergency Telecommunications Services
Before doing so, however, let us discuss the problem that ETS (and
therefore IEPS) is intended to solve and the architecture of the
system. The Emergency Telecommunications Service [ITU.ETS.E106] is a
successor to and generalization of two services used in the United
States: Multi-Level Precedence and Preemption (MLPP), and the
Government Emergency Telecommunication Service (GETS). Services
based on these models are also used in a variety of countries
throughout the world, both Public Switched Telephone Network (PSTN)
and Global System for Mobile Communications (GSM)-based. Both of
these services are designed to enable an authorized user to obtain
service from the telephone network in times of crisis. They differ
primarily in the mechanisms used and number of levels of precedence
acknowledged.
1.1.1. Multi-Level Preemption and Precedence
The Assured Service is designed as an IP implementation of an
existing ITU-T/NATO/DoD telephone system architecture known as
Multi-Level Precedence and Preemption [ITU.MLPP.1990]
[ANSI.MLPP.Spec] [ANSI.MLPP.Supp], or MLPP. MLPP is an architecture
for a prioritized call handling service such that in times of
emergency in the relevant NATO and DoD commands, the relative
importance of various kinds of communications is strictly defined,
allowing higher-precedence communication at the expense of lower-
precedence communications. This document describes NATO and US
Department of Defense uses of MLPP, but the architecture and standard
are applicable outside of these organizations.
These precedences, in descending order, are:
Flash Override Override: used by the Commander in Chief, Secretary
of Defense, and Joint Chiefs of Staff, commanders of combatant
commands when declaring the existence of a state of war.
Commanders of combatant commands when declaring Defense Condition
One or Defense Emergency or Air Defense Emergency and other
national authorities that the President may authorize in
conjunction with Worldwide Secure Voice Conferencing System
conferences. Flash Override Override cannot be preempted. This
precedence level is not enabled on all DoD networks.
Flash Override: used by the Commander in Chief, Secretary of
Defense, and Joint Chiefs of Staff, commanders of combatant
commands when declaring the existence of a state of war.
Commanders of combatant commands when declaring Defense Condition
One or Defense Emergency and other national authorities the
President may authorize. Flash Override cannot be preempted in
the DSN.
Flash: reserved generally for telephone calls pertaining to command
and control of military forces essential to defense and
retaliation, critical intelligence essential to national survival,
conduct of diplomatic negotiations critical to the arresting or
limiting of hostilities, dissemination of critical civil alert
information essential to national survival, continuity of federal
government functions essential to national survival, fulfillment
of critical internal security functions essential to national
survival, or catastrophic events of national or international
significance.
Immediate: reserved generally for telephone calls pertaining to
situations that gravely affect the security of national and allied
forces, reconstitution of forces in a post-attack period,
intelligence essential to national security, conduct of diplomatic
negotiations to reduce or limit the threat of war, implementation
of federal government actions essential to national survival,
situations that gravely affect the internal security of the
nation, Civil Defense actions, disasters or events of extensive
seriousness having an immediate and detrimental effect on the
welfare of the population, or vital information having an
immediate effect on aircraft, spacecraft, or missile operations.
Priority: reserved generally for telephone calls requiring
expeditious action by called parties and/or furnishing essential
information for the conduct of government operations.
Routine: designation applied to those official government
communications that require rapid transmission by telephonic means
but do not require preferential handling.
MLPP is intended to deliver a higher probability of call completion
to the more important calls. The rule, in MLPP, is that more
important calls override less important calls when congestion occurs
within a network. Station-based preemption is used when a more
important call needs to be placed to either party in an existing
call. Trunk-based preemption is used when trunk bandwidth needs to
be reallocated to facilitate a higher-precedence call over a given
path in the network. In both station- and trunk-based preemption
scenarios, preempted parties are positively notified, via preemption
tone, that their call can no longer be supported. The same
preemption tone is used, regardless of whether calls are terminated
for the purposes of station- of trunk-based preemption. The
remainder of this discussion focuses on trunk-based preemption
issues.
MLPP is built as a proactive system in which callers must assign one
of the precedence levels listed above at call initiation; this
precedence level cannot be changed throughout that call. If an
elevated status is not assigned by a user at call initiation time,
the call is assumed to be "routine". If there is end-to-end capacity
to place a call, any call may be placed at any time. However, when
any trunk group (in the circuit world) or interface (in an IP world)
reaches a utilization threshold, a choice must be made as to which
calls to accept or allow to continue. The system will seize the
trunk(s) or bandwidth necessary to place the more important calls in
preference to less important calls by preempting an existing call (or
calls) of lower precedence to permit a higher-precedence call to be
placed.
More than one call might properly be preempted if more trunks or
bandwidth is necessary for this higher precedence call. A video call
(perhaps of 384 KBPS, or 6 trunks) competing with several lower-
precedence voice calls is a good example of this situation.
1.1.2. Government Emergency Telecommunications Service
A US service similar to MLPP and using MLPP signaling technology, but
built for use in civilian networks, is the Government Emergency
Telecommunications Service (GETS). This differs from MLPP in two
ways: it does not use preemption, but rather reserves bandwidth or
queues calls to obtain a high probability of call completion, and it
has only two levels of service: "Routine" and "Priority".
GETS is described here as another example. Similar architectures are
applied by other governments and organizations.
1.2. Definition of Call Admission
Traditionally, in the PSTN, Call Admission Control (CAC) has had the
responsibility of implementing bandwidth available thresholds (e.g.,
to limit resources consumed by some traffic) and determining whether
a caller has permission (e.g., is an identified subscriber, with
identify attested to by appropriate credentials) to use an available
circuit. IEPS, or any emergency telephone service, has additional
options that it may employ to improve the probability of call
completion:
o The call may be authorized to use other networks that it would not
normally use;
o The network may preempt other calls to free bandwidth;
o The network may hold the call and place it when other calls
complete; or
o The network may use different bandwidth availability thresholds
than are used for other calls.
At the completion of CAC, however, the caller either has a circuit
that he or she is authorized to use or has no circuit. Since the act
of preemption or consideration of alternative bandwidth sources is
part and parcel of the problem of providing bandwidth, the
authorization step in bandwidth provision also affects the choice of
networks that may be authorized to be considered. The three cannot
be separated. The CAC procedure finds available bandwidth that the
caller is authorized to use and preemption may in some networks be
part of making that happen.
1.3. Assumptions about the Network
IP networks generally fall into two categories: those with
constrained bandwidth, and those that are massively over-provisioned.
In a network where over any interval that can be measured (including
sub-second intervals) capacity exceeds offered load by at least 2:1,
the jitter and loss incurred in transit are nominal. This is
generally a characteristic of properly engineered Ethernet LANs and
of optical networks (networks that measure their link speeds in
multiples of 51 MBPS); in the latter, circuit-switched networking
solutions such as Asynchronous Transfer Mode (ATM), MPLS, and GMPLS
can be used to explicitly place routes, which improves the odds a
bit.
Between those networks, in places commonly called "inter-campus
links", "access links", or "access networks", for various reasons
including technology (e.g., satellite links) and cost, it is common
to find links whose offered load can approximate or exceed the
available capacity. Such events may be momentary or may occur for
extended periods of time.
In addition, primarily in tactical deployments, it is common to find
bandwidth constraints in the local infrastructure of networks. For
example, the US Navy's network afloat connects approximately 300
ships, via satellite, to five network operation centers (NOCs), and
those NOCs are in turn interconnected via the Defense Information
Systems Agency (DISA) backbone. A typical ship may have between two
and six radio systems aboard, often at speeds of 64 KBPS or less. In
US Army networks, current radio technology likewise limits tactical
communications to links below 100 KBPS.
Over this infrastructure, military communications expect to deploy
voice communication systems (30-80 KBPS per session) and video
conferencing using MPEG 2 (3-7 MBPS) and MPEG 4 (80 KBPS to 800
KBPS), in addition to traditional mail, file transfer, and
transaction traffic.
1.4. Assumptions about Application Behavior
Parekh and Gallagher published a series of papers [Parekh1] [Parekh2]
analyzing what is necessary to ensure a specified service level for a
stream of traffic. In a nutshell, they showed that to predict the
behavior of a stream of traffic in a network, one must know two
things:
o the rate and arrival distribution with which traffic in a class is
introduced to the network, and
o what network elements will do, in terms of the departure
distribution, injected delay jitter, and loss characteristics,
with the traffic they see.
