Glossary: Difference between revisions

From VoIPmonitor.org
Jump to navigation Jump to search
No edit summary
No edit summary
 
(2 intermediate revisions by one other user not shown)
Line 1: Line 1:
[[Category:GUI manual]]
[[Category:GUI manual]]
{{DISPLAYTITLE:Glossary of VoIP & Monitoring Terms}}


{{DISPLAYTITLE:Desired Title:test}}
'''This glossary defines key terms and metrics related to VoIP quality and network performance, explaining how each is measured and utilized within the VoIPmonitor application.'''


=Packet loss=
== Core Network Quality Metrics ==
Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is one of the three main error types encountered in digital communications. Packet loss can be caused by signal degradation over the network medium due to multi-path fading, packet drop because of channel congestion, corrupted packets rejected in-transit, faulty networking hardware, faulty network drivers or normal routing routines.
These metrics describe the fundamental health and stability of the network path carrying the VoIP traffic.


==VoIPmonitor loss==
=== Packet Loss ===
VoIPmonitor detects packet loss and stores loss distribution to 10 loss intervals so it is able to find larger consecutive losses. This is important because between two calls with two percent package loss, one with random losses throughout will be heard much better than one with a string of consecutive losses.
Packet loss occurs when one or more data packets traveling across a network fail to reach their destination. It is a critical issue in VoIP, as lost audio packets can result in audible gaps, clicks, or dropouts in the conversation. Common causes include network congestion, faulty hardware, signal degradation, or misconfigured network devices.


=Packet delay variation PDV=
==== In the Context of VoIPmonitor ====
In computer networking, packet delay variation (PDV) is the difference in end-to-end one-way delay between selected packets in a flow with any lost packets being ignored. The effect is sometimes referred to as jitter and although not in electronics, usage of the term jitter may cause confusion. In this document jitter will always mean PDV.  
VoIPmonitor detects packet loss for each call direction (caller and callee) and stores a detailed distribution of these losses. Instead of just a single percentage, it records how many consecutive packets were lost in various intervals. This is crucial because a 2% loss spread randomly is far less noticeable than a single 2-second burst of 100% loss, even though the average percentage might be similar.


The delay is from the start of the packet being transmitted at the source to the end of the packet being received at the destination. A component of the delay which does not vary from packet to packet can be ignored, hence if the packet sizes are the same and packets always take the same time to be processed at the destination then the packet arrival time at the destination could be used instead of the time the end of the packet is received. For interactive real-time applications, e.g., VoIP, PDV can be a serious issue and hence VoIP transmissions may need Quality of Service-enabled networks to provide a high-quality channel.
=== Packet Delay Variation (PDV) or Jitter ===
In networking, Packet Delay Variation (PDV) is the measure of how much the arrival time of packets differs from their expected, consistent interval. For VoIP, this is commonly referred to as **jitter**. High jitter means packets are arriving in erratic, unpredictable bursts, which can severely degrade voice quality even if no packets are lost.


The effects of PDV in multimedia streams can be removed by a properly sized jitter buffer at the receiver, which may only cause a detectable delay before the start of media playback.
==== In the Context of VoIPmonitor ====
VoIPmonitor measures PDV by comparing the arrival time of each RTP packet against the expected interval (typically 20ms for most codecs). It records the number of packets that exceed certain delay thresholds. This detailed breakdown is more valuable than a single average jitter value, as it helps identify specific patterns of delay. The default PDV intervals measured are:
* 50–70ms
* 70–90ms
* 90–120ms
* 120–150ms
* 150–300ms
* >300ms


== VoIPmonitor Packet delay variation ==
=== Post-Dial Delay (PDD) ===
VoIPmonitor compares each RTP packet if the delay differs from the optimal value (for most cases the delay between two RTP packets are 20ms). If the delay is higher than 50ms it will be counted to one of PDV intervals which is stored for each RPT direction in cdr table. There are those PDV intervals: 50 – 70ms, 70 – 90ms, 90 – 120ms, 120 – 150ms, 150-200ms, > 300ms.  
Post-Dial Delay is the time a user waits from the moment they finish dialing the last digit to the moment they hear feedback, such as a ringback tone or a busy signal. Long PDD can create a poor user experience, as callers may think the call has failed and hang up prematurely.


