Sales Meeting - Draware A/S | Tlf. (+45) 45 76 20 21 | Vi

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Transcript Sales Meeting - Draware A/S | Tlf. (+45) 45 76 20 21 | Vi

Network Instruments VoIP Analysis

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

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

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

VoIP Terms

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

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

Observer’s Jitter Measurement

In aggregate… and per call…

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.

Observer’s Call Quality Scoring

In aggregate… per call… and Expert…

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.

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.

Observer’s Burst and Gap Density

In aggregate… and per call…

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.

Observer’s QoS/TOS/Precedence

In aggregate… per call… and Decode…

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

Observer’s Codecs Used Display

In aggregate… per call… and Decode…

Thank you