For example, TCP tunes its effective window (the amount of data it
sends per round trip interval) so that the ratio of the window and
the round trip interval approximate the available capacity in the
network. As long as the round trip delay remains roughly stable and
loss is nominal (which are primarily behaviors of the network), TCP
is able to maintain a predictable level of throughput. In an
environment where loss is random or in which delays wildly vary, TCP
behaves in a far less predictable manner.
Voice and video systems, in the main, are designed to deliver a fixed
level of quality as perceived by the user. (Exceptions are systems
that select rate options over a broad range to adapt to ambient loss
characteristics. These deliver broadly fluctuating perceived quality
and have not found significant commercial applicability.) Rather,
they send traffic at a rate specified by the codec depending on what
it perceives is required. In an MPEG-4 system, for example, if the
camera is pointed at a wall, the codec determines that an 80 KBPS
data stream will describe that wall and issues that amount of
traffic. If a person walks in front of the wall or the camera is
pointed an a moving object, the codec may easily send 800 KBPS in its
effort to accurately describe what it sees. In commercial broadcast
sports, which may line up periods in which advertisements are
displayed, the effect is that traffic rates suddenly jump across all
channels at certain times because the eye-catching ads require much
more bandwidth than the camera pointing at the green football field.
As described in [RFC1633], when dealing with a real-time application,
there are basically two things one must do to ensure Parekh's first
requirement. To ensure that one knows how much offered load the
application is presenting, one must police (measure load offered and
discard excess) traffic entering the network. If that policing
behavior has a debilitating effect on the application, as non-
negligible loss has on voice or video, one must admit sessions
judiciously according to some policy. A key characteristic of that
policy must be that the offered load does not exceed the capacity
dedicated to the application.
In the network, the other thing one must do is ensure that the
application's needs are met in terms of loss, variation in delay, and
end-to-end delay. One way to do this is to supply sufficient
bandwidth so that loss and jitter are nominal. Where that cannot be
accomplished, one must use queuing technology to deterministically
apply bandwidth to accomplish the goal.
1.5. Desired Characteristics in an Internet Environment
The key elements of the Internet Emergency Preference Service include
the following:
Precedence Level Marking each call: Call initiators choose the
appropriate precedence level for each call based on the user-
perceived importance of the call. This level is not to be changed
for the duration of the call. The call before and the call after
are independent with regard to this level choice.
Call Admission/Preemption Policy: There is likewise a clear policy
regarding calls that may be in progress at the called instrument.
During call admission (SIP/H.323), if they are of lower
precedence, they must make way according to a prescribed
procedure. All callers on the preempted call must be informed
that the call has been preempted, and the call must make way for
the higher-precedence call.
Bandwidth Admission Policy: There is a clear bandwidth admission
policy: sessions may be placed that assert any of several levels
of precedence, and in the event that there is demand and
authorization is granted, other sessions will be preempted to make
way for a call of higher precedence.
Authentication and Authorization of calls placed: Unauthorized
attempts to place a call at an elevated status are not permitted.
In the telephone system, this is managed by controlling the policy
applied to an instrument by its switch plus a code produced by the
caller identifying himself or herself to the switch. In the
Internet, such characteristics must be explicitly signaled.
Voice handling characteristics: A call made, in the telephone
system, gets a circuit and provides the means for the callers to
conduct their business without significant impact as long as their
call is not preempted. In a VoIP system, one would hope for
essentially the same service.
Defined User Interface: If a call is preempted, the caller and the
callee are notified via a defined signal, so that they know that
their call has been preempted and that at this instant there is no
alternative circuit available to them at that precedence level.
A VoIP implementation of the Internet Emergency Preference Service
must, by definition, provide those characteristics.
1.6. The Use of Bandwidth as a Solution for QoS
There is a discussion in Internet circles concerning the relationship
of bandwidth to QoS procedures, which needs to be put to bed before
this procedure can be adequately analyzed. The issue is that it is
possible and common in certain parts of the Internet to solve the
problem with bandwidth. In LAN environments, for example, if there
is significant loss between any two switches or between a switch and
a server, the simplest and cheapest solution is to buy the next
faster interface: substitute 100 MBPS for 10 MBPS Ethernet, 1 gigabit
for 100 MBPS, or, for that matter, upgrade to a 10-gigabit Ethernet.
Similarly, in optical networking environments, the simplest and
cheapest solution is often to increase the data rate of the optical
path either by selecting a faster optical carrier or deploying an
additional lambda. In places where the bandwidth can be over-
provisioned to a point where loss or queuing delay are negligible,
10:1 over-provisioning is often the cheapest and surest solution and,
by the way, offers a growth path for future requirements. However,
there are many places in communication networks where the provision
of effectively infinite bandwidth is not feasible, including many
access networks, satellite communications, fixed wireless, airborne
and marine communications, island connections, and connections to
regions in which fiber optic connections are not cost-effective. It
is in these places where the question of resource management is
relevant. Specifically, we do not recommend the deployment of
significant QoS procedures on links in excess of 100 MBPS apart from
the provision of aggregated services that provide specific protection
to the stability of the network or the continuity of real-time
traffic as a class, as the mathematics of such circuits do not
support this as a requirement.
In short, the fact that we are discussing this class of policy
control says that such constrictions in the network exist and must be
dealt with. However much we might like to, in those places we are
not solving the problem with bandwidth.
2. Solution Proposal
A typical voice or video network, including a backbone domain, is
shown in Figure 1.
............... ......................
. . . .
. H H H H . . H H H H .
. /----------/ . . /----------/ .
. R SIP . . R R .
. \ . . / \ .
. R H H H . ....... / \ .
. /----------/ .. ../ R SIP .
. R .. /. /----------/ .
..... ..\. R-----R . H H H H .
...... .\ / \ . .
. \ / \ . .
. R-----------R ....................
. \ / .
. \ / .
. R-----R .
. .
............
SIP = SIP Proxy
H = SIP-enabled Host (Telephone, call gateway or PC)
R = Router
/---/ = Ethernet or Ethernet Switch
Figure 1: Typical VoIP or Video/IP Network
Reviewing the figure above, it becomes obvious that Voice/IP and
Video/IP call flows are very different than call flows in the PSTN.
In the PSTN, call control traverses a switch, which in turn controls
data handling services like ATM or Time Division Multiplexing (TDM)
switches or multiplexers. While they may not be physically co-
located, the control plane software and the data plane services are
closely connected; the switch routes a call using bandwidth that it
knows is available. In a voice/video-on-IP network, call control is
completely divorced from the data plane: It is possible for a
telephone instrument in the United States to have a Swedish telephone
number if that is where its SIP proxy happens to be, but on any given
call for it to use only data paths in the Asia/Pacific region, data
paths provided by a different company, and, often, data paths provided
by multiple companies/providers.
Call management therefore addresses a variety of questions, all of
which must be answered:
o May I make this call from an administrative policy perspective?
Am I authorized to make this call?
o What IP address correlates with this telephone number or SIP URI?
o Is the other instrument "on hook"? If it is busy, under what
circumstances may I interrupt?
o Is there bandwidth available to support the call?
o Does the call actually work, or do other impairments (loss, delay)
make the call unusable?
2.1. Call Admission/Preemption Procedure
Administrative Call Admission is the objective of SIP and H.323. It
asks fundamental questions like "What IP address is the callee at?"
and "Did you pay your bill?".
For a specialized policy like call preemption, two capabilities are
necessary from an administrative perspective: [RFC4412] provides a
way to communicate policy-related information regarding the
precedence of the call; and [RFC4411] provides a reason code when a
call fails or is refused, indicating the cause of the event. If it
is a failure, it may make sense to redial the call. If it is a
policy-driven preemption, even if the call is redialed it may not be
possible to place the call. Requirements for this service are
further discussed in [RFC3689].
The SIP Communications Resource Priority Header (or RP Header) serves
the call setup process with the precedence level chosen by the
initiator of the call. The syntax is in the form:
Resource Priority: namespace.priority level
The "namespace" part of the syntax ensures the domain of significance
to the originator of the call, and this travels end-to-end to the
destination (called) device (telephone). If the receiving phone does
not support the namespace, it can easily ignore the setup request.