The main advantage over traditional standard jitter metric value is that you can search calls for specific delays characteristics.
== Mechanisms to Handle Network Issues ==
These are techniques used by VoIP endpoints to mitigate the effects of poor network quality.


= Jitter buffer =
=== Jitter Buffer (or De-Jitter Buffer) ===
A jitter buffer is a temporary storage area at the receiving end of a VoIP call. It intentionally delays incoming audio packets slightly, allowing them to be reordered and played out in a smooth, continuous stream, effectively hiding the negative effects of network jitter (PDV). The size of the buffer determines the maximum amount of jitter it can compensate for. Some systems use **adaptive jitter buffers** that can dynamically change their size based on current network conditions.


Jitter buffers or de-jitter buffers are used to counter PDV (jitter) introduced by queuing in packet switched networks a continuous stream of audio (or video) is transmitted over the network The maximum jitter that can be countered by a de-jitter buffer is equal to the buffering delay introduced before starting the play-out of the mediastream. In the context of packet-switched networks, the term packet delay variation is often preferred over jitter.
=== Packet Loss Concealment (PLC) ===
Some systems use sophisticated delay-optimal de-jitter buffers that are capable of adapting the buffering delay to changing network jitter characteristics. These are known as adaptive de-jitter buffers and the adaptation logic is based on the jitter estimates calculated from the arrival characteristics of the media packets. Adaptive de-jittering involves introducing discontinuities in the media play-out, which may be irritating to the listener or viewer. Adaptive de-jittering is usually used for audio play-outs that feature a VAD/DTX encoded audio, which  allows the lengths of the silence periods to be adjusted, thus minimizing the perceptible impact of the adaptation.
Packet Loss Concealment (PLC) is a technique used to mask the effects of lost audio packets. Since retransmitting lost packets is not feasible in real-time voice conversations, the receiving device must intelligently "guess" what the missing audio sounded like.


=MOS score=
==== Common PLC Techniques ====
Mean opinion score (MOS) is a test that has been used for decades in telephonnetworks to obtain the human user's view of the quality of the network. Historically, and implied by the word Opinion in its name, MOS was a subjective measurement where listeners would sit in a "quiet room" and score call quality as they perceived it; per ITU-T recommendation P.800, "The talker should be seated in a quiet room with volume between 30 and 120 m3 and a reverberation time less than 500 ms (preferably in the range 200-300 ms). The room noise level must be below 30 dBA with no dominant peaks in the spectrum." Measuring Voice over IP (VoIP) is more objective, and is instead a calculation based on performance of the IP network over which it is carried. The calculation, which is defined in the ITU-T PESQ P.862 standard. Like most standards, the implementation is somewhat open to interpretation by the equipment or software manufacturer. Moreover, due to technological progress of phone manufacturers, a calculated MOS of 3.9 in a VoIP network may actually sound better than the formerly subjective score of > 4.0.
* '''Zero Insertion:''' Lost audio frames are simply replaced with silence. This is the crudest method.
* '''Waveform Substitution:''' The system repeats the last known audio frame to fill the gap. This is a common and effective technique, an example of which is defined in ITU recommendation G.711 Appendix I.
* '''Model-Based Methods:''' Advanced algorithms use speech models to interpolate or extrapolate the missing audio, providing a more natural-sounding result.


In multimedia (audio, voice telephony, or video) especially when codecs are used to compress the bandwidth requirement (for example, of a digitized voice connection from the standard 64 kilobit/second PCM modulation), the MOS provides a numerical indication of the perceived quality of received media from the users' perspective after compression and/or transmission. The MOS is expressed as a single number in the range 1 to 5, where 1 is lowest perceived audio quality, and 5 is the highest.  
== Voice Quality & Performance Measurement ==
These metrics are used to quantify the quality of a call or the performance of a network.


