It is apparent that with the introduction of new technologies
such as Voice over IP and digital video, network managers and
administrators have a tough time keeping up with ever-increasing
bandwidth requirements. Such technologies are brought with historically
high expectations for reliability and quality. Today’s networks
must treat these services as high priority. These traditionally
“best effort” Local Area Network protocols (Ethernet
etc.) face a difficult time handling these High Priority requirements.
Quality of Service (QoS) promises better handling of these new
challenges; increasing reliability and quality.
Network administrators have two major types of QoS techniques
available. They can attempt to negotiate, reserve and hard-set
capacity for certain types of service (hard QoS), or just prioritize
data without reserving any “capacity setting” (soft
QoS). This paper will discuss both hard and soft QoS techniques
including 802.1P, IP Precedence, Differentiated Services, Resource
Reservation Protocol (RSVP) and ATM specific priority resources.
The paper will also explain how to implement QoS features on Transition
Networks’ Management Aggregation Converter.
QoS stands for Quality of Service. In QoS the bandwidth, error
rates and latency can be monitored, sampled and possibly improved.
QoS also delivers the set of tools to help deliver data efficiently
by reducing the impact of delay during peak times when networks
are approaching full capacity. QoS does not add capacity; nor
does it multiplex the signals like WDM. It simply tries to manage
data traffic better so that top priority traffic will not be compromised.
QoS helps manage the use of bandwidth by applying a set of tools
like priority scheme, so certain packets (mission critical –
must go packets) will be forwarded first.
QoS vs. Class of Service (CoS)
QoS is often used in conjunction with Class of Service. The shortest
definition of CoS would be “a grouping”. CoS defines
groups of traffic with a specific type of service, QoS manages
this type of service and assures that it is delivered. Similar
types of data such as Voice, Live Video, or streaming video and
large file transfer can be grouped together in a service class
and treated with a same level of service priority.
The need for QoS
Many users believe that more bandwidth will resolve the problem.
Throwing more bandwidth though may not work anymore. Voice over
IP Telephony and other new technologies such as a networked video
security, remote monitoring, and recording over IP networks are
becoming more popular. They have begun to penetrate traditionally
data orientated networks, forcing network administrators and managers
to employ measures such as QoS to accommodate these technologies
efficiently and without any backslash to the performance of an
Multiservice traffic is difficult to handle efficiently because
each type of traffic requires different transfer rate and and
each has a different tolerance for delay or packet sequencing.
The original best effort LAN protocols were designed for applications
such as basic connectivity between stations, file transfer, e-mail,
MRPs, and later on the Internet.
These applications are not compromised by packet
delay so as long as connection was established and transfer of
data happened in less then irritating manner, the network served
its purpose. Also, multiservice traffic has to peacefully coexist
with the infrastructure in place. In many instances VoIP has to
be routed back to the PSTN (see Figure 1) in order to complete
the necessary call or IP video has to be broadcast over the existing
close circuit TV (CCTV).
The network bandwidth is still important, but it
is no longer the only factor to consider for implementing future
technologies. The new specific characteristics of this traffic
(delay, jitter etc.) need to be read, understood, and implemented.
One of the keys in delivering voice or video over
any media is the maintenance of a level of quality. The quality
of voice or video may deteriorate as a result of three factors:
Compression ratios are inversely proportional to the quality of
a voice signal that is transmitted over the network, and is inferior
to what the user is accustomed with Plain Old Telephone (POTS).
The lower the compression the higher the throughput necessary
to transmit voice packets, increasing the possibility of network
congestion and consequently the loss of quality. Compression can
be easily controlled by users.
Packet loss on the network
Packets get lost on the network, which is not a problem for traditional
applications. The quality of traditional applications such as
file transfers is immune to packet loss because these losses are
recognized by the network and retransmitted. VoIP products reconstruct
the packets if the number is minimal. The rule of thumb is that
no more than 10% of packets should be lost in VOIP networks otherwise
the voice quality will be compromised.
Delay in data networks is not that critical. Waiting for a web
page to load is not as irritating as a silence in your receiver
when you are in the middle of an important conversation. A maximum
delay of 150ms is the rule of thumb for one-way latency to achieve
similar quality to POTS voice.