This ability to denote the domain of origin allows Service Level
Agreements (SLAs) to be in place to limit the ability of an unknown
requester to gain preferential treatment into an IEPS domain.
For the DSN infrastructure, the header would look like this for a
routine precedence level call:
Resource Priority: dsn.routine
The precedence level chosen in this header would be compared to the
requester's authorization profile to use that precedence level. This
would typically occur in the SIP first-hop Proxy, which can challenge
many aspects of the call setup request including the requester's
choice of precedence levels (verifying that they are not using a
level they are not authorized to use).
The DSN has 5 precedence levels of IEPS, in descending order:
dsn.flash-override
dsn.flash
dsn.immediate
dsn.priority
dsn.routine
The US Defense Red Switched Network (DRSN), as another example that
was IANA-registered in [RFC4412], has 6 levels of precedence. The
DRSN simply adds one precedence level higher than flash-override to
be used by the President and a select few others:
drsn.flash-override-override
Note that the namespace changed for this level. The lower 5 levels
within the DRSN would also have this as their namespace for all
DRSN-originated call setup requests.
The Resource-Priority Header (RPH) informs both the use of
Differentiated Services Code Points (DSCPs) by the callee (who needs
to use the same DSCP as the caller to obtain the same data path
service) and to facilitate policy-based preemption of calls in
progress, when appropriate.
Once a call is established in an IEPS domain, the Reason Header for
Preemption, described in [RFC4411], ensures that all SIP nodes are
synchronized to a preemption event occurring either at the endpoint
or in a router that experiences congestion. In SIP, the normal
indication for the end of a session is for one end system to send a
BYE Method request as specified in [RFC3261]. This, too, is the
proper means for signaling a termination of a call due to a
preemption event, as it essentially performs a normal termination
with additional information informing the peer of the reason for the
abrupt end: it indicates that a preemption occurred. This will be
used to inform all relevant SIP entities, and whether this was an
endpoint-generated preemption event, or that the preemption event
occurred within a router along the communications path (described in
Section 2.3.1).
Figure 2 is a simple example of a SIP call setup that includes the
layer 7 precedence of a call between Alice and Bob. After Alice
successfully sets up a call to Bob at the "Routine" precedence level,
Carol calls Bob at a higher precedence level (Immediate). At the SIP
layer (this has nothing to do with RSVP yet; that example, involving
SIP and RSVP signaling, is in the appendix), once Bob's user agent
(phone) receives the INVITE message from Carol, his UA needs to make
a choice between retaining the call to Alice and sending Carol a
"busy" indication, or preempting the call to Alice in favor of
accepting the call from Carol. That choice in IEPS networks is a
comparison of Resource Priority headers. Alice, who controlled the
precedence level of the call to Bob, sent the precedence level of her
call to him at "Routine" (the lowest level within the network).
Carol, who controls the priority of the call signal to Bob, sent her
priority level to "Immediate" (higher than "Routine"). Bob's UA
needs to (under IEPS policy) preempt the call from Alice (and provide
her with a preemption indication in the call termination message).
Bob needs to successfully answer the call setup from Carol.
UA Alice UA Bob UA Carol
| INVITE (RP: Routine) | |
|--------------------------->| |
| 200 OK | |
|<---------------------------| |
| ACK | |
|--------------------------->| |
| RTP | |
|<==========================>| |
| | |
| | INVITE (RP: Immediate) |
| |<----------------------------|
| ************************************************ |
| *Resource Priority value comparison by Bob's UA* |
| ************************************************ |
| | |
| BYE (Reason: UA preemption) |
|<---------------------------| |
| | 200 OK |
| |---------------------------->|
| 200 OK (BYE) | |
|--------------------------->| |
| | ACK |
| |<----------------------------|
| | RTP |
| |<===========================>|
| | |
Figure 2: Priority Call Establishment and Termination at SIP Layer
Nothing in this example involved mechanisms other than SIP. It is
also assumed each user agent recognized the Resource-Priority header
namespace value in each message. Therefore, it is assumed that the
domain allowed Alice, Bob, and Carol to communicate. Authentication
and Authorization are discussed later in this document.
2.2. Voice Handling Characteristics
The Quality of Service architecture used in the data path is that of
[RFC2475]. Differentiated Services uses a flag in the IP header
called the DSCP [RFC2474] to identify a data stream, and then applies
a procedure called a Per Hop Behavior, or PHB, to it. This is
largely as described in [RFC2998].
In the data path, the Expedited Forwarding PHB [RFC3246] [RFC3247]
describes the fundamental needs of voice and video traffic. This PHB
entails ensuring that sufficient bandwidth is dedicated to real-time
traffic to ensure that variation in delay and loss rate are minimal,
as codecs are hampered by excessive loss [G711.1] [G711.3]. In parts
of the network where bandwidth is heavily over-provisioned, there may
be no remaining concern. In places in the network where bandwidth is
more constrained, this may require the use of a priority queue. If a
priority queue is used, the potential for abuse exists, meaning that
it is also necessary to police traffic placed into the queue to
detect and manage abuse. A fundamental question is "where does this
policing need to take place?". The obvious places would be the
first-hop routers and any place where converging data streams might
congest a link.
Some proposals mark traffic with various code points appropriate to
the service precedence of the call. In normal service, if the
traffic is all in the same queue and EF service requirements are met
(applied capacity exceeds offered load, variation in delay is
minimal, and loss is negligible), details of traffic marking should
be irrelevant, as long as packets get into the right service class.
Then, the major issues are appropriate policing of traffic,
especially around route changes, and ensuring that the path has
sufficient capacity.
The real-time voice/video application should be generating traffic at
a rate appropriate to its content and codec, which is either a
constant bit rate stream or a stream whose rate is variable within a
specified range. The first-hop router should be policing traffic
originated by the application, as is performed in traditional virtual
circuit networks like Frame Relay and ATM. Between these two checks
(at what some networks call the Data Terminal Equipment (DTE) and
Data Communications Equipment (DCE)), the application traffic should
be guaranteed to be within acceptable limits. As such, given
bandwidth-aware call admission control, there should be minimal
actual loss. The cases where loss would occur include cases where
routing has recently changed and CAC has not caught up, or cases
where statistical thresholds are in use in CAC and the data streams
happen to coincide at their peak rates.
If it is demonstrated that routing transients and variable rate beat
frequencies present a sufficient problem, it is possible to provide a
policing mechanism that isolates intentional loss among an ordered
set of classes. While the ability to do so, by various algorithms,
has been demonstrated, the technical requirement has not. If
dropping random packets from all calls is not appropriate,
concentrating random loss in a subset of the calls makes the problem
for those calls worse; a superior approach would reject or preempt an
entire call.
Parekh's second condition has been met: we must know what the network
will do with the traffic. If the offered load exceeds the available
bandwidth, the network will remark and drop the excess traffic. The
key questions become "How does one limit offered load to a rate less
than or equal to available bandwidth?" and "How much traffic does one
admit with each appropriate marking?"
2.3. Bandwidth Admission Procedure
Since many available voice and video codecs require a nominal loss
rate to deliver acceptable performance, Parekh's first requirement is
that offered load be within the available capacity. There are
several possible approaches.
An approach that is commonly used in H.323 networks is to limit the
number of calls simultaneously accepted by the gatekeeper. SIP
networks do something similar when they place a stateful SIP proxy
near a single ingress/egress to the network. This is able to impose
an upper bound on the total number of calls in the network or the
total number of calls crossing the significant link. However, the
gatekeeper has no knowledge of routing, so the engineering must be
very conservative and usually presumes a single ingress/egress or the
failure of one of its data paths. While this may serve as a short-
term work-around, it is not a general solution that is readily
deployed. This limits the options in network design.
[RFC1633] provides for signaled admission for the use of capacity.
The recommended approach is explicit capacity admission, supporting
the concepts of preemption. An example of such a procedure uses the
Resource Reservation Protocol [RFC2205] [RFC2209] (RSVP). The use of
Capacity Admission using RSVP with SIP is described in [RFC3312].
While call counting is specified in H.323, network capacity admission
is not integrated with H.323 at this time.