MOS tests for voice are specified by ITU-T recommendation P.800
=== Mean Opinion Score (MOS) ===
 
MOS is a standardized numerical rating of perceived voice quality, ranging from 1 (bad) to 5 (excellent). Originally a subjective test where human listeners would rate call quality, it is now typically calculated objectively using algorithms like the one defined in the ITU-T P.862 (PESQ) standard.
The MOS is generated by averaging the results of a set of standard, subjective tests where a number of listeners rate the audio quality of test sentences read aloud by both male and female speakers over the communications medium being tested. A listener is required to give each sentence a rating using the following rating scheme:


{| class="wikitable"
{| class="wikitable"
|+ MOS Ratings
|-
|-
! MOS !! Quality !! Impairment
! MOS !! Quality !! Impairment
Line 50: Line 64:
|}
|}


 
==== Codecs and MOS ====
The MOS is the arithmetic mean of all the individual scores, and can range from 1 (worst) to 5
Different audio codecs have different maximum theoretical MOS scores due to their compression methods.
(best).
 
Compressor/decompressor (codec) systems and digital signal processing (DSP) are commonly used in voice communications, and can be configured to conserve bandwidth, but there is a trade-off between voice quality and bandwidth conservation. The best codecs provide the most bandwidth conservation while producing the least degradation of voice quality. Bandwidth can be measured quantitatively, but voice quality requires human interpretation, although estimates of voice quality can be made by automatic test systems.
 
As an example, the following are mean opinion scores for one implementation of different codecs
 
{| class="wikitable"
{| class="wikitable"
|+ Example Codec MOS Scores
|-
|-
! Codec !! Data rate [kbit/s] !! MOS
! Codec !! Data rate [kbit/s] !! Typical MOS
|-
|-
| G.711 (ISDN) || 64 || 4.1
| G.711 (ISDN) || 64 || 4.1
Line 71: Line 80:
|-
|-
| G.723.1 r63 || 6.3 || 3.9
| G.723.1 r63 || 6.3 || 3.9
|-
| GSM EFR || 12.2 || 3.8
|-
|-
| G.726 ADPCM || 32 || 3.85
| G.726 ADPCM || 32 || 3.85
|-
| G.729a || 8 || 3.7
|-
| GSM FR || 12.2 || 3.5
|}
|}


== VoIPmonitor MOS prediction ==
==== VoIPmonitor's Parametric MOS ====
VoIPmonitor transforms [[Glossary#Packet_delay_variation_PDV|PDV]] and packet loss into MOS score according to ITU-T E‑model (please note that jitter is [[Glossary#Packet_delay_variation_PDV|PDV]])­. The voipmonitor MOS does not represent audio signal but network parameters. Because the relation between PDV and MOS score depends on jitterbuffer implementation voipmonitor implements three jitterbuffer simulators and thus 3 MOS scores:
VoIPmonitor calculates a **parametric MOS score** based on network transport quality (Packet Loss and PDV), not the audio signal itself. It simulates how a jitter buffer would perform and derives a MOS score from the simulated packet loss. It provides three MOS scores based on different jitter buffer models:
 
* '''MOS F1:''' Simulates a fixed 50ms jitter buffer. It is very sensitive to jitter.
*MOS F1 – fixed jitterbuffer simulator up to 50 ms buffer. Any PDV higher than 50ms will produce packet loss even though there is no packet loss in the stream.
* '''MOS F2:''' Simulates a fixed 200ms jitter buffer.
*MOS F2 – fixed jitterbuffer simulator up to 200 ms buffer. Any PDV higher than 200ms will produce packet loss.
* '''MOS adapt:''' Simulates an adaptive jitter buffer that can grow up to 500ms.
*MOS adapt – adaptive jitterbuffer simulator up to 500ms buffer. Any PDV higher than current buffer length which is changing adaptively will produce packet loss.
 
If a call is long enough and there are only a few packet loss / PDV problems the MOS score can be averaged to good values although user remembers that the call had problems. This can happen on calls >15 minutes. We plan in future to calculate MOS scores after 20 seconds intervals and remember the worst MOS score.
 
