Glossary / Quality Metrics and Troubleshooting
🔍 Quality Metrics and Troubleshooting covers the scores, symptoms, and diagnostic tools used to measure and fix VoIP call quality. This section contains 16 terms, from industry-standard quality scores like MOS to common problems like one-way audio and the tools used to investigate them.
On this page: MOS · R-Factor/R-Value · E-Model · One-Way Audio · No Audio · Choppy/Robotic Audio · Echo · Call Drops · Registration Failures · SIP Trace/SIP Capture · Wireshark · sngrep · Homer · CDR · RTP Timeout · SRTP Authentication Failure · Jitter Buffer Statistics
MOS (Mean Opinion Score)
A numerical score from 1.0 to 5.0 that rates the perceived quality of a voice call. A score of 4.0 or above is considered good quality, similar to a traditional landline. Scores below 3.5 indicate noticeable problems such as distortion, delay, or clipping. MOS was originally based on human listeners rating calls, but modern systems calculate it automatically using algorithms that analyse packet loss, jitter, and latency.
Related: R-Factor/R-Value · E-Model · Jitter
R-Factor / R-Value
A score from 0 to 100 that measures voice quality based on network conditions. An R-Factor above 80 is considered good; below 60 is poor. It accounts for delay, packet loss, jitter, and the codec being used. The R-Factor can be converted directly to a MOS score. Network engineers use it because it is easier to work with than MOS for diagnosing specific network problems.
Related: MOS · E-Model
E-Model
An ITU-T standard (G.107) used to predict voice call quality by combining many factors into a single R-Factor score. The E-Model considers the codec, packet loss rate, delay, jitter, and equipment quality. It is the mathematical framework behind most automated MOS and R-Factor measurements. You will rarely interact with the E-Model directly, but it powers the quality scores your Cloud PBX dashboard shows.
Related: MOS · R-Factor/R-Value
One-Way Audio
A common VoIP problem where one person can hear the other, but not the reverse. Typically caused by a firewall or NAT (Network Address Translation) device blocking the return audio stream (RTP packets). The call connects and signalling works fine, but the media path is broken in one direction. Fixing it usually involves opening the correct ports or enabling the SIP ALG on your router.
No Audio
A VoIP problem where the call connects (you see a timer running) but neither party can hear anything. This is typically caused by a firewall blocking RTP traffic in both directions, incorrect RTP port settings, or a SRTP mismatch (one side expects encrypted audio, the other sends unencrypted). Checking firewall rules and verifying that both endpoints use the same media settings usually resolves it.
Related: One-Way Audio · SRTP Authentication Failure · RTP
Choppy / Robotic Audio
A voice quality issue where the speech sounds broken, metallic, or robotic. It is caused by packet loss (missing audio data), excessive jitter (packets arriving out of order), or insufficient bandwidth. Small amounts of packet loss (1-2%) can already make speech hard to understand. The fix involves improving the network: use QoS settings to prioritise voice traffic, upgrade bandwidth, or switch to a wired connection.
Echo (Troubleshooting)
Hearing your own voice repeated back to you during a call, usually with a slight delay. In VoIP, echo is most often caused by acoustic feedback (speaker sound leaking into the microphone) or impedance mismatches at the point where VoIP meets a traditional phone line. Modern phones and PBX systems include echo cancellation algorithms. If echo persists, check the volume levels and headset quality.
Related: Codec · MOS
Call Drops
Calls that disconnect unexpectedly during a conversation. Common causes include unstable internet connections, SIP session timeouts, NAT binding expiry (the router forgets the connection), or server-side issues. Frequent call drops may also indicate that SIP keep-alive messages are not configured, so the firewall closes the connection after a period of silence.
Related: NAT · Registration Failures · RTP Timeout
Registration Failures
When a SIP phone or PBX cannot register with the server, meaning it cannot make or receive calls. The phone will typically show "Not Registered" or a similar error. Causes include wrong credentials, incorrect server address, blocked SIP ports, expired TLS certificates, or network problems. Check the SIP registration status in your phone's settings as the first troubleshooting step.
SIP Trace / SIP Capture
A diagnostic recording of all SIP messages exchanged between devices. A SIP trace shows every INVITE, 200 OK, BYE, and error response in sequence, along with timestamps and IP addresses. It is the most important tool for diagnosing call setup failures, audio issues, and registration problems. Most PBX systems have a built-in SIP trace feature.
Related: Wireshark · sngrep · Homer
Wireshark
A free, open-source network analysis tool used to capture and inspect all network traffic on a computer or server. For VoIP troubleshooting, Wireshark can decode SIP messages, display RTP audio streams, calculate MOS scores, and generate call flow diagrams. It is the most widely used tool for deep VoIP diagnostics. It runs on Windows, macOS, and Linux.
Related: SIP Trace/SIP Capture · RTP
sngrep
A command-line tool designed specifically for viewing SIP traffic in real time. It displays SIP call flows as visual "ladder diagrams" directly in the terminal. It is lighter and faster than Wireshark for quick SIP debugging on Linux servers. Network administrators use sngrep when they need to check SIP messages without a graphical interface.
Related: SIP Trace/SIP Capture · Wireshark
Homer
An open-source VoIP monitoring and troubleshooting platform. Homer captures SIP messages and stores them in a searchable database. It provides a web interface for searching call records, viewing call flow diagrams, and analysing quality statistics over time. It is designed for large-scale VoIP environments where thousands of calls happen every day.
Related: SIP Trace/SIP Capture · CDR
CDR (Call Detail Record)
A log entry that records the details of every call: who called, who was called, the start time, duration, whether it was answered, and often a quality score. CDRs are generated by the PBX or SIP provider for every call. They are used for billing, troubleshooting, compliance, and usage reporting. Your Cloud PBX dashboard typically lets you search and export CDRs.
RTP Timeout
A condition where the PBX or phone stops receiving RTP (audio) packets for a set period and automatically ends the call. This is a safety mechanism to clean up "ghost calls" where one side has disconnected but the other never received a BYE message. A typical RTP timeout is 30 to 60 seconds of silence. If RTP timeouts happen frequently, it points to network instability.
Related: RTP · Call Drops
SRTP Authentication Failure
An error that occurs when two devices try to set up encrypted audio but their SRTP settings do not match. For example, one device offers SRTP with SDES keys, but the other expects unencrypted RTP. The result is typically no audio. To fix it, ensure both endpoints use the same encryption method (SRTP on or off, same key exchange method).
Related: SRTP · No Audio
Jitter Buffer Statistics
Data reported by your phone or PBX about how the jitter buffer is performing. The jitter buffer temporarily stores incoming audio packets to smooth out timing differences. Key statistics include buffer size, packets discarded (too late to play), and buffer overruns (too many packets). High discard rates indicate network jitter problems. Adjusting the buffer size can improve audio quality.
Related: Jitter · Choppy/Robotic Audio · MOS
Related Sections
🎵 Audio, Media and Codecs — RTP, codecs, and audio media terms
🔒 Security — TLS, SRTP, and VoIP fraud prevention
🔗 SIP Protocol — SIP methods, responses, and headers
🚀 Deployment and Infrastructure — NAT, firewalls, failover, and network setup
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