2.3.1. RSVP Admission Using Policy for Both Unicast and Multicast
Sessions
RSVP is a resource reservation setup protocol providing the one-way
(at a time) setup of resource reservations for multicast and unicast
flows. Each reservation is set up in one direction (meaning one
reservation from each end system; in a multicast environment, N
senders set up N reservations). These reservations complete a
communication path with a deterministic bandwidth allocation through
each router along that path between end systems. These reservations
set up a known quality of service for end-to-end communications and
maintain a "soft-state" within a node. The meaning of the term "soft
state" is that in the event of a network outage or change of routing,
these reservations are cleared without manual intervention, but must
be periodically refreshed. In RSVP, the refresh period is by default
30 seconds, but may be as long as is appropriate.
RSVP is a locally-oriented process, not a globally- or domain-
oriented one like a routing protocol or H.323 Call Counting.
Although it uses the local routing databases to determine the routing
path, it is only concerned with the quality of service for a
particular or aggregate flow through a device. RSVP is not aware of
anything other than the local goal of QoS and its RSVP-enabled
adjacencies, operating below the network layer. The process by
itself neither requires nor has any end-to-end network knowledge or
state. Thus, RSVP can be effective when it is enabled at some nodes
in a network without the need to have every node participate.
HOST ROUTER
_____________________________ ____________________________
| _______ | | |
| | | _______ | | _______ |
| |Appli- | | | |RSVP | | | |
| | cation| | RSVP <---------------------------> RSVP <---------->
| | <--> | | | _______ | | |
| | | |process| _____ | ||Routing| |process| _____ |
| |_._____| | -->Policy| || <--> -->Policy||
| | |__.__._| |Cntrl|| ||process| |__.__._| |Cntrl||
| |data | | |_____|| ||__.____| | | |_____||
|===|===========|==|==========| |===|==========|==|==========|
| | --------| | _____ | | | --------| | _____ |
| | | | ---->Admis|| | | | | ---->Admis||
| _V__V_ ___V____ |Cntrl|| | _V__V_ __V_____ |Cntrl||
| | | | | |_____|| | | | | ||_____||
| |Class-| | Packet | | | |Class-| | Packet | |
| | ifier|==>Schedulr|================> ifier|==>Schedulr|=========>
| |______| |________| |data | |______| |________| data
| | | |
|_____________________________| |____________________________|
Figure 3: RSVP in Hosts and Routers
Figure 3 shows the internal process of RSVP in both hosts (end
systems) and routers, as shown in [RFC2209].
RSVP uses the phrase "traffic control" to describe the mechanisms of
how a data flow receives quality of service. There are 3 different
mechanisms to traffic control (shown in Figure 2 in both hosts and
routers). They are:
A packet classifier mechanism: This resolves the QoS class for each
packet; this can determine the route as well.
An admission control mechanism: This consists of two decision
modules: admission control and policy control. Determining
whether there are satisfactory resources for the requested QoS is
the function of admission control. Determining whether the user
has the authorization to request such resources is the function of
policy control. If the parameters carried within this flow fail,
either of these two modules errors the request using RSVP.
A packet scheduler mechanism: At each outbound interface, the
scheduler attains the guaranteed QoS for that flow.
2.3.2. RSVP Scaling Issues
As originally written, there was concern that RSVP had scaling
limitations due to its data plane behavior [RFC2208]. This either
has not proven to be the case or has in time largely been corrected.
Telephony services generally require peak call admission rates on the
order of thousands of calls per minute and peak call levels
comparable to the capacities of the lines in question, which is
generally on the order of thousands to tens of thousands of calls.
Current RSVP implementations admit calls at the rate of hundreds of
calls per second and maintain as many calls in progress as memory
configurations allow.
In edge networks, RSVP is used to signal for individual microflows,
admitting the bandwidth. However, Differentiated Services is used
for the data plane behavior. Admission and policing may be performed
anywhere, but need only be performed in the first-hop router (which,
if the end system sending the traffic is a DTE, constitutes a DCE for
the remaining network) and in routers that have interfaces threatened
by congestion. In Figure 1, these would normally be the links that
cross network boundaries.
2.3.3. RSVP Operation in Backbones and Virtual Private Networks (VPNs)
In backbone networks, networks that are normally awash in bandwidth,
RSVP and its affected data flows may be carried in a variety of ways.
If the backbone is a maze of tunnels between its edges (true of MPLS
networks, networks that carry traffic from an encryptor to a
decryptor, and also VPNs), applicable technologies include [RFC2207],
[RFC2746], and [RFC2983]. An IP tunnel is, simplistically put, a IP
packet enveloped inside another IP packet as a payload. When IPv6 is
transported over an IPv4 network, encapsulating the entire v6 packet
inside a v4 packet is an effective means to accomplish this task. In
this type of tunnel, the IPv6 packet is not read by any of the
routers while inside the IPv4 envelope. If the inner packet is RSVP
enabled, there must be an active configuration to ensure that all
relevant backbone nodes read the RSVP fields; [RFC2746] describes
this.
This is similar to how IPsec tunnels work. Encapsulating an RSVP
packet inside an encrypted packet for security purposes without
copying or conveying the RSVP indicators in the outside IP packet
header would make RSVP inoperable while in this form of a tunnel.
[RFC2207] describes how to modify an IPsec packet header to allow for
RSVP awareness by nodes that need to provide QoS for the flow or
flows inside a tunnel.
Other networks may simply choose to aggregate the reservations across
themselves as described in [RFC3175]. The problem with an individual
reservation architecture is that each flow requires a non-trivial
amount of message exchange, computation, and memory resources in each
router between each endpoint. Aggregation of flows reduces the
number of completely individual reservations into groups of
individual flows that can act as one for part or all of the journey
between end systems. Aggregates are not intended to be from the
first router to the last router within a flow, but to cover common
paths of a large number of individual flows.
Examples of aggregated data flows include streams of IP data that
traverse common ingress and egress points in a network and also
include tunnels of various kinds. MPLS LSPs, IPsec Security
Associations between VPN edge routers, IP/IP tunnels, and Generic
Routing Encapsulation (GRE) tunnels all fall into this general
category. The distinguishing factor is that the system injecting an
aggregate into the aggregated network sums the PATH and RESV
statistical information on the un-aggregated side and produces a
reservation for the tunnel on the aggregated side. If the bandwidth
for the tunnel cannot be expanded, RSVP leaves the existing
reservation in place and returns an error to the aggregator, which
can then apply a policy such as IEPS to determine which session to
refuse. In the data plane, the DSCP for the traffic must be copied
from the inner to the outer header, to preserve the PHB's effect.
One concern with this approach is that this leaks information into
the aggregated zone concerning the number of active calls or the
bandwidth they consume. In fact, it does not, as the data itself is
identifiable by aggregator address, deaggregator address, and DSCP.
As such, even if it is not advertised, such information is
measurable.
2.3.4. Interaction with the Differentiated Services Architecture
In the PATH message, the DCLASS object described in [RFC2996] is used
to carry the determined DSCP for the precedence level of that call in
the stream. This is reflected back in the RESV message. The DSCP
will be determined from the authorized SIP message exchange between
end systems by using the R-P header. The DCLASS object permits both
bandwidth admission within a class and the building up of the various
rates or token buckets.
2.3.5. Admission Policy
RSVP's basic admission policy, as defined, is to grant any user
bandwidth if there is bandwidth available within the current
configuration. In other words, if a new request arrives and the
difference between the configured upper bound and the currently
reserved bandwidth is sufficiently large, RSVP grants use of that
bandwidth. This basic policy may be augmented in various ways, such
as using a local or remote policy engine to apply AAA procedures and
further qualify the reservation.
2.3.5.1. Admission for Variable Rate Codecs
For certain applications, such as broadcast video using MPEG-1 or
voice without activity detection and using a constant bit rate codec
such as G.711, this basic policy is adequate apart from AAA. For
variable rate codecs, such as MPEG-4 or a voice codec with Voice
Activity Detection, however, this may be deemed too conservative. In
such cases, two basic types of statistical policy have been studied
and reported on in the literature: simple over-provisioning, and
approximation to ambient load.
