“The audio keeps cutting out” is one of the most common complaints from users of network-based phones. The trouble is that this sentence can have ten different causes, from a faulty cable on one handset to an internet link that fills up whenever a large file is uploaded. Changing settings at random usually makes things worse. You need to work from the symptom to the cause.
Three Enemies of Voice: Latency, Jitter and Packet Loss
- Latency: the time it takes voice to travel from one end to the other. The common recommendation is to keep one-way latency below about 150 milliseconds; beyond that, people start talking over each other without meaning to.
- Jitter: packets arriving at irregular intervals. Phones compensate for some of it with a jitter buffer, but high jitter leads to choppy, robotic audio.
- Packet Loss: voice packets are not retransmitted, so even a small amount of loss is noticeable in the audio.
From Symptom to Cause
- One-way or no audio: almost always a NAT or firewall problem. SIP messages get through, but the voice stream (RTP) cannot find its way back. Check the public IP and local network settings in the phone system, make sure the RTP port range is open, and confirm SIP ALG is disabled.
- Call drops after about 30 seconds: usually means a SIP acknowledgment did not reach its destination; again, follow the NAT trail.
- Robotic or choppy audio: jitter or packet loss along the path, from a saturated internet link to weak Wi-Fi.
- Echo: if the other party hears their own voice, suspect the headset or speakerphone on your side, and for landline calls, the analog gateway settings.
- Problem on only one phone: check that phone’s cable, switch port, firmware or headset.
Many IP phones show jitter and packet loss statistics on their call status screen. If users take a photo of this screen when the problem occurs, troubleshooting becomes much faster.
Step-by-Step Troubleshooting
- Define the scope of the problem: all users or one person? Internal calls or only external ones? All the time or at certain hours? If internal calls are fine and external calls are not, look at the internet link or the trunk.
- Measure the path: use
ping, and preferablymtr, toward the provider to measure latency and packet loss at the time of the problem. - Check link usage: if the problem coincides with backups, file uploads or updates, the link is saturated.
- Check the signaling: on Asterisk-based servers, the
sngreptool shows the SIP message flow in a readable way and is very useful for finding NAT problems and dropped calls. - Analyze the voice stream: with Wireshark you can capture a call’s RTP packets and see jitter and packet loss precisely.
Lasting Solutions
- QoS at the network edge: mark voice traffic with a DSCP value of
EFand enable queuing on the router or firewall at the internet edge. Prioritization inside the local network is not enough. Shape outbound traffic slightly below the link’s real speed so that queue control stays with your own equipment, not the modem or the provider. - A separate VLAN for voice and healthy switch ports.
- Cable instead of Wi-Fi for phones and for computers running softphones.
- The right codec: a lower-bandwidth codec on limited links, and avoiding unnecessary transcoding in the phone system.
Common Mistakes
- Upgrading internet speed instead of managing traffic; a faster link still fills up with one heavy upload.
- Opening every port on the firewall to “fix” one-way audio.
- Troubleshooting without recording the time and numbers of problem calls.
Checklist
- The scope of the problem (user, call type, time) is defined.
- Latency, jitter and packet loss were measured at the time of the problem.
- SIP ALG is disabled and the phone system’s NAT settings are correct.
- QoS is enabled on the internet edge equipment.
- Phones are cabled and on the voice VLAN.
Call quality is not fixed by guesswork; it is fixed by measurement. ITPlus724 engineers can review your network and phone system and find the root cause of your call quality problems.