VoIPmonitor uses our own equations which is calculated based on packet loss simulation using PESQ subjective MOS score. We have simulated random packet loss between RTP sender and receiver on a scale from 0 - 20% using Markov model distribution. Degraded audio signal for every packet loss simulation is compared with original sound by the PESQ which produces MOS score. Resulting data is on following chart where there are three surfaces. Top surface is MOS score (which is on Z axe) for G.711 codec with [[Glossary#PLC|PLC]] implementation (asterisks internal [[Glossary#PLC|PLC]]). Middle surface is for G.729 with native [[Glossary#PLC|PLC]] and the bottom surface is for G.711 without [[Glossary#PLC|PLC]].
 
 
[[File:mos.png]]
 
VoIPmonitor sniffer uses the G.711 [[Glossary#PLC|PLC]] surface variant for all calls regardless on codec and this is the reason why the MOS score starts at 4.5 for every good call regardless on codec. This is our intention because parametric MOS score is designed in our application for searching for calls with bad packet loss / PDV combinations regardless on codec or for watching sudden changes in MOS scores across whole SIP trunks. Mixing G.729 and G.711 MOS scores would be difficult to know if 3.9 MOS score (which is the highest number for G.729) is bad because of G.729 calls or if 3.9 is bad due to packet loss /PDV drops in G.711 calls.
 
And how the MOS score is exactly calculated? Based on our simulation data we have created approximate function which transforms data based on Ppl and BurstR into MOS score. The function is hardcoded directly in the sniffer.
 
= Post Dial Delay (PDD) =
Post Dial Delay (PDD) is experienced by the customer originating the call from the time the final digit is dialled to the point at which they hear ring tone or other in-band information. Where the originating network is required to play an announcement before completing the call then this definition of PDD excludes the duration of such announcements.
 
=RTCP=
The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in the RTP specification RFC 3550 superseding its original standardization in 1996 (RFC 1889).RTCP provides out-of-band statistics and control information for an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not transport any media streams itself. Typically RTP will be sent on an even-numbered UDP port, with RTCP messages being sent over the next higher odd-numbered port. The primary function of RTCP is to provide feedback on the quality of service (QoS) in media distribution by periodically sending statistics information to participants in a streaming multimedia session.RTCP gathers statistics for a media connection and information such as transmitted octet and packet counts, lost packet counts, jitter, and round-trip delay time. An application may use this information to control quality of service parameters, perhaps by limiting flow, or using a different codec.VoIPmonitor (version >= 5) is able to parse and store RTCP statistics. For each call RTCP jitter, fraction loss and total loss is saved for each direction.
 
= ACD =
 
The average call duration is a measurement that reflects an average length of telephone calls.
 
= ASR =
 
Answer Seizure Ratio
 
ASR is a measure of network quality defined in ITU SG2 Recommendation E.411. Its calculated by taking the number of successfully answered calls and dividing by the total number of calls attempted (seizures). Since busy signals and other rejections by the called number count as call failures, the calculated ASR value can vary depending on user behavior.
 
= PLC =
 
Packet loss concealment (PLC) is a technique to mask the effects of packet loss in VoIP communications. Because the voice signal is sent as packets on a VoIP network, they may travel different routes to get to destination. At the receiver a packet might arrive very late, corrupted or simply might not arrive. One of the cases in which the last situation could happen is where a packet is rejected by a server which has a full buffer and cannot accept any more data. In a VoIP connection, error-control techniques such as ARQ are not feasible and the receiver should be able to cope with packet loss.
 
== PLC techniques ==
* '''Zero insertion''': the lost speech frames are replaced with zero
* '''Waveform substitution''': the missing gap is reconstructed by repeating a portion of already received speech. The simplest form of this would be to repeat the last received frame. Other techniques account for fundamental frequency, gap duration etc. Waveform substitution methods are popular because of their simplicity to understand and implement. An example of such an algorithm is proposed in ITU recommendation G.711 Appendix I.
* '''Model-based methods''': an increasing number of algorithms that take advantage of speech models of interpolating and extrapolating speech gaps are being introduced and developed.
 
= Percentiles =
 
Some values are expressed as %95 or %99 which is 95th percentile respectively 99th percentile. For example if MOS score %95 is 3 it tells that at least 5% of all calls have the MOS score 3 or worse. It is better to watch %95 or %99 than average or min/max values because average/min/max do not tell well that 5% of all calls are bad.
 
Example how the percentile is calculated for MOS score. Lets have 100 calls where only last 5 calls have MOS score 3.1, 2.5, 3.2, 1.0, 2.9.
 
