Skip to main content

Voice Quality

What can I do about voice quality issues (dropouts, distortions, echoes, no sound)?​

The quality of VoIP calls can be negatively affected by various factors.
This can range from short interruptions to distortions, echoes, or even a complete loss of audio transmission.

Dropouts​

Dropouts in audio transmission can have several causes:

Packet Loss​

If a network component is malfunctioning or the internet connection is unstable, packet loss may occur.
This issue is most noticeable in VoIP calls.
For example, on a 1 Gbps internet connection with a 10% packet loss issue, a speed test might still show a throughput of 900 Mbps.
This would allow normal web browsing but would make VoIP calls difficult, as a 10% packet loss in an audio stream can make speech unintelligible.

Wi-Fi​

Since Wi-Fi is a shared medium, only one device can transmit at a time while all others must wait.
Each client is assigned a time slot for transmission.
If too many clients and neighboring Wi-Fi networks are in range, the time slots for each client become too short,
causing audio transmission issues.

To check if this is the cause, try using the sipcall app on a smartphone.
Compare call quality with Wi-Fi enabled and Wi-Fi disabled (using mobile data instead).
If possible, we recommend using mobile data, as it is optimized for audio transmission.

High Latency​

If a network component cannot process incoming packets immediately, high latency occurs.
This negatively impacts call quality because delayed audio packets are discarded.
Possible causes include:

  • A firewall's NAT performance not being sufficient for the internet connection speed.
  • A firewall being overloaded with other services like VPN traffic.

Distortions​

Distorted (metallic) voices are usually caused by overly compressed audio codecs.

For our recommended codecs, see the codec overview.

Echoes​

Echoes can usually be eliminated by enabling "Echo Cancellation" in most devices.
If you are using a softphone, be aware of physical feedback between the microphone and speakers.
We recommend using a USB headset for the best experience.

No Sound​

Firewall Blocking RTP Transmission​

If the audio stream from the other party does not arrive, a firewall may be blocking the transmission.

For all necessary firewall settings, refer to our firewall configuration guide.

SIP-ALG​

Many firewalls and routers have a SIP-ALG function enabled by default.
This was designed to help SIP devices work behind NAT.
However, SIP-ALG often causes issues, such as misrouting audio streams.

Please disable SIP-ALG in your network equipment.

Last updated on