Simple over-provisioning sets the bandwidth admission limit higher
than the desired load, on the assumption that a session that admits a
certain bandwidth will in fact use a fraction of the bandwidth. For
example, if MPEG-4 data streams are known to use data rates between
80 and 800 KBPS and there is no obvious reason that sessions would
synchronize (such as having commercial breaks on 15 minute
boundaries), one could imagine estimating that the average session
consumes 400 KBPS and treating an admission of 800 KBPS as actually
consuming half the amount.
One can also approximate to average load, which is perhaps a more
reliable procedure. In this case, one maintains a variable that
measures actual traffic through the admitted data's queue,
approximating it using an exponentially weighted moving average.
When a new reservation request arrives, if the requested rate is less
than the difference between the configured upper bound and the
current value of the moving average, the reservation is accepted, and
the moving average is immediately increased by the amount of the
reservation to ensure that the bandwidth is not promised out to
several users simultaneously. In time, the moving average will decay
from this guard position to an estimate of true load, which may offer
a chance to another session to be reserved that would otherwise have
been refused.
Statistical reservation schemes such as these are overwhelmingly
dependent on the correctness of their configuration and its
appropriateness for the codecs in use. However, they offer the
opportunity to take advantage of statistical multiplexing gains that
might otherwise be missed.
2.3.5.2. Interaction with Complex Admission Policies, AAA, and
Preemption of Bandwidth
Policy is carried and applied as described in [RFC2753]. Figure 4,
below, is the basic conceptual model for policy decisions and
enforcement in an Integrated Services model. This model was created
to provide the ability to monitor and control reservation flows based
on user identify, specific traffic and security requirements, and
conditions that might change for various reasons, including a
reaction to a disaster or emergency event involving the network or
its users.
Network Node Policy server
______________
| ______ |
| | | | _____
| | PEP | | | |------------->
| |______|<---|---->| PDP |May use LDAP,SNMP,COPS...for accessing
| ^ | | | policy database, authentication, etc.
| | | |_____|------------->
| __v___ |
| | | | PDP = Policy Decision Point
| | LPDP | | PEP = Policy Enforcement Point
| |______| | LPDP = Local Policy Decision Point
|______________|
Figure 4: Conceptual Model for Policy Control of Routers
The Network Node represents a router in the network. The Policy
Server represents the point of admission and policy control by the
network operator. Policy Enforcement Point (PEP) (the router) is
where the policy action is carried out. Policy decisions can be
either locally present in the form of a Local Policy Decision Point
(LPDP), or in a separate server on the network called the Policy
Decision Point. The easier the instruction set of rules, the more
likely this set can reside in the LPDP for speed of access reasons.
The more complex the rule set, the more likely this is active on a
remote server. The PDP will use other protocols (LDAP, SNMP, etc.)
to request information (e.g., user authentication and authorization
for precedence level usage) to be used in creating the rule sets of
network components. This remote PDP should also be considered where
non-reactive policies are distributed out to the LPDPs.
Taking the above model as a framework, [RFC2750] extends RSVP's
concept of a simple reservation to include policy controls, including
the concepts of Preemption [RFC3181] and Identity [RFC3182],
specifically speaking to the usage of policies that preempt calls
under the control of either a local or remote policy manager. The
policy manager assigns a precedence level to the admitted data flow.
If it admits a data flow that exceeds the available capacity of a
system, the expectation is that the RSVP-affected RSVP process will
tear down a session among the lowest precedence sessions it has
admitted. The RESV Error resulting from that will go to the receiver
of the data flow and be reported to the application (SIP or H.323).
That application is responsible for disconnecting its call, with a
reason code of "bandwidth preemption".
2.4. Authentication and Authorization of Calls Placed
It will be necessary, of course, to ensure that any policy is applied
to an authenticated user; the capabilities assigned to an
authenticated user may be considered authorized for use in the
network. For bandwidth admission, this will require the utilization
of [RFC2747] [RFC3097]. In SIP and H.323, AAA procedures will also
be needed.
2.5. Defined User Interface
The user interface -- the chimes and tones heard by the user --
should ideally remain the same as in the PSTN for those indications
that are still applicable to an IP network. There should be some new
effort generated to update the list of announcements sent to the user
that don't necessarily apply. All indications to the user, of
course, depend on positive signals, not unreliable measures based on
changing measurements.
3. Security Considerations
This document outlines a networking capability composed entirely of
existing specifications. It has significant security issues, in the
sense that a failure of the various authentication or authorization
procedures can cause a fundamental breakdown in communications.
However, the issues are internal to the various component protocols
and are covered by their various security procedures.
4. Acknowledgements
This document was developed with the knowledge and input of many
people, far too numerous to be mentioned by name. However, key
contributors of thoughts include Francois Le Faucheur, Haluk
Keskiner, Rohan Mahy, Scott Bradner, Scott Morrison, Subha Dhesikan,
and Tony De Simone. Pete Babendreier, Ken Carlberg, and Mike Pierce
provided useful reviews.
5. References
5.1. Normative References
[RFC3689] Carlberg, K. and R. Atkinson, "General Requirements
for Emergency Telecommunication Service (ETS)", RFC
3689, February 2004.
[RFC3690] Carlberg, K. and R. Atkinson, "IP Telephony
Requirements for Emergency Telecommunication
Service (ETS)", RFC 3690, February 2004.
Integrated Services Architecture References
[RFC1633] Braden, B., Clark, D., and S. Shenker, "Integrated
Services in the Internet Architecture: an
Overview", RFC 1633, June 1994.
[RFC2205] Braden, B., Zhang, L., Berson, S., Herzog, S., and
S. Jamin, "Resource ReSerVation Protocol (RSVP) --
Version 1 Functional Specification", RFC 2205,
September 1997.
[RFC2207] Berger, L. and T. O'Malley, "RSVP Extensions for
IPSEC Data Flows", RFC 2207, September 1997.
[RFC2208] Mankin, A., Baker, F., Braden, B., Bradner, S.,
O'Dell, M., Romanow, A., Weinrib, A., and L. Zhang,
"Resource ReSerVation Protocol (RSVP) Version 1
Applicability Statement Some Guidelines on
Deployment", RFC 2208, September 1997.
[RFC2209] Braden, B. and L. Zhang, "Resource ReSerVation
Protocol (RSVP) -- Version 1 Message Processing
Rules", RFC 2209, September 1997.
[RFC2746] Terzis, A., Krawczyk, J., Wroclawski, J., and L.
Zhang, "RSVP Operation Over IP Tunnels", RFC 2746,
January 2000.
[RFC2747] Baker, F., Lindell, B., and M. Talwar, "RSVP
Cryptographic Authentication", RFC 2747, January
2000.
[RFC2750] Herzog, S., "RSVP Extensions for Policy Control",
RFC 2750, January 2000.
[RFC2753] Yavatkar, R., Pendarakis, D., and R. Guerin, "A
Framework for Policy-based Admission Control", RFC
2753, January 2000.
[RFC2996] Bernet, Y., "Format of the RSVP DCLASS Object", RFC
2996, November 2000.
[RFC2998] Bernet, Y., Ford, P., Yavatkar, R., Baker, F.,
Zhang, L., Speer, M., Braden, R., Davie, B.,
Wroclawski, J., and E. Felstaine, "A Framework for
Integrated Services Operation over Diffserv
Networks", RFC 2998, November 2000.
[RFC3097] Braden, R. and L. Zhang, "RSVP Cryptographic
Authentication -- Updated Message Type Value", RFC
3097, April 2001.
[RFC3175] Baker, F., Iturralde, C., Le Faucheur, F., and B.
Davie, "Aggregation of RSVP for IPv4 and IPv6
Reservations", RFC 3175, September 2001.
[RFC3181] Herzog, S., "Signaled Preemption Priority Policy
Element", RFC 3181, October 2001.
[RFC3182] Yadav, S., Yavatkar, R., Pabbati, R., Ford, P.,
Moore, T., Herzog, S., and R. Hess, "Identity
Representation for RSVP", RFC 3182, October 2001.
[RFC3312] Camarillo, G., Marshall, W., and J. Rosenberg,
"Integration of Resource Management and Session
Initiation Protocol (SIP)", RFC 3312, October 2002.
Differentiated Services Architecture References
[RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black,
"Definition of the Differentiated Services Field
(DS Field) in the IPv4 and IPv6 Headers", RFC 2474,
December 1998.
