Special Feature

Priority Promotion Scheme (PPS)—An Autonomous and Distributed Admission
Control for End-to-end Quality of Service for Interactive Multimedia Services
Shunsuke Mori, Yasuro Kawarasaki, Hideki Kataoka,
and Naotaka Morita†
Abstract
Quality of service (QoS) control, especially admission control, is essential for satisfactory end-to-end
session-type communications such as VoIP (voice over IP) or video chatting over an IP (Internet protocol) network. Here, we introduce the priority promotion scheme as an autonomous and distributed
admission control technique suitable for large and diversified networks. The appropriate combination of
the underlying principle of the Internet with the carrier’s needs makes end-to-end QoS, which used tolook hard to achieve, more practical and deployable soon.

1. Need for admission control technology
In the conventional form of telecommunication
where the telephone service is provided over a circuit-switched network, the line from the caller to thereceiver is reserved at each switch at the time thecaller dials. Thus, the call is established only if theentire circuit can be reserved; if the circuit cannot be
reserved, the request to establish communication isrejected. Each circuit has a fixed bandwidth of 64
kbit/s, and no mutual bandwidth interference occurs.
This special mechanism of circuit switching guarantees stable communication quality after call establishment and has provided customers with highly satisfactory network service.
However, consider the provision of interactive multimedia communication services such as VoIP, videophone, or video conferencing provided over an IP
network. Quality degradation at the packet level due
to low transport speed or router processing delay isalready being eliminated, and the VoIP service iscoming into its own. Looking at it at the session level,
the decision as to whether or not communication canbe set up is made by SIP (session initiation protocol)
proxy machines, which are different from routers. In† NTT Network Service Systems Laboratories
Musashino-shi, 180-8585 Japan
E-mail: morita.naotaka@lab.ntt.co.jp

22

that case, the admission decision is made without
regard for the router and transport path usage situation, so communication quality is not strictly guaranteed. In the worst case, the bandwidth of the transport
path is insufficient, and intermittent data loss occurs.
That can lead to disruption of the video stream andthe breaking up of the voice stream, which reducesuser satisfaction. It can also affect communicationthat has already been established, so the degradationof quality during communication, which does not
happen in conventional circuit-switched networks,
may make it difficult to apply usage-based charging.
Considering extension to collaboration-type communication services, in which image communication isimportant, session-by-session admission control
technology is required.
2. Application of admission control technology
to a large-scale network
Various methods referred to as admission control
have been proposed as a means of checking for surplus bandwidth on the transport path and deciding
whether or not communication can be establishedprior to the establishment of a new session [1]. All
these methods are claimed to be usable in a largescale network of several million subscribers, but several questions remain, such as whether or not they areapplicable to diverse networks that include user netNTT Technical Review

Special Feature

works and corporate networks, whether they can be
implemented economically, and how high the barriers
to introduction are.
One key to the success of the Internet is that minimal control is implemented in the network itself:
most functions are implemented in the end systems.
That approach allows a low-cost network configuration. Learning from this lesson, we turned our attention to a method of implementing admission control
by distributing over the terminals the processing thathas been centralized in the nodes of the conventional
network. Based on the fundamental idea proposed in[2], we have extended measurement-based admission
control to a comprehensive network system toachieve end-to-end QoS (quality of service).
3. Introduction to priority promotion scheme
The method for implementing autonomous and distributed quality control is called the priority promotion scheme (PPS) [3]. PPS deals with important
issues that pose problems for a practical service. It
begins with the remaining bandwidth estimation bymeasurement: to confirm that the transport path has
sufficient bandwidth remaining to establish a new
session, the terminal itself sends trial packets
between the source and destination terminals at the
bandwidth needed for the new session. The destination terminal measures the quality degradation when
the trial packets are received and estimates the available bandwidth on the basis of that information todecide whether or not to admit the session (Fig. 1). In
PPS, we initially selected the packet loss rate as a criterion, because it is easy to handle and can show theremaining bandwidth.