* order all MOS calls by the best MOS score to the lowest (4.5, 4.5, ..., 3.2, 3.1, 2.9, 2.5, 1.0)
* remove first 95% of all calls (3.2, 3.1, 2.9, 2.5, 1.0)
* take the first number from the left of the remaining 5% which is 3.2
 
In this example the MOS score 95th percentile is 3.2. The average MOS score is 4.4, Min is 1.0 Max is 4.5. As you can see the average / min / max are not much useful but the %95 percentile tells that we have problem with 5% of all calls.
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 


[[File:mos.png|VoIPmonitor calculates MOS based on simulated packet loss and PDV, using a pre-calculated surface for G.711 with PLC.]]


By default, VoIPmonitor uses a calculation based on the G.711 codec with PLC for all calls. This provides a consistent baseline for comparing network quality across all calls, regardless of the actual codec used.


=== RTCP (RTP Control Protocol) ===
RTCP is a companion protocol to RTP that provides out-of-band statistics and control information for a media stream. It gathers metrics directly from the endpoints, such as transmitted packet counts, lost packet counts, jitter, and round-trip delay time. VoIPmonitor can parse these RTCP reports and store the statistics for each call, providing an alternative, endpoint-reported view of call quality.


=== Answer-Seizure Ratio (ASR) ===
ASR is a carrier-grade metric for network quality, defined by ITU E.411. It measures the percentage of successfully answered calls out of the total number of attempted calls (seizures). It is calculated as:
`(Total Answered Calls / Total Seizures) * 100`
A low ASR can indicate network problems, but it is also affected by user behavior (e.g., busy signals, unanswered calls).


=== Network Effectiveness Ratio (NER) ===
NER is a similar metric to ASR but is designed to measure only the network's ability to deliver a call, excluding user behavior. For NER, calls that reach the destination but are rejected (e.g., busy, no answer) are still counted as "successful" from the network's perspective. VoIPmonitor allows you to configure which SIP response codes are considered successful for NER calculation in the system settings.


=== Average Call Duration (ACD) ===
ACD is the average length of all answered telephone calls. A very low ACD, especially when combined with a low ASR, can indicate problems with call quality (e.g., users hanging up due to poor audio).


== Statistical Concepts ==


=== Percentiles ===
A percentile is a measure indicating the value below which a given percentage of observations in a group of observations falls. VoIPmonitor often uses the 95th or 99th percentile (shown as `%95` or `%99`) for metrics like MOS.


For example, a `MOS %95` score of 3.2 means that 5% of all calls had a MOS score of 3.2 or worse. This is a much more useful indicator of systemic problems than an average, which can be easily skewed by a large number of good calls.


.
== AI Summary for RAG ==
'''Summary:''' This document is a glossary of key performance indicators (KPIs) and technical concepts related to VoIP quality analysis. It defines terms such as Packet Loss, Packet Delay Variation (PDV), Jitter, and Post-Dial Delay (PDD). It also explains technologies designed to mitigate network issues, like Jitter Buffers and Packet Loss Concealment (PLC). A significant portion is dedicated to the Mean Opinion Score (MOS), detailing how it is calculated, the differences between subjective and parametric MOS, and how VoIPmonitor specifically simulates three types of MOS scores (F1, F2, adapt) based on network conditions. Finally, it defines carrier-level metrics like Answer-Seizure Ratio (ASR), Network Effectiveness Ratio (NER), and Average Call Duration (ACD), and explains statistical concepts like Percentiles.
'''Keywords:''' glossary, packet loss, pdv, jitter, jitter buffer, mos, mos score, pesq, pdd, rtcp, asr, ner, acd, plc, packet loss concealment, percentile, g.711, g.729, codec, quality, kpi, metric
'''Key Questions:'''
* What is the difference between Jitter and Packet Delay Variation (PDV)?
* How does VoIPmonitor calculate MOS? What are MOS F1, F2, and adapt?
* What is the difference between ASR and NER?
* What is Packet Loss Concealment (PLC)?
* How do I interpret a 95th percentile MOS score?
* What is a Jitter Buffer and what does it do?

