Glossary

Packet loss

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.

VoIPmonitor 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 delay variation PDV

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.

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.

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.

VoIPmonitor Packet delay variation

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.

The main advantage over traditional standard jitter metric value is that you can search calls for specific delays characteristics.

Jitter buffer

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. 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.

MOS score

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.

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.

MOS tests for voice are specified by ITU-T recommendation P.800

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:

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

The MOS is the arithmetic mean of all the individual scores, and can range from 1 (worst) to 5 (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

Codec	Data rate [kbit/s]	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
GSM EFR	12.2	3.8
G.726 ADPCM	32	3.85
G.729a	8	3.7
GSM FR	12.2	3.5

VoIPmonitor MOS prediction

VoIPmonitor transforms PDV and Packet loss into MOS score according to ITU-T E‑model­ meanings the MOS does not represent audio signal but network parameters. Because the relation of PDV and MOS scores depends on jitterbuffer implementation voipmonitor implements three MOS scores.

• MOS F1 – fixed jitterbuffer simulator up to 50 ms buffer
• MOS F2 – fixed jitterbuffer simulator up to 200 ms buffer
• MOS adapt – adaptive jitterbuffer simulator up to 500ms buffer

VoIPmonitor assumes that the call uses a G711 codec with a maximum MOS score of 4.5. That's why calls do not have “right” subjective 4.1. The reason is that you can easily filter all calls for the same MOS score regardless of the codec used. If you want the actual MOS score for G.729 – there is option in the sniffer (check /etc/voipmonitor.conf). The MOS score should not be taken as a definitive value. Delay/loss distribution and other paratmeters must be checked as well. This value is just for quick filtering of potentially bad calls.

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.