Network managers face a new challenge with voice
and security applications. Traditional POTS is highly reliable
in terms of transport and reliability. It is hard to imagine the
situation when you have no dial tone in our phone even during
the worst storm. While it was acceptable to wait 5 seconds loading
a web page, it is impossible to tolerate such a delay during a
conference call with the customer. It is impossible to accept
a voice breaking off or any noticeable latency.
Such expectations are being brought to the “opportunistic
– best effort” networks creating the need for QoS.
First in First Out (FIFO) systems so commonly used in opportunistic
networks have to be replaced by more sophisticated often dynamic
resource allocation tools starting with 802.1P and all the way
One important condition to be met in order for
the QoS to be successful is that it has to be employed and managed
end to end, across several LANs and WANs (see Figure 2). This
can guarantee all the bottlenecks are addressed and that voice/video
will not be distorted. If QoS is employed only on the portion
of the network, anything that has to go out of this network through
the “bottleneck” will be treated and forwarded in
the order it was received, at the available speed and with a possible
Protocols differ in their natural ability to properly
handle high volumes of traffic and some offer traditionally higher
“reliability”. ATM is a very successful protocol in
the multimedia applications because ATM can provide guaranteed
rates and connections that are so valuable to voice or video transmission.
ATM prioritizes the traffic by assigning it to one of four service
classes. Each class can receive a priority level. There are the
following four ATM service priorities/queues:
Constant Bit Rate (CBR) absolute guarantee of a service level
(VoIP or standard voice circuits).
Variable Bit Rate (VBR) for variable burstable transmission
rates with a very good throughput, but no guarantee as to the
consistency over time (FTP, streaming).
Available Bit Rate (ABR) offers a minimum guarantee.
Unspecified Bit Rate (UBR) which makes no guarantees, whatever
bandwidth is left can be used.
ATM can work along the priority settings done in the Ethernet
LAN. ATM though, due to lower LAN penetration will not be able
to solve all LAN QoS issues. This will have to be done by a “protocol
of choice for LANs” - Ethernet.
Ethernet represents more of an opportunistic protocol. Ethernet
is a connection less broadcast protocol and is “Best Effort
- As soon as I can”. Ethernet was designed to be less complex
and hence less expensive. When data is transmitted Ethernet allocates
the maximum possible bandwidth to this transfer until the network
runs out of its bandwidth. Consequently the “critical traffic”
is treated as any other transfer, so it pretty much drowns in
the sea of less critical/significant data. This means that it
will do just fine with voice and video in the time where there
is no congestion.
Networks are seldom designed for the worst case scenario (max
overload) so QoS helps effectively manage what we have at our
disposal without magically adding bandwidth.
Today’s enterprise networks are experiencing increased
complexity and packet equality becomes a song of the past. Below
(Table 1) we define different types of traffic, their bandwidth
requirements, and delay tolerance for each of them. The tolerance
score will explain how tolerant users are towards each service
when things do not go as smoothly as we would want them to go.
Table 1: Performance Requirements by Media Type
Relative Delay Tolerance
128 - 1mbps
Data transfer (e-mail, fserv etc.)
128 - 1mbps
Customer Chat (text)
Voice & Video (MPEG-2)
Medium / Low
256Kbps– 1 Mbps
256Kbps– 1 Mbps
Our tolerance of delays dictates what kind of time factor to
implement. Users’ relatively high tolerance towards delay
in such services as internet browsing, web hosting, data transfer
or fax enables network designers to allow for transport buffering
of such services. Real-time applications require establishing
benchmarks for service.
Figure 3: Possible Bottlenecks in VoIP Implementation
Clearly the bandwidth requirements mentioned above do not produce
any hiccups within a small enterprise operation. 100Mbps Fast
Ethernet can support these services, but what happens in bigger
organizations? What about end-to-end, when data to go “outside”
the enterprise LAN? Again, one of the key technical issues with
QoS is that it must be supported end-to-end to be effective (see
Figure 2). IP telephony and video conferencing, unlike basic Internet
surfing, must have a minimum transfer rate guaranteed so that
they can function properly.