Latest revision as of 09:48, 30 June 2025


This glossary defines key terms and metrics related to VoIP quality and network performance, explaining how each is measured and utilized within the VoIPmonitor application.

Core Network Quality Metrics

These metrics describe the fundamental health and stability of the network path carrying the VoIP traffic.

Packet Loss

Packet loss occurs when one or more data packets traveling across a network fail to reach their destination. It is a critical issue in VoIP, as lost audio packets can result in audible gaps, clicks, or dropouts in the conversation. Common causes include network congestion, faulty hardware, signal degradation, or misconfigured network devices.

In the Context of VoIPmonitor

VoIPmonitor detects packet loss for each call direction (caller and callee) and stores a detailed distribution of these losses. Instead of just a single percentage, it records how many consecutive packets were lost in various intervals. This is crucial because a 2% loss spread randomly is far less noticeable than a single 2-second burst of 100% loss, even though the average percentage might be similar.

Packet Delay Variation (PDV) or Jitter

In networking, Packet Delay Variation (PDV) is the measure of how much the arrival time of packets differs from their expected, consistent interval. For VoIP, this is commonly referred to as **jitter**. High jitter means packets are arriving in erratic, unpredictable bursts, which can severely degrade voice quality even if no packets are lost.

In the Context of VoIPmonitor

VoIPmonitor measures PDV by comparing the arrival time of each RTP packet against the expected interval (typically 20ms for most codecs). It records the number of packets that exceed certain delay thresholds. This detailed breakdown is more valuable than a single average jitter value, as it helps identify specific patterns of delay. The default PDV intervals measured are:

  • 50–70ms
  • 70–90ms
  • 90–120ms
  • 120–150ms
  • 150–300ms
  • >300ms

Post-Dial Delay (PDD)

Post-Dial Delay is the time a user waits from the moment they finish dialing the last digit to the moment they hear feedback, such as a ringback tone or a busy signal. Long PDD can create a poor user experience, as callers may think the call has failed and hang up prematurely.

Mechanisms to Handle Network Issues

These are techniques used by VoIP endpoints to mitigate the effects of poor network quality.

Jitter Buffer (or De-Jitter Buffer)

A jitter buffer is a temporary storage area at the receiving end of a VoIP call. It intentionally delays incoming audio packets slightly, allowing them to be reordered and played out in a smooth, continuous stream, effectively hiding the negative effects of network jitter (PDV). The size of the buffer determines the maximum amount of jitter it can compensate for. Some systems use **adaptive jitter buffers** that can dynamically change their size based on current network conditions.

Packet Loss Concealment (PLC)

Packet Loss Concealment (PLC) is a technique used to mask the effects of lost audio packets. Since retransmitting lost packets is not feasible in real-time voice conversations, the receiving device must intelligently "guess" what the missing audio sounded like.

Common PLC Techniques

  • Zero Insertion: Lost audio frames are simply replaced with silence. This is the crudest method.
  • Waveform Substitution: The system repeats the last known audio frame to fill the gap. This is a common and effective technique, an example of which is defined in ITU recommendation G.711 Appendix I.
  • Model-Based Methods: Advanced algorithms use speech models to interpolate or extrapolate the missing audio, providing a more natural-sounding result.

Voice Quality & Performance Measurement

These metrics are used to quantify the quality of a call or the performance of a network.

Mean Opinion Score (MOS)

MOS is a standardized numerical rating of perceived voice quality, ranging from 1 (bad) to 5 (excellent). Originally a subjective test where human listeners would rate call quality, it is now typically calculated objectively using algorithms like the one defined in the ITU-T P.862 (PESQ) standard.

MOS Ratings
MOS Quality Impairment
5 Excellent Imperceptible
4 Good Perceptible but not annoying
3 Fair Slightly annoying
2 Poor Annoying
1 Bad Very annoying

Codecs and MOS

Different audio codecs have different maximum theoretical MOS scores due to their compression methods.

