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Measuring VoIP Call Quality with MOS

VoIP Quality Mean Opinion Score (MOS)

The Mean Opinion Score (MOS) has been a commonly-used metric to measure the overall voice call quality for decades. MOS score is a rating from 1 to 5 of the perceived quality of a voice call, 1 being the lowest score and 5 the highest for excellent quality. It has been standardized by the International Telecommunications Union ITU-T.

MOS was originally developed for traditional voice calls but has been adapted to Voice over IP (VoIP) in the ITU-T PESQ P.862. The standard defines how to calculate MOS score for VoIP calls based on multiple factors such as the specific codec used for the VoIP call. Each VoIP codec (ex: G.711, G.722, G.723.1, G.729) behaves differently. Some codecs such as G.711 are uncompressed for higher quality but use more bandwidth than compressed codecs such as the G.729.

The MOS score we measure is the G.711 codec, which is by far the most commonly used codec for VoIP calls. The maximum MOS in VoIP for a G.711 call is 4.4 (even though the standard sets 5 as highest).

MOS Score vs Call Quality

The following table lists the different qualities and the lower MOS limit for each of them. The limit values are from the ITU-T standards.

Performance Measurement Definitions

Mean Opinion Score (MOS).  MOS is a measure (score) of the audio fidelity, or clarity, of a voice call. It is a statistical measurement that predicts how the average user would perceive the clarity of each call. The VoIP MOS SLA provides that the Applicable Network performance will not drop below 3.8 where MOS is calculated using the standards-based E-model (ITU-T G.107).

Jitter.  Also known as delay variation, jitter is defined as the variation or difference in the end-to-end delay between received packets of an IP or packet stream. When certain packets of information arrive out of order and the conversation becomes jumbled. If jitter is creating a delay of more than 50ms, your call quality will degrade significantly, resulting in choppy voice or temporary glitches.    

Latency.  In the context of VoIP latency, all latency of concern is one-way latency.  One-way latency is measured by counting the total time it takes a packet to travel from its source to its destination.  For digital networking and packet-switched networks, the primary elements are the transmission media and intermediate switching node processing.  All medium from fiber optics to coaxial cables take some time to transmit a packet from a source to a destination. Transmission delays depend on packet size; smaller packets take less time to transmit the complete packet to the destination than larger packets.  Once a full or partial packet reaches a switching node it must be processed to be consumed or re-transmitted to the next destination.  Normally reflected in milliseconds from source to destination.  Generally, we consider scores below 70ms healthy, 70-120ms moderate and possibly impactful and anything over 120ms to be alarming.

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 either caused by errors in data transmission or network congestion whereby a packet is discarded by an intermediate switching node by queue limitation or queuing policy.  Packet loss is measured as a percentage of packets lost with respect to packets sent.  Packet loss is one possible contributor to one-way audio.


Poor Jitter, Latency or Packet Loss can be the biproduct of one or any combination of the following:

  • Firewall / gateway TCP or UDP port settings
  • Low or oversubscribed bandwidth
  • Non-prioritized VoIP packets from the gateway router
  • SIP ALG (Application Layer Gateway) activation
  • Public / private Internet services and dependencies
  • Generation of large SIP packet payloads from paging or all-call paging practices
  • VPN routing that may not be optimal for VoIP SIP traffic to the provider
  • Distance between a customer endpoint and the VoIP provider datacenter