[RFC2475] Blake, S., Black, D., Carlson, M., Davies, E.,
Wang, Z., and W. Weiss, "An Architecture for
Differentiated Services", RFC 2475, December 1998.
[RFC2983] Black, D., "Differentiated Services and Tunnels",
RFC 2983, October 2000.
[RFC3246] Davie, B., Charny, A., Bennet, J., Benson, K., Le
Boudec, J., Courtney, W., Davari, S., Firoiu, V.,
and D. Stiliadis, "An Expedited Forwarding PHB
(Per-Hop Behavior)", RFC 3246, March 2002.
[RFC3247] Charny, A., Bennet, J., Benson, K., Boudec, J.,
Chiu, A., Courtney, W., Davari, S., Firoiu, V.,
Kalmanek, C., and K. Ramakrishnan, "Supplemental
Information for the New Definition of the EF PHB
(Expedited Forwarding Per-Hop Behavior)", RFC 3247,
March 2002.
Session Initiation Protocol and Related References
[RFC2327] Handley, M. and V. Jacobson, "SDP: Session
Description Protocol", RFC 2327, April 1998.
[RFC3261] 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.
[RFC4411] Polk, J., "Extending the Session Initiation
Protocol (SIP) Reason Header for Preemption
Events", RFC 4411, February 2006.
[RFC4412] Schulzrinne, H. and J. Polk, "Communications
Resource Priority for the Session Initiation
Protocol (SIP)", RFC 4412, February 2006.
5.2. Informative References
[ANSI.MLPP.Spec] American National Standards Institute,
"Telecommunications - Integrated Services Digital
Network (ISDN) - Multi-Level Precedence and
Preemption (MLPP) Service Capability", ANSI
T1.619-1992 (R1999), 1992.
[ANSI.MLPP.Supp] American National Standards Institute, "MLPP
Service Domain Cause Value Changes", ANSI ANSI
T1.619a-1994 (R1999), 1990.
[G711.1] Viola Networks, "Netally VoIP Evaluator", January
2003, <http://www.brainworks.de/Site/hersteller/
viola_networks/Dokumente/Compr_Report_Sample.pdf>.
[G711.3] Nortel Networks, "Packet Loss and Packet Loss
Concealment", 2000, <http://www.nortelnetworks.com/
products/01/succession/es/collateral/
tb_pktloss.pdf>.
[ITU.ETS.E106] International Telecommunications Union,
"International Emergency Preference Scheme for
disaster relief operations (IEPS)", ITU-T
Recommendation E.106, October 2003.
[ITU.MLPP.1990] International Telecommunications Union, "Multilevel
Precedence and Preemption Service (MLPP)", ITU-T
Recommendation I.255.3, 1990.
[Parekh1] Parekh, A. and R. Gallager, "A Generalized
Processor Sharing Approach to Flow Control in
Integrated Services Networks: The Multiple Node
Case", INFOCOM 1993: 521-530, 1993.
[Parekh2] Parekh, A. and R. Gallager, "A Generalized
Processor Sharing Approach to Flow Control in
Integrated Services Networks: The Single Node
Case", INFOCOM 1992: 915-924, 1992.
Appendix A. 2-Call Preemption Example Using RSVP
This appendix will present a more complete view of the interaction
among SIP, SDP, and RSVP. The bulk of the material is referenced
from [RFC2327], [RFC3312], [RFC4411], and [RFC4412]. There will be
some discussion on basic RSVP operations regarding reservation paths;
this will be mostly from [RFC2205].
SIP signaling occurs at the Application Layer, riding on a UDP/IP or
TCP/IP (including TLS/TCP/IP) transport that is bound by routing
protocols such as BGP and OSPF to determine the route the packets
traverse through a network between source and destination devices.
RSVP is riding on top of IP as well, which means RSVP is at the mercy
of the IP routing protocols to determine a path through the network
between endpoints. RSVP is not a routing protocol. In this
appendix, there will be an escalation of building blocks getting to
how the many layers are involved in SIP. QoS Preconditions require
successful RSVP signaling between endpoints prior to SIP successfully
acknowledging the setup of the session (for voice, video, or both).
Then we will present what occurs when a network overload occurs
(congestion), causing a SIP session to be preempted.
Three diagrams in this appendix show multiple views of the same
example of connectivity for discussion throughout this appendix. The
first diagram (Figure 5) is of many routers between many endpoints
(SIP user agents, or UAs). There are 4 UAs of interest; those are
for users Alice, Bob, Carol, and Dave. When a user (the human) of a
UA gets involved and must do something to a UA to progress a SIP
process, this will be explicitly mentioned to avoid confusion;
otherwise, when Alice is referred to, it means Alice's UA (her
phone).
RSVP reserves bandwidth in one direction only (the direction of the
RESV message), as has been discussed, IP forwarding of packets are
dictated by the routing protocol for that portion of the
infrastructure from the point of view of where the packet is to go
next.
The RESV message traverses the routers in the reverse path taken by
the PATH message. The PATH message establishes a record of the route
taken through a network portion to the destination endpoint, but it
does not reserve resources (bandwidth). The RESV message back to the
original requester of the RSVP flow requests for the bandwidth
resources. This means the endpoint that initiates the RESV message
controls the parameters of the reservation. This document specifies
in the body text that the SIP initiator (the UAC) establishes the
parameters of the session in an INVITE message, and that the INVITE
recipient (the UAS) must follow the parameters established in that
INVITE message. One exception to this is which codec to use if the
UAC offered more than one to the UAS. This exception will be shown
when the INVITE message is discussed in detail later in the appendix.
If there was only one codec in the SDP of the INVITE message, the
parameters of the reservation will follow what the UAC requested
(specifically to include the Resource-Priority header namespace and
priority value).
Here is the first figure with the 4 UAs and a meshed routed
infrastructure between each. For simplicity of this explanation,
this appendix will only discuss the reservations from Alice to Bob
(one direction) and from Carol to Dave (one direction). An
interactive voice service will require two one-way reservations that
end in each UA. This gives the appearance of a two-way reservation,
when indeed it is not.
Alice -----R1----R2----R3----R4------ Bob
| \ / \ / \ / |
| \/ \/ \/ |
| /\ /\ /\ |
| / \ / \ / \ |
Carol -----R5----R6----R7----R8------ Dave
Figure 5: Complex Routing and Reservation Topology
The PATH message from Alice to Bob (establishing the route for the
RESV message) will be through routers:
Alice -> R1 -> R2 -> R3 -> R4 -> Bob
The RESV message (and therefore the reservation of resources) from
Bob to Alice will be through routers:
Bob -> R4 -> R3 -> R2 -> R1 -> Alice
The PATH message from Carol to Dave (establishing the route for the
RESV message) will be through routers:
Carol -> R5 -> R2 -> R3 -> R8 -> Dave
The RESV message (and therefore the reservation of resources) from
Dave to Carol will be through routers:
Dave -> R8 -> R3 -> R2 -> R5 -> Carol
The reservations from Alice to Bob traverse a common router link:
between R3 and R2 and thus a common interface at R2. Here is where
there will be congestion in this example, on the link between R2 and
R3. Since the flow of data (in this case voice media packets)
travels the direction of the PATH message, and RSVP establishes
reservation of resources at the egress interface of a router, the
interface in Figure 6 shows that Int7 will be what first knows about
a congestion condition.
Alice Bob
\ /
\ /
+--------+ +--------+
| | | |
| R2 | | R3 |
| Int7-------Int5 |
| | | |
+--------+ +--------+
/ \
/ \
Carol Dave
Figure 6: Reduced Reservation Topology
Figure 6 illustrates how the messaging between the UAs and the RSVP
messages between the relevant routers can be shown to understand the
binding that was established in [RFC3312] (more suitably titled "SIP
Preconditions for QoS" from this document's point of view).
We will assume all devices have powered up and received whatever
registration or remote policy downloads were necessary for proper
operation. The routing protocol of choice has performed its routing
table update throughout this part of the network. Now we are left to
focus only on end-to-end communications and how that affects the
infrastructure between endpoints.
The next diagram (Figure 7) (nearly identical to Figure 1 from
[RFC3312]) shows the minimum SIP messaging (at layer 7) between Alice
and Bob for a good-quality voice call. The SIP messages are numbered
to identify special qualities of each. During the SIP signaling,
RSVP will be initiated. That messaging will also be discussed below.