To ensure that trial packets do not degrade the quality of existing communication sessions and to clearly
reveal degradation in the trial packets caused byinsufficient bandwidth, a variant of differentiated services (DiffServ) priority control is used within the
network, with the trial packets assigned a low priority and the actual media packets assigned a high priority for transmission. If the reception quality of thetrial packets is good, the source judges that the new
session can be admitted and the priority is then raised(promoted) from low to high. The priority promotion
scheme gets its name from this feature. If the reception quality is not good, the session will be terminated or suspended and given another chance later. The
use of the trial affects how the service appears to theend-user, as described later.
The overall configuration diagram with relevant
protocols for PPS is illustrated in Fig. 2. We assume
SIP for the session control provided by the SIP proxy.
The packets are transmitted by realtime transport protocol (RTP). The packets for testing the reception
quality use realtime transport control protocol(RTCP). These protocols are all standardized ones.
The transport system (routers, layer-2 switches, etc.)
performs DiffServ priority control, and the actual
media packets are transmitted with higher priority
than the trial packets. Considering that the interactive
multimedia service provided by PPS should share thesame path with other non-realtime services, the upper
limit of the available bandwidth is regulated for PPS.
This characteristic is described later. When PPS is
implemented, this transfer operation, which is calledmeasurable forwarding per-hop behavior (MF-PHB),
is assumed to be implemented at each node in thetransport system.

Source terminal

Destination terminal

Transmission of trial packets:
Low priority

Qualitymeasurement

Report of reception quality

Admission judgment(success)

Transmission of actual media packets:
High priority

Fig. 1. PPS terminal behavior.

Vol. 2 No. 10 Oct. 200423

Special Feature

Customer management,
monitoring, and charging
Access control andprimary monitoring

Measurement andadmission control

Secondary monitoring
SIP
proxy

SIP
Edge router

Source terminal

SIP

Media
monitoring
server

Generation ofreception report
Edge routerCore router

Destination terminal

RTP/RTCP
Priority control(MF-PHB)

Fig. 2. Overall PPS configuration diagram.

4. Monitoring to identify invalid operation
In general, in packet network admission control,
the admission decision is based on the initial communication bandwidth report, so if more packets than thereported value flow into the network because ofincorrect or malicious terminal operations, then theactual packet volume will deviate from the initiallyassumed estimate, making it impossible to satisfy thedesired communication quality. Therefore, to maintain quality within the network, packets other than theadmitted media packets and any packets that exceed
the admitted bandwidth must be stopped at the edge
routers or some such location. This is a kind of access
control and primary monitoring. In Fig. 2, it is shown
performed at the edge router.
In PPS, because the admission decision is made at
the terminal, there is concern that a false report ofsuccessful reception might be issued as a result ofincorrect or malicious terminal operations, eventhough the quality of the trial packets was not acceptable. If any equipment in the network is carelessabout this, the quality in the network cannot be maintained. This is a fatal drawback for terminal-oriented
techniques.
To prevent such a situation, we devised a methodthat involves selective monitoring of terminal operation by a media monitoring server (MMS). When amedia stream is selected for monitoring, the SIP
proxy reports the IP address of the MMS instead of
the IP address of the original destination in the communication sent in response to the SIP request of theterminal being monitored. The result is that the originating terminal begins sending the media stream to24

the MMS, which allows the MMS to monitor for
legitimate operation and relay the media stream. This
concept achieves economical deployment of PPS in areal commercial network, while following the terminal-oriented Internet concept.
5. Coping with variable-bandwidth media streams
With PPS, in order to make the admission decision,
the terminal estimates the available transport pathbandwidth by sending trial packets. The available
bandwidth is the transport path bandwidth minus the
bandwidth being used by other terminals at the
moment. Fluctuation in the bandwidth being used byother terminals may reduce the accuracy of the estimation and result in degradation of the quality of theadmitted session. In video image communication for
example, the amount of bandwidth needed depends
strongly on movement in the video scene. Therefore,
if the available bandwidth is estimated at a time when
there is little movement in the video of the establishedsessions of the other terminals, the used bandwidth
will be small and the available bandwidth will beoverestimated. If the admission decision is made onthat basis, it will be judged that the transmission canbe received without degradation of quality, but actually the transport path bandwidth might be exceeded
when the other terminals resume communication at
the reported bandwidths, thus increasing the overall
used bandwidth. This is another serious issue inherent to measurement-based admission control.
We therefore devised an extension to the queuing
scheme at the router (MF-PHB). It transfers previously established sessions but limits the bandwidth
NTT Technical Review

