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. 126.96.36.199. 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. 188.8.131.52. 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:firstname.lastname@example.org SIP/2.0 Via: SIP/2.0/TCP pc33.usmc.example.mil:5060 ;branch=z9hG4bK74bf9 Max-Forwards: 70 From: Alice <sip:email@example.com>;tag=9fxced76sl To: Bob <sip:firstname.lastname@example.org> Call-ID: email@example.com CSeq: 31862 INVITE Require: 100rel, preconditions, resource-priority Resource-Priority: dsn.routine Contact: <sip:firstname.lastname@example.org> 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:email@example.com>;tag=9fxced76sl To: Bob <sip:firstname.lastname@example.org>;tag=8321234356 Call-ID: email@example.com CSeq: 31862 INVITE RSeq: 813520 Resource-Priority: dsn.routine Contact: <sip:firstname.lastname@example.org> 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:email@example.com SIP/2.0 Via: SIP/2.0/TCP pc33.usmc.example.mil:5060 ;branch=z9hG4bK74bfa From: Alice <sip:firstname.lastname@example.org>;tag=9fxced76sl To: Bob <sip:email@example.com> Call-ID: firstname.lastname@example.org Resource-Priority: dsn.routine Contact: <sip:email@example.com> 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:firstname.lastname@example.org Via: SIP/2.0/TCP pc33.usmc.example.mil:5060 ;branch=z9hG4bK74bfa From: Alice <sip:email@example.com>;tag=9fxced76sl To: Bob <sip:firstname.lastname@example.org> Call-ID: email@example.com Resource-Priority: dsn.routine Contact: < sip:firstname.lastname@example.org > 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:email@example.com SIP/2.0 Via: SIP/2.0/TCP swp34.usmc.example.mil ;branch=z9hG4bK776asegma To: Alice <sip:firstname.lastname@example.org> From: Bob <sip:email@example.com>;tag=192820774 Reason: preemption ;cause=2 ;text=reserved resourced preempted Call-ID: firstname.lastname@example.org CSeq: 6187 BYE Contact: <sip:email@example.com> 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: firstname.lastname@example.org James Polk Cisco Systems 2200 East President George Bush Turnpike Richardson, Texas 75082 USA Phone: +1-817-271-3552 EMail: email@example.com
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