UA Alice UA Bob
| |
| |
|-------------(1) INVITE SDP1--------------->|
| | Note 1
|<------(2) 183 Session Progress SDP2--------| |
***|********************************************|***<-+
* |----------------(3) PRACK------------------>| *
* | | * Where
* |<-----------(4) 200 OK (PRACK)--------------| * RSVP
* | | * is
* | | * signaled
***|********************************************|***
|-------------(5) UPDATE SDP3--------------->|
| |
|<--------(6) 200 OK (UPDATE) SDP4-----------|
| |
|<-------------(7) 180 Ringing---------------|
| |
|-----------------(8) PRACK----------------->|
| |
|<------------(9) 200 OK (PRACK)-------------|
| |
| |
|<-----------(10) 200 OK (INVITE)------------|
| |
|------------------(11) ACK----------------->|
| |
| RTP (within the reservation) |
|<==========================================>|
| |
Figure 7: SIP Reservation Establishment Using Preconditions
The session initiation starts with Alice wanting to communicate with
Bob. Alice decides on an IEPS precedence level for their call (the
default is the "routine" level, which is for normal everyday calls,
but a priority level has to be chosen for each call). Alice puts
into her UA Bob's address and precedence level and (effectively) hits
the send button. This is reflected in SIP with an INVITE Method
Request message [M1]. Below is what SIP folks call a well-formed SIP
message (meaning it has all the headers that are mandatory to
function properly). We will pick on the US Marine Corps (USMC) for
the addressing of this message exchange.
[M1 - INVITE from Alice to Bob, RP=Routine, QOS=e2e and mandatory]
INVITE sip:bob@usmc.example.mil SIP/2.0
Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
To: Bob <sip:bob@usmc.example.mil>
Call-ID: 3848276298220188511@pc33.usmc.example.mil
CSeq: 31862 INVITE
Require: 100rel, preconditions, resource-priority
Resource-Priority: dsn.routine
Contact: <sip:alice@usmc.example.mil>
Content-Type: application/sdp
Content-Length: 191
v=0
o=alice 2890844526 2890844526 IN IP4 usmc.example.mil
c=IN IP4 10.1.3.33
t=0 0
m=audio 49172 RTP/AVP 0 4 8
a=rtpmap:0 PCMU/8000
a=curr:qos e2e none
a=des:qos mandatory e2e sendrecv
From the INVITE above, Alice is inviting Bob to a session. The upper
half of the lines (above the line "v=0") is SIP headers and header
values, and the lower half is Session Description Protocol (SDP)
lines. SIP headers (after the first line, called the Status line)
are not mandated in any particular order, with one exception: the Via
header. It is a SIP hop (through a SIP Proxy) route path that has a
new Via header line added by each SIP element this message traverses
towards the destination UA. This is similar in function to an RSVP
PATH message (building a reverse path back to the originator of the
message). At any point in the message's path, a SIP element knows
the path to the originator of the message. There will be no SIP
Proxies in this example, because for Preconditions, Proxies only make
more messages that look identical (with the exception of the Via and
Max-Forwards headers), and it is not worth the space here to
replicate what has been done in SIP RFCs already.
SIP headers that are used for Preconditions are as follows:
o Require header, which contains 3 option tags: "100rel" mandates a
reliable provisional response message to the conditions requesting
in this INVITE (knowing they are special), "preconditions"
mandates that preconditions are attempted, and "resource-priority"
mandates support for the Resource-Priority header. Each of these
option tags can be explicitly identified in a message failure
indication from the called UA to tell the calling UA exactly what
was not supported.
Provided that this INVITE message is received as acceptable, this
will result in the 183 "Session Progress" message from Bob's UA, a
reliable confirmation that preconditions are required for this
call.
o Resource-Priority header, which denotes the domain namespace and
precedence level of the call on an end-to-end basis.
This completes SIP's functions in session initiation. Preconditions
are requested, required, and signaled for in the SDP portion of the
message. SDP is carried in what's called a SIP message body (much
like the text in an email message is carried). SDP has special
properties (see [RFC2327] for more on SDP, or the MMUSIC WG for
ongoing efforts regarding SDP). SDP lines are in a specific order
for parsing by end systems. Dialog-generating (or call-generating)
SDP message bodies all must have an "m=" line (or media description
line). Following the "m=" line are zero or more "a=" lines (or
Attribute lines). The "m=" line in Alice's INVITE calls for a voice
session (this is where video is identified also) using one of 3
different codecs that Alice supports (0 = G.711, 4 = G.723, and 18 =
G.729) that Bob gets to choose from for this session. Bob can choose
any of the 3. The first a=rtpmap line is specific to the type of
codec these 3 are (PCMU). The next two "a=" lines are the only
identifiers that RSVP is to be used for this call. The second "a="
line:
a=curr:qos e2e none
identifies the "current" status of qos at Alice's UA. Note:
everything in SDP is with respect to the sender of the SDP message
body (Alice will never tell Bob how his SDP is; she will only tell
Bob about her SDP).
"e2e" means that capacity assurance is required from Alice's UA to
Bob's UA; thus, a lack of available capacity assurance in either
direction will fail the call attempt.
"none" means there is no reservation at Alice's UA (to Bob) at
this time.
The final "a=" line (a=des) identifies the "desired" level of qos:
a=des:qos mandatory e2e sendrecv
"mandatory" means this request for qos MUST be successful, or the
call fails.
"e2e" means RSVP is required from Alice's UA to Bob's UA.
"sendrecv" means the reservation is in both directions.
As discussed, RSVP does not reserve bandwidth in both directions, and
it is up to the endpoints to have 2 one-way reservations if that
particular application (here, voice) requires it. Voice between
Alice and Bob requires 2 one-way reservations. The UAs will be the
focal points for both reservations in both directions.
Message 2 is the 183 "Session Progress" message sent by Bob to Alice,
which indicates to Alice that Bob understands that preconditions are
required for this call.
[M2 - 183 "Session Progress"]
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
;branch=z9hG4bK74bf9 ;received=10.1.3.33
From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
To: Bob <sip:bob@usmc.example.mil>;tag=8321234356
Call-ID: 3848276298220188511@pc33.usmc.example.mil
CSeq: 31862 INVITE
RSeq: 813520
Resource-Priority: dsn.routine
Contact: <sip:bob@usmc.example.mil>
Content-Type: application/sdp
Content-Length: 210
v=0
o=bob 2890844527 2890844527 IN IP4 usmc.example.mil
c=IN IP4 10.100.50.51
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=curr:qos e2e none
a=des:qos mandatory e2e sendrecv
a=conf:qos e2e recv
The only interesting header in the SIP portion of this message is the
RSeq header, which is the "Reliable Sequence" header. The value is
incremented for every Reliable message that's sent in this call setup
(to make sure none are lost or to ignore duplicates).
Bob's SDP indicates several "a=" line statuses and picks a codec for
the call. The codec picked is in the m=audio line (the "0" at the
end of this line means G.711 will be the codec).
The a=curr line gives Alice Bob's status with regard to RSVP
(currently "none").
The a=des line also states the desire for mandatory qos e2e in both
directions.
The a=conf line is new. This line means Bob wants confirmation that
Alice has 2 one-way reservations before Bob's UA proceeds with the
SIP session setup.
This is where "Note-1" applies in Figure 7. At the point that Bob's
UA transmits this 183 message, Bob's UA (the one that picked the
codec, so it knows the amount of bandwidth to reserve) transmits an
RSVP PATH message to Alice's UA. This PATH message will take the
route previously discussed in Figure 5:
Bob -> R4 -> R3 -> R2 -> R1 -> Alice
This is the path of the PATH message, and the reverse will be the
path of the reservation setup RESV message, or:
Alice -> R1 -> R2 -> R3 -> R4 -> Bob
Immediately after Alice transmits the RESV message towards Bob, Alice
sends her own PATH message to initiate the other one-way reservation.
Bob, receiving that PATH message, will reply with a RESV.
All this is independent of SIP. However, during this time of
reservation establishment, a Provisional Acknowledgement (PRACK) [M3]
is sent from Alice to Bob to confirm the request for confirmation of
2 one-way reservations at Alice's UA. This message is acknowledged
with a normal 200 OK message [M4]. This is shown in Figure 7.