Special Featureallotted to the trial packets. This method can completely avoid quality degradation in established sessions, even if media streams have variable bandwidth
requirements.
6. Measurable Forwarding Per-Hop Behavior(MF-PHB)
Before explaining our extension to MF-PHB, let us
review the basic operation of MF-PHB. To distinguish between them, we use the term ‘static mode’ torefer to the ordinary method and ‘elastic mode’ torefer to the extended method. The bandwidth usage
states of these two modes of MF-PHB are shown in
Fig. 3. The vertical axis in the figure represents theused bandwidth and the horizontal axis representstime.
In the static mode, to reserve a constant bandwidth
for non-PPS services (e.g., best-effort data communication), the bandwidth is kept below an upper limit,
BWmf. Then, at the same time, BWmf is reserved for
PPS use so that the bandwidth available to PPS services is not reduced by the use of the other services.
Neither the trial packets nor the actual media stream
used by PPS can use more bandwidth than BWmf
(Fig. 3(a)).
The elastic mode, however, features separate upper
bandwidth limits for the trial and actual media

H: High-priority actual media stream
M: Low-priority trial media stream
Parentheses indicate media stream suffering packet loss.

（Ｈ）
（M）
M

H

M

BWmf: Maximum bandwidth available for the trial and actual media streams.
BWm: Trial media stream is not forwarded if the total flow exceeds this value.
BWh: Actual media stream is not forwarded if the total flow exceeds this value.

Media stream withvarying bandwidth
usage experiences
packet loss whenBWmf is exceeded.

（Ｈ）

H

Having explained the technical aspects of PPS endto-end admission control, we now introduce someexamples of services that can be implemented with
PPS. PPS provides the user with a means of confirming that communication can be accomplished without

Even for a variable-bandwidth
media stream, transfer can bedone without packet loss
provided BWh is not exceeded.

Transport
bandwidth

（M）
BWmf

7. Examples of services using PPS

H

BWh
Used bandwidth

Used bandwidth

Transport
bandwidth

Trial media streamthat exceeds BWmf
experiences packet
loss, so transfer ofactual media stream
does not begin.

streams. For a total flow of trial and actual media
streams of up to BWm, either type of media stream
can be sent without packet loss. If BWm is exceeded,
however, some trial packets are discarded, while actual media packets are transferred. In this way, new sessions are admitted as long as BWm is not exceeded,
but are rejected if it is exceeded. Thus, transmission
without loss of actual packets is possible, even if the
bandwidth used by the actual media stream changes
temporarily and is transmitted at a bandwidth higher
than BWm. If an upper limit is not set for the bandwidth available to the actual media stream, then the
bandwidth may become unavailable for use by otherservices. Therefore, the maximum bandwidth BWh isset (Fig. 3(b)).
When the elastic mode is applied, PPS can be usedin usage scenarios such as the switching operation at
the time of line failure and hand-over in a mobile
environment. There are significant benefits when weapply PPS to a commercial network.

（M）
Ｈ
BWm

H
When the used bandwidth is below BWmf, thetrial media stream is forwarded, so transfer of
the actual media stream can begin.

Ｈ

（M）
M

H

H

M

H
H

Time

(a) Static modeTime(b) Elastic mode

Fig. 3. MF-PHB router output.

Vol. 2 No. 10 Oct. 2004

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Special Feature

CallerSuccess

Start thecommunication

CalleeCallerCalleeINVITEINVITE

Session progressRinging

Quality measurement

OK
Low-quality
communication

Quality report (RTCP)

Start thecommunication

Ringing

ReINVITE

OK
High-quality
communication

Quality measurement
Success

Quality report (RTCP)
High-quality
communication

Higher quality(a) Pre-arranged guaranteedbandwidth service(b) Service with bandwidth reservation
during communication

Fig. 4. Service examples.

quality degradation by actually conducting a trial
transmission. The trial packets can be used in various
services, depending on what stage of communication
the trial packets are transmitted.
(1) Pre-arranged guaranteed-bandwidth service(Fig. 4(a))
This is the same as the current telephone service.
When the user on the originating side (the caller) lifts
the receiver and dials, trial packets are transmitted at
the bandwidth corresponding to the desired video
image size and sound quality. Likewise, the terminalon the receiving side (the callee) also begins transmitting trial packets. If it is found that there is no
quality degradation in either direction, transmissionof the actual media stream begins and the receiving
terminal rings. The conversation begins when thereceiving user picks up the receiver. If quality degradation is found for the trial packets traveling in either
direction, the caller is informed that the line is busy
and transmission of the trial packets is stopped.
Depending on the estimated available bandwidth,communication with a smaller image or lower audio
quality may be possible.
(2) Service with bandwidth reservation duringcommunication (Fig. 4(b))
When the caller dials, the callee’s terminal rings
immediately. If the caller answers, transmission by
best-effort or other non-PPS priority begins. Then, if
the users feel that the quality is poor and choose highquality PPS communication, even if they have to pay
for it, a PPS admission trial is executed and the users
26

