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Agenda
• Concept of Quality of Service (QoS)
• What is Quality of Service (QoS)?
• QoS Metrics and Measurements
• Network Packet Loss and Its Characteristics
• Delay, Delay Variation (Jitter) and Its Characteristics

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What is Quality of Service
(QoS)?
• Quality of Service (QoS) is defined as
“the collective effect of service
performance, which determine the
degree of satisfaction of a user of the
service” in ITU-T Rec.

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What is Quality of Service
(QoS)?
QoS is generally referred to as
• Network Quality of Service (NQoS) with a
focus on quality of IP network performance

in contrast to

• End-to-end QoS which also includes the quality


of the terminal/end-device or segments/devices
related with Switched Communication Networks
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(SCN) such as PSDN, ISDN and GSM
Network QoS vs. End-to-End QoS

• In VoIP applications, Network QoS covers the quality for IP transmission segment
which starts from:
• When IP packets leave the terminal or end device to another VoIP terminal through
the IP network in a PC-to-PC call scenario
(e.g., a PC or a laptop running a voip software such as Skype)
• When IP packets leave a media gateway in a PSTN/IP combined network to another
5 media gateway in a phone-to-phone call scenario
Network QoS vs. End-to-End QoS

• In VoIP applications, the end-to-end QoS is also regarded as mouth-to-ear


quality
(quality of a VoIP call from a user speaking to a handset’s microphone at one end to another user listening on the phone
at the other end)

• This is mainly for one-way listening speech quality without consideration of


interactivity
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(for conversation quality)
QoS Metrics & Measurements

• Network QoS metrics generally include:


• packet loss
• delay
• jitter
(main impairments affecting voice quality in VoIP applications)

• These metrics can be measured or monitored by either:


• Active (intrusive) measurement
or
• Passive (nonintrusive) measurement

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Active (intrusive) network QoS
measurement

• Probe packets (e.g., ICMP ping packets) are sent into the network
and further compared with the echo packet to obtain the network
performance metrics such as:
• Packet loss percentage
• Maximum/average/minimum Round Trip Time (RTT)
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Passive (nonintrusive) network QoS
measurement

• There are no probe packets involved

• A measurement/monitoring (e.g.,Wireshark) is normally Trace data example


run to monitor network performance and its behavior by with sequence number and
delay
analyzing headers of IP packets of the interested
traffic (e.g., traffic in a VoIP session)

• From the trace data, packet loss, delay and delay


9 variation metrics can be calculated
Network Packet Loss and Its
Characteristics
• Two kinds of packet loss:

• Congestive Loss: It is caused by network congestions


at bottleneck links along the path due to router buffer
queuing overflow (This kind of loss is bursty in nature)

• Non-Congestive Loss is mainly due to lossy links such


as mobile/wireless networks and ADSL access networks
and is of random nature

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Delay, Delay Variation (Jitter)
& Its Characteristics

• IP network delay is the time a packet sent out


from the sender (Point A) to the time it
reaches the receiver (Point B)

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Delay Components
IP network delay

IP network delay mainly consists of the following components:


• Propagation delay: depends only on the physical distance of the
communications path and the communication medium
• Transmission delay: the sum of the time it takes the network interfaces in the
routers to send out the packet along the path
• Nodal Processing delay: the sum of the time it takes in the routers to decide
where (which interface) to send the packet based on packet header analysis and the
routing table
• Queuing delay: the time a packet has to spend in the queues of the routers
along the path. It is mainly caused by network congestion

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Delay Components (Cont.)
Other delays

If we consider the end-to-end delay, the following delay components


incurred at the sender and the receiver have to be taken into account:
• Codec delay
• Packetization delay
• Playout buffer delay

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Delay Components (Cont.)

• Codec delay: delay used by the encoder and decoder (codec) to encode the speech
samples into the speech bitstream and decode back into speech samples

• For modern hybrid codecs, speech compression is based on a speech frame (normally 10–
30 ms)
• Some codecs also need a look-ahead time (about a half or a quarter of a speech frame) to
complete the encoding process
• Codec processing also needs one frame length time

• Total codec delay is the sum of the speech frame length × 2, plus look-ahead time
Codec delay = 2×FrameSize+Look ahead

• For waveform-based codecs, such as PCM and ADPCM, encoding is sample-based, instead
of
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Codec Algorithmic Delay

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Delay Components (Cont.)

• Packetization delay: time needed to build data packets at the sender, as well
as to strip off packet headers at the receiver
For example, for AMR codec, if one packet contains two speech frames, then the
packetization delay equals: 2×20 = 40 ms

• Playout buffer delay: time waited at the playout buffer at the receiver side
The end-to-end delay can be expressed as:

• For VoIP applications, if a codec, packetization size and jitter buffer are fixed,
then end-to-end delay is mainly affected by network delay
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Delay Variation (Jitter) and Its
Characteristics
• In VoIP applications, packets are sent out at a fixed time interval at the sender side

As packets may traverse along different route/path which may incur different IP network
delay, this will cause packets arriving at the receiver side with different packet interval

The variation of delay is called jitter. The playout buffer at the receiver side is
used to alleviate the impact of jitter and to guarantee a smooth playout of
audio

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Delay Variation (Jitter) and Its
Characteristics
• If a packet arrives too late (see the packet i + 2 as an example), the packet will be dropped
out by the playout buffer (this is called late arrival packet loss in contrast to network packet
loss which is lost in the network)

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Thank you!

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Questions

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