A 100Mbps LAN connection cannot guarantee a voice connection
with another LAN over 128Kbps WAN connection. Due to the nature
and requirements of this communication, the connection has to
be continuous and there is no room for voice buffering.
So in times of congestion - what can be done to ensure those
critical pieces are flowing and the delay is minimal?
You need to first define what kind of traffic causes the bottleneck,
and where the bottleneck is located. This may very well identify
one of the following causes for congestion:
Too many packets are being sent over the network by users. A closer
look at large traffic patterns can sometimes quickly identify
the cause if it is deemed unnecessary. Necessary traffic can also
cause congestion because the existing network cannot provide sufficient
switching and routing
capability. Segmenting the network can help.
If your network is still congested and you cannot throw more
bandwidth at it, you can apply the set of the following tools:
Reserve/Limit the bandwidth for certain Users
Reserve/Limit the bandwidth for certain Applications
Select the applications that can be stopped
Having defined who and what gets priority, network administrators
have a set of tools to implement these rules.. For instance, they
can give priority to certain users based on their IP address (source
address). Or they might prioritize by segment either through subnet
mask or destination address. Prioritizing application means that
all Voice over IP services get a higher priority than let’s
Devices read the instructions as to the priority or bandwidth
allocation and queue packets in the following four types of queues:
Priority queuing from high priority queue to low priority.
Packets are sent from the queues of higher priority first (as
explained in the IEEE 802.1P).
Weighted fair queuing. It allows for guaranteed bandwidth
services, but over the same, shared link.
Class based Queuing divides user traffic into classes. These
classes are assigned based on IP addresses, protocols and application
The IEEE 802.1P is a signaling technique for prioritizing network
traffic at the data-link/MAC sublayer (OSI Reference Model Layer
2). The 802.1P header includes a three-bit field for prioritization,
which allows packets to be grouped into various traffic classes.
The IEEE 802.1P compliant switches pick up on this tag (the packet
contains a 32-bit tag header located after a destination and source
address header), read it, and put the packet in the appropriate
priority queue. No bandwidth is reserved nor requested by this
There are eight levels (0-7) of priority and consequently
eight queues that could be created (see Figure 4). Level Seven
represents the highest priority. This will be assigned for mission-critical
applications. Level 6 & 5 is designed for delay-sensitive
applications such as interactive video and voice. Levels four
and below, are suitable for regular enterprise data transfer,
as well as streaming video. Level zero is assigned for a traffic
that can tolerate all the drawbacks of a best-effort protocol.
The switch will analyze the packet based in the
“P” tag and will place it in the appropriate priority
Queue for sending. The user can have as many as eight priority
queues. An adjustable algorithm is employed to choose how many
packets are being sent from each queue before the packets in the
lower priority queue are sent.
Transition Networks Management Aggregation Converter
(MAC) is a converter that allows the remote end of the network
to be managed. One of the MAC’s many features includes supporting
802.1P packets. The MAC reads the 802.1P tag and places incoming
packets in either a High Priority Queue or a Low Priority Queue.The
network manager defines the priority level threshold (0-7) that
determines if a packet is placed in the high priority queue or
the low priority queue. . For example if the threshold is set
to 4, a “P” tag of 5 will be forwarded to the High
Priority Queue while a packet with a Tag of “3” will
be placed in Low priority queue. The MAC converter also implements
a user adjustable algorithm for packet queue selection. As shown
in Figure 5, 15 packets from High priority queue will be sent
and then one packet from the Low priority queue will be sent before
MAC converter comes back to high priority queue again.
In addition to queuing, MAC converters will also
enable users to disable/enable Pause in higher priority applications
so that real-time traffic (Voice) will not be paused in times
of congestion. (see Figure 7)
All converter management can be performed by a fully SNMP compliant
Graphical User interface (GUI) Software - Focal Point™ or
it can also be managed via the web based management using any
QoS 802.1P is an efficient tool for prioritization within a
LAN. QoS can also be accompanied by IP precedence or Differentiated
Services - Layer 3 QoS mechanisms to achieve inter LAN prioritizing.