As soon as the RSVP is successfully completed at Alice's UA (knowing
that it was the last in the two-way cycle or reservation
establishment), at the SIP layer an UPDATE message [M5] is sent to
Bob's UA to inform his UA that the current status of RSVP (or qos) is
"e2e" and "sendrecv".
[M5 - UPDATE to Bob that Alice has qos e2e and sendrecv]
UPDATE sip:bob@usmc.example.mil SIP/2.0
Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
;branch=z9hG4bK74bfa
From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
To: Bob <sip:bob@usmc.example.mil>
Call-ID: 3848276298220188511@pc33.usmc.example.mil
Resource-Priority: dsn.routine
Contact: <sip:alice@usmc.example.mil>
CSeq: 10197 UPDATE
Content-Type: application/sdp
Content-Length: 191
v=0
o=alice 2890844528 2890844528 IN IP4 usmc.example.mil
c=IN IP4 10.1.3.33
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=curr:qos e2e send
a=des:qos mandatory e2e sendrecv
This is shown by the matching table that can be built from the a=curr
line and a=des line. If the two lines match, then no further
signaling needs take place with regard to "qos". [M6] is the 200 OK
acknowledgement of this synchronization between the two UAs.
[M6 - 200 OK to the UPDATE from Bob indicating synchronization]
SIP/2.0 200 OK sip:bob@usmc.example.mil
Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
;branch=z9hG4bK74bfa
From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
To: Bob <sip:bob@usmc.example.mil>
Call-ID: 3848276298220188511@pc33.usmc.example.mil
Resource-Priority: dsn.routine
Contact: < sip:alice@usmc.example.mil >
CSeq: 10197 UPDATE
Content-Type: application/sdp
Content-Length: 195
v=0
o=alice 2890844529 2890844529 IN IP4 usmc.example.mil
c=IN IP4 10.1.3.33
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=curr:qos e2e sendrecv
a=des:qos mandatory e2e sendrecv
At this point, the reservation is operational and both UAs know it.
Bob's UA now rings, telling Bob the user that Alice is calling him.
([M7] is the SIP indication to Alice that this is taking place).
Nothing up until now has involved Bob the user. Bob picks up the
phone (generating [M10], from which Alice's UA responds with the
final ACK), and RTP is now operating within the reservations between
the two UAs.
Now we get to Carol calling Dave. Figure 6 shows a common router
interface for the reservation between Alice to Bob, and one that will
also be the route for one of the reservations between Carol to Dave.
This interface will experience congestion in our example.
Carol is now calling Dave at a Resource-Priority level of
"Immediate", which is higher in priority than Alice to Bob's
"routine". In this continuing example, Router 2's Interface-7 is
congested and cannot accept any more RSVP traffic. Perhaps the
offered load is at interface capacity. Perhaps Interface-7 is
configured with a fixed amount of bandwidth it can allocate for RSVP
traffic, and it has reached its maximum without one of the
reservations going away through normal termination or forced
termination (preemption).
Interface-7 is not so full of offered load that it cannot transmit
signaling packets, such as Carol's SIP messaging to set up a call to
Dave. This should be by design (that not all RSVP traffic can starve
an interface from signaling packets). Carol sends her own INVITE
with the following important characteristics:
[M1 - INVITE from Carol to Dave, RP=Immediate, QOS=e2e and mandatory]
This packet does *not* affect the reservations between Alice and Bob
(SIP and RSVP are at different layers, and all routers are passing
signaling packets without problems). Dave sends his M2:
[M2 - 183 "Session Progress"]
with the SDP chart of:
a=curr:qos e2e none
a=des:qos mandatory e2e sendrecv
a=conf:qos e2e recv
indicating he understands RSVP reservations are required e2e for this
call to be considered successful. Dave sends his PATH message. The
PATH message does *not* affect Alice's reservation; it merely
establishes a path for the RESV reservation setup message to take.
To keep this example simple, the PATH message from Dave to Carol took
this route (which we make different from the route in the reverse
direction):
Dave -> R8 -> R7 -> R6 -> R5 -> Carol
causing the reservation to be this route:
Carol -> R5 -> R6 -> R7 -> R8 -> Dave
The Carol-to-Dave reservation above will not traverse any of the same
routers as the Alice-to-Bob reservation. When Carol transmits her
RESV message towards Dave, she immediately transmits her PATH message
to set up the complementary reservation.
The PATH message from Carol to Dave be through routers:
Carol -> R5 -> R2 -> R3 -> R8 -> Dave
Thus, the RESV message will be through routers:
Dave -> R8 -> R3 -> R2 -> R5 -> Carol
This RESV message will traverse the same routers, R3 and R2, as the
Alice-to-Bob reservation. This RESV message, when received at
Interface-7 of R2, will create a congestion situation such that R2
will need to make a decision on whether:
o to keep the Alice-to-Bob reservation and error the new RESV from
Dave, or
o to error the reservation from Alice to Bob in order to make room
for the Carol-to-Dave reservation.
Alice's reservation was set up in SIP at the "routine" precedence
level. This will equate to a comparable RSVP priority number (RSVP
has 65,535 priority values, or 2*32 bits per [RFC3181]). Dave's RESV
equates to a precedence value of "immediate", which is a higher
priority. Thus, R2 will preempt the reservation from Alice to Bob
and allow the reservation request from Dave to Carol. The proper
RSVP error is the ResvErr that indicates preemption. This message
travels downstream towards the originator of the RESV message (Bob).
This clears the reservation in all routers downstream of R2 (meaning
R3 and R4). Once Bob receives the ResvErr message indicating
preemption has occurred on this reservation, Bob's UA transmits a SIP
preemption indication back towards Alice's UA. This accomplishes two
things: first, it informs all SIP Servers that were in the session
setup path that wanted to remain "dialog stateful" per [RFC3261], and
second, it informs Alice's UA that this was a purposeful termination,
and to play a preemption tone. The proper indication in SIP of this
termination due to preemption is a BYE Method message that includes a
Reason Header indicating why this occurred (in this case, "Reserved
Resources Preempted"). Here is the message from Bob to Alice that
terminates the call in SIP.
BYE sip:alice@usmc.example.mil SIP/2.0
Via: SIP/2.0/TCP swp34.usmc.example.mil
;branch=z9hG4bK776asegma
To: Alice <sip:alice@usmc.example.mil>
From: Bob <sip:bob@usmc.example.mil>;tag=192820774
Reason: preemption ;cause=2 ;text=reserved resourced preempted
Call-ID: 3848276298220188511@pc33.usmc.example.mil
CSeq: 6187 BYE
Contact: <sip:bob@usmc.example.mil>
When Alice's UA receives this message, her UA terminates the call,
sends a 200 OK to Bob to confirm reception of the BYE message, and
plays a preemption tone to Alice the user.
The RESV message from Dave successfully traverses R2, and Carol's UA
receives it. Just as with the Alice-to-Bob call setup, Carol sends
an UPDATE message to Dave, confirming she has QoS "e2e" in "sendrecv"
directions. Bob acknowledges this with a 200 OK that gives his
current status (QoS "e2e" and "sendrecv"), and the call setup in SIP
continues to completion.
In summary, Alice set up a call to Bob with RSVP at a priority level
of Routine. When Carol called Dave at a high priority, their call
would have preempted any lower priority calls if there were a
contention for resources. In this case, it occurred and affected the
call between Alice and Bob. A router at this congestion point
preempted Alice's call to Bob in order to place the higher-priority
call between Carol and Dave. Alice and Bob were both informed of the
preemption event. Both Alice and Bob's UAs played preemption
indications. What was not mentioned in this appendix was that this
document RECOMMENDS that router R2 (in this example) generate a
syslog message to the domain administrator to properly manage and
track such events within this domain. This will ensure that the
domain administrators have recorded knowledge of where such events
occur, and what the conditions were that caused them.
Authors' Addresses
Fred Baker
Cisco Systems
1121 Via Del Rey
Santa Barbara, California 93117
USA
Phone: +1-408-526-4257
Fax: +1-413-473-2403
EMail: fred@cisco.com
James Polk
Cisco Systems
2200 East President George Bush Turnpike
Richardson, Texas 75082
USA
Phone: +1-817-271-3552
EMail: jmpolk@cisco.com
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