Example Codec MOS Scores
Codec Data rate [kbit/s] Typical MOS
G.711 (ISDN) 64 4.1
iLBC 15.2 4.14
AMR 12.2 4.14
G.729 8 3.92
G.723.1 r63 6.3 3.9
G.726 ADPCM 32 3.85

VoIPmonitor's Parametric MOS

VoIPmonitor calculates a **parametric MOS score** based on network transport quality (Packet Loss and PDV), not the audio signal itself. It simulates how a jitter buffer would perform and derives a MOS score from the simulated packet loss. It provides three MOS scores based on different jitter buffer models:

  • MOS F1: Simulates a fixed 50ms jitter buffer. It is very sensitive to jitter.
  • MOS F2: Simulates a fixed 200ms jitter buffer.
  • MOS adapt: Simulates an adaptive jitter buffer that can grow up to 500ms.

VoIPmonitor calculates MOS based on simulated packet loss and PDV, using a pre-calculated surface for G.711 with PLC.

By default, VoIPmonitor uses a calculation based on the G.711 codec with PLC for all calls. This provides a consistent baseline for comparing network quality across all calls, regardless of the actual codec used.

RTCP (RTP Control Protocol)

RTCP is a companion protocol to RTP that provides out-of-band statistics and control information for a media stream. It gathers metrics directly from the endpoints, such as transmitted packet counts, lost packet counts, jitter, and round-trip delay time. VoIPmonitor can parse these RTCP reports and store the statistics for each call, providing an alternative, endpoint-reported view of call quality.

Answer-Seizure Ratio (ASR)

ASR is a carrier-grade metric for network quality, defined by ITU E.411. It measures the percentage of successfully answered calls out of the total number of attempted calls (seizures). It is calculated as: `(Total Answered Calls / Total Seizures) * 100` A low ASR can indicate network problems, but it is also affected by user behavior (e.g., busy signals, unanswered calls).

Network Effectiveness Ratio (NER)

NER is a similar metric to ASR but is designed to measure only the network's ability to deliver a call, excluding user behavior. For NER, calls that reach the destination but are rejected (e.g., busy, no answer) are still counted as "successful" from the network's perspective. VoIPmonitor allows you to configure which SIP response codes are considered successful for NER calculation in the system settings.

Average Call Duration (ACD)

ACD is the average length of all answered telephone calls. A very low ACD, especially when combined with a low ASR, can indicate problems with call quality (e.g., users hanging up due to poor audio).

Statistical Concepts

Percentiles

A percentile is a measure indicating the value below which a given percentage of observations in a group of observations falls. VoIPmonitor often uses the 95th or 99th percentile (shown as `%95` or `%99`) for metrics like MOS.

For example, a `MOS %95` score of 3.2 means that 5% of all calls had a MOS score of 3.2 or worse. This is a much more useful indicator of systemic problems than an average, which can be easily skewed by a large number of good calls.

AI Summary for RAG

Summary: This document is a glossary of key performance indicators (KPIs) and technical concepts related to VoIP quality analysis. It defines terms such as Packet Loss, Packet Delay Variation (PDV), Jitter, and Post-Dial Delay (PDD). It also explains technologies designed to mitigate network issues, like Jitter Buffers and Packet Loss Concealment (PLC). A significant portion is dedicated to the Mean Opinion Score (MOS), detailing how it is calculated, the differences between subjective and parametric MOS, and how VoIPmonitor specifically simulates three types of MOS scores (F1, F2, adapt) based on network conditions. Finally, it defines carrier-level metrics like Answer-Seizure Ratio (ASR), Network Effectiveness Ratio (NER), and Average Call Duration (ACD), and explains statistical concepts like Percentiles. Keywords: glossary, packet loss, pdv, jitter, jitter buffer, mos, mos score, pesq, pdd, rtcp, asr, ner, acd, plc, packet loss concealment, percentile, g.711, g.729, codec, quality, kpi, metric Key Questions:

  • What is the difference between Jitter and Packet Delay Variation (PDV)?
  • How does VoIPmonitor calculate MOS? What are MOS F1, F2, and adapt?
  • What is the difference between ASR and NER?
  • What is Packet Loss Concealment (PLC)?
  • How do I interpret a 95th percentile MOS score?
  • What is a Jitter Buffer and what does it do?
Retrieved from "https://www.voipmonitor.org//doc/index.php?title=Glossary&oldid=5329"