The IP protocol includes the Type of Service (ToS) an 8-bit field,
intended for use in packet prioritization. It allocates three
of the ToS bits to create up to 8 priority levels and three bits
to describe delay sensitivity, as well as packet loss. Transition
Networks’ MAC converter is transparent to these packets.
Another very popular method of QoS in the enterprise is Differentiated
Services. It is an efficient method of managing traffic based
on its class. Differentiated Services (Diffserv) prioritizes certain
types of traffic like voice traffic over other types of communications.
It works by categorizing IP packets into classes. The six bits
in the type-of-service byte contained in the IP header of each
packet, specifies a particular behavior type which determines
the packet-forwarding scheme and priority.
Differentiated services can offer the following:
Expedited Forwarding (EF), which defines minimum delay and
jitter. Preferred mode for the VoIP.
Assured Forwarding (AF), which introduces three selectable
packet drop rates. During congestion, packets with a high drop
precedence are discarded. Thus enabling the more important traffic
marked with lower drop precedence to get through.
Best effort picks up the remains of the bandwidth not allocated
to EF and AF.
DiffServ can be used as a QoS mechanism in enterprise networks.
It is scalable. Almost all new router products as well as end-products
such as VoIP phones support DiffServ and can tag the packets with
the appropriate per-hop behavior type. Differentiated Services
marking at the edge is read and understood at the core and the
packets are forwarded based on the above mentioned priority schemes.
Transition Networks’ Mac Converter passes these packets
Such QoS services are not part of any negotiation or signaling
between devices themselves. These rules are assigned by local
network administrators who understand the above mentioned reasons
for congestion and adjust priorities for users, applications or
services accordingly. These assigned tags are passed in the packet
and are NOT subject to change during the process of auto-negotiation
or other forms of signaling. Such approach is called SOFT QoS.
802.1P, IP Precedence and DiffServ are the examples of soft QoS
Hard QoS describes the process during which the devices on the
network through signaling can negotiate, request and adjust priority
levels for different types of traffic based on the previously
Hard QoS includes protocols such as Integrated Services/Resource
Reservation Protocol .
Integrated Services/ Resource Reservation Protocol (RSVP)
RSVP enables network devices such as routers or switches to request
the necessary /guaranteed bandwidth from other devices on the
network for a particular traffic type (e.g. VoIP). Desired delay
variances can also be defined in this approach. The RSVP sends
a request to reserve specific bandwidth or switching/forwarding
capability from other devices on the network. This requirement
sent over the network is called flow specification. The requirements
can result in three desired transfer types:
Rate-sensitive - VoIP requires a guaranteed bit-rate service
established bandwidth for video streaming applications.
Delay-sensitive - VoIP requires max delay to be defined and
this maximum not allowed to be exceeded.
Voice, audio and video traffic put increasing pressure on both
LAN and WAN networks. Users are accustomed to the high reliability
and high quality of standard voice and video technologies. Although
the transport medium is changing our expectations remain the same.
VoIP will not be as fortunate as mobile technology where users
sacrifice high quality for the convenience of having a phone on
the road. LANs and their, in most cases, opportunistic/best effort
protocols face a difficult time handling these high expectations
On the other hand, the IP video and phone implementation projects
will happen at the increased pace as companies are already starting
to fully embrace the cost savings generated by these technologies.
This accelerating process will not be matched by an adequate growth
in network capacity. QoS is an attractive alternative to aimlessly
adding bandwidth to the network.
When voice data becomes part of a network the priority has to
be given to the voice packets to “meet” expected high
quality of voice calls. ATM, has designed ability for successful
QoS, yet since Ethernet runs on 85% of LANs QoS has to efficiently
run on this platform as well. 802.1P, IP Precedence, and DiffServ
– (soft QoS techniques) help administrators prioritize different
types of traffic without any resource reservations. RSVP a Hard
QoS technique will help reserve a required level of capacity to
support QoS effort. None of these techniques are failure-proof.
QoS has to be planned from end-to-end so the bottlenecks are identified
Finally, QoS will not do magic, and it will not relieve the
responsibility of the network managers to plan, and allocate resources
accordingly. But the various elements that comprise QoS can offer
powerful tools to enable network managers to improve network performance.