Network Instruments VoIP Analysis

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Network InstrumentsVoIP Analysis
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VoIP Basics

• What is VoIP?
• Packetized voice traffic sent over an IP network
• Competes with other traffic on the network
• A new technology that needs real-time, consistent
  monitoring
• Sensitive to delay
• Understanding VoIP Begins with Understanding
  Delay
• Normal (not sensitive to delay)example FTP,
  HTTP, e-mail, etc.
• Tolerant (sensitive to delay, loss tolerant,
  buffered by receiver) example streaming video,
  Internet radio, etc.
• Real-time (delay and loss sensitive) example
  VoIP

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Common VoIP Troubleshooting Myths

• Myth 1
• Running VoIP without Quality of Service is
  acceptable
• Myth 2
• No VoIP Site Survey is necessary
• Myth 3
• Voice conversations are secure

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VoIP Monitoring and Analysis Challenges

• Current, competing tools were designed for lab
  use
• No method of quickly determining status and
  health
• No mechanism for understanding aggregate call
  quality
• VoIP dependencies are not implemented properly
• Separate tools increase learning curve, reduce ROI

Observer
Other VoIP tools
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VoIP Terms

• Jitter
• R-Factor / MOS
• QoS / TOS / Precedence
• Burstiness / Gap / Gap Duration
• Compression Techniques (Codecs)

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Jitter

• What is it?
• Jitter is the variation in the time between
  packets transmitted and received.
• For example, if a packet stream leaves a device
  with 30 ms packet spacing and arrives with 50 ms
  packet spacing, the jitter is 20 ms.
• Adjusting jitter buffers can help at the expense
  of increased latency and thus, clipping. Jitter
  buffer overflow will introduce dropped packets.
• Why measure it?
• Understanding jitter gives you the hard facts to
  help improve call quality

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Observers Jitter Measurement
In aggregate
and per call
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Call Quality Scoring

• What is it?
• R-factor
• Identify call quality using a single source of
  visibility (actual live calls)
• Based on E-Model
• Scale 1-100, with the theoretical maximum being
  93.2 after typical degradation
• MOS
• Determine user satisfaction level with a call
• Takes into account a number of different factors
  (handset quality, ambient noise, network
  performance)
• Scale 1-5, with 4.0 and higher considered
  satisfied, 4.5 and higher is extremely satisfied
• When using a simulated call, traffic can be
  captured at the destination and compared to the
  original sent data to identify degradation.
• Why measure Call Quality Scoring?
• Provides objective and subjective scores to
  evaluate existing conditions and to compare with
  historical conditions.

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Observers Call Quality Scoring
In aggregate
per call
and Expert
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Burstiness and Burst Density

• What is it?
• A burst is a period of time characterized by high
  rates of packet loss.
• Burst Percentage is the of time bursts are
  occurring.
• Burst Density is the rate of VoIP data packets
  lost during a burst period.
• Why measure it?
• Higher rates affect call quality, especially when
  coupled with long Average Burst Duration times.
• Possible reason for packet loss include network
  congestion, media failure, and link failure.

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Gap Density and Duration

• What is it?
• Bursts are periods characterized by high rates of
  packet loss.
• Gaps are the periods between bursts.
• A gap is a period of time characterized by lower
  levels of packet loss than the burst periods
  that bound it.
• Gap Density is the percent rate of packet loss
  during the gaps.
• Average Gap Duration is measured in time.
• Why measure it?
• Knowing the gap helps define the burst.
• In most cases, packet loss during gaps is
  rendered insignificant by packet loss concealment
  techniques built into the VoIP infrastructure.

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Observers Burst and Gap Density
In aggregate
and per call
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Settings for QoS / Precedence

• Support for multiple definitions of Quality of
  Service (QoS)
• Also known as Precedence
• Also known as Type Of Service (TOS)
• What is it?
• QoS is a bit setting used by routers and switches
  to prioritize packet flow.
• Why measure it?
• Incorrectly set QoS can lead to contention of
  VoIP and other data on a network.
• Contention will lead to delays in packet
  delivery, and thus reduce call quality.

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Observers QoS/TOS/Precedence
In aggregate
per call
and Decode
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Compression Techniques

• Codec is a term for Coder/Decoder
• Different compression techniques (codecs)
• G.711 64kbps (no compression)
• G.729 8kbps
• G.723 6.3kbps, 5.3kbps
• Higher compression reduces R-Factor and MOS but
  also reduces potential contention

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Observers Codecs Used Display
In aggregate
per call
and Decode
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Thank you