Professional Documents
Culture Documents
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
• 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
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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
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Delay, Delay Variation (Jitter)
& Its Characteristics
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Delay Components
IP network delay
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Delay Components (Cont.)
Other delays
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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
14 frame-based, the codec delay is only 2 sampling intervals
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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