can switch to the high-priority communication if the
bandwidth is available. This service can also be used
as a last resort to begin communication with besteffort or other low priority when high-quality communication cannot be established with a pre-arranged
guaranteed bandwidth service, allowing the users toincrease the quality later.
8. Current achievements and future potential of PPS
Concerning the observation time (a key PPS parameter), we learned that an interval of about one second is sufficient to reveal that the margin for the packet loss rate is 1% or less for a practical network configuration and communication in which the bandwidth does not change (e.g., VoIP without compression). For communication in which the bandwidth
does change, we are clarifying the quality that can beachieved according to the degree of change and thedegree of multiplexing. We have verified that mostcommercial routers can perform MF-PHB, which is
newly introduced here, if the queuing configuration is
manually optimized, something that is not usually
done. The terminal observation function makes use of
the RTCP function, which has already been established for VoIP. Regarding the selected monitoringby the MMS, we have shown that there are no technical difficulties in terminal-behavior monitoring bylaboratory trials. All the results to date indicate that
there are no technical obstacles and that PPS is a technology that has very high capabilities with respect to
NTT Technical Review

Special Feature

the functions and achievable quality required for theterminal and the network [4]. PPS employs packet
discarding (the simplest behavior for routers) to judge
quality and operates at the terminal where the service
is finally received. Therefore, PPS should find widespread use as a general admission control technology
for the large-scale and complex packet-based networks of the future.
References
[1]
[2]
[3]
[4]

K. Mase, “Scalable Admission Control Method for the Internet,”
IEICE Journal, Vol. 85, No. 9, pp. 655-661, 2002.
V. Elek, G. Karlsson, and R. Ronngren, “Admission control based on
end-to-end measurement,” INFOCOM2000, 2000.
N. Morita, “Framework of Priority Promotion Scheme,” IETF, draftmorita-tsvwg-pps-01.txt, Oct. 2003.
“An architectural framework for support of quality of service (QoS)
in packet networks,” ITU-T, Y.1291, 2004.

Shunsuke Mori
Engineer, Network Software Service Project,
NTT Network Service Systems Laboratories.
He received the B.E. and M.E. degrees in electrical and electronic engineering from Chuo University, Tokyo in 1998 and 2000, respectively.
He joined NTT-East in 2000, and was engaged inwork on system integrations and network integrations for customers as a systems engineer.
After transferring to NTT in 2002, he started
researching IP telephony and interactive multimedia service provisioning over carrier-grade IP
networks. He is a member of the Institute of Electronics, Information and Communication Engineers (IEICE) of Japan.

Yasuro Kawarasaki
Senior Research Engineer, Network Software
Service Project, NTT Network Service Systems
Laboratories.
He received the B.E. and M.E. degrees in computer science from the University of ElectroCommunications, Chofu, Tokyo in 1985 and1987, respectively. After joining the NTT Laboratories in 1987, he studied intelligent network
systems. In 2002, he started researching IP telephony and interactive multimedia service provisioning over carrier-grade IP networks. He is amember of IEICE.

Hideki Kataoka
Senior Research Engineer, Network Software
Service Project, NTT Network Service Systems
Laboratories.
He received the B.E. and M.E. degrees inadvanced applied electronics from Tokyo Institute of Technology, Tokyo in 1979 and 1981,
respectively. He joined Musashino ElectricalCommunication Laboratories of Nippon Telegraph and Telephone Public Corporation (nowNTT) in 1981, where he has been engaged in
R&D of broadband and high-speed switchingsystems, optical switches, ATM switches, and IP
networks. He is a member of IEICE.

Naotaka Morita
Senior Research Engineer, Supervisor, Network Software Service Project, NTT Network
Service Systems Laboratories.
He received the B.E. and M.E. degrees from
Nagoya University, Nagoya, Aichi, in 1985 and1987, respectively. Since joining NTT Laboratories in 1987, he has been engaged in work on
communication protocols and traffic management for ATM and B-ISDN. After a two-year
assignment in the strategic network planningdepartment at NTT Headquarters, he started
researching IP telephony and interactive multimedia service provisioning over carrier-grade IP
networks. He has been an active participant in
ITU-T since the early 1990s. Since 1997, he has
been a rapporteur of SG 13. He is a member of
IEICE.

Vol. 2 No. 10 Oct. 200427