Network Instruments VoIP Analysis. VoIP Basics  What is VoIP?  Packetized voice traffic sent over an IP network  Competes with other traffic on the.

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Presentation transcript:

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, , 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 Other VoIP tools Observer

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