Audio Troubleshooting

Audio issues in a VoIP system can in general be divided into 3 main points:

End User Device
Network
Codec - wrong codec often results in no audio or one-way audio

In general it can be a good idea to do a network capture and inspect the data to get an idea on where the issue lies:

Zenitel End User Device

Make sure the device is running the latest version

It is recommended to perform a reset to factory values on the audio settings. This should be done on all devices in the system, and especially after an upgrade from below version 7.5.

Factory Reset Audio Settings

Other things to check:

Does the device show the correct model in the web GUI? (Main -> Information, on the top)
Is the correct audio input configured? (Microphone type, line in etc)
For custom device using kit, maybe Handset or Headset settings needs to be used?

One way audio

Can be caused by wrong codec settings. Zenitel devices generally prefer G722.
Firewall blocking UDP ports
Wrong audio input settings. Some devices has an option for choosing between MEMS microphone and external electret microphone. Make sure the right one is selected.
Improperly connected microphone or speaker
Defective or damaged microphone or speaker

No audio

Can be caused by wrong codec settings. Zenitel devices generally prefer G722.
Firewall blocking UDP ports
Improperly connected microphone or speaker
Defective or damaged microphone or speaker

Feedback

Feedback occurs as an unwanted echoing or howling sound that happens when audio from a speaker is picked up by a microphone and sent back into the call.

Lower speaker volume
Reduce microphone gain
Increase AEC (automatic echo cancellation). Please note this can affect audio quality.
Physically move one (or both) devices further away from each other
If experienced during group call:
- Use Recall instead
- Use the Group Exclusion feature

Choppy Audio

There are a number of things that can cause choppy audio. Most are due to network efficiency.

QoS

Not supporting and prioritizing VoIP traffic in the network can cause packet loss, that in turn will be experienced as choppy and bad quality audio. Enabling and correctly configuring QoS (Quality of Service) ensuring enough bandwidth will help avoid packet loss and jitter issues.

In Zenitel devices the audio packets are marked with the DiffServ EF class to enable Quality of Service (QoS) prioritization.

Jitterbuffer

A jitter buffer will save up packets of audio in order to stack them in the correct sequence before passing it on to the receiver. The jitter buffer can then compensate for audio packets arriving late and out of sequence.

Zenitel uses dynamic jitter:

50ms on normal latency and 10ms on low latency''

Packet loss

Packet loss will heavily affect call handling and call quality. This can be caused by:

Network congestion - occurs if a network is overloading and reaching maximum capacity. Often due to poorly configured networks and lack of QoS
Network hardware - outdated network hardware can malfunction and cause unnecessary delays in the network causing packet loss. Make sure your switches, routers, modems and firewalls are up do date to support your needs.
Software bugs - software is in the network every step of the way. Software bugs can cause treatment of VoIP audio and data to malfunction, this leading to poor audio quality. Fixed with a software update of network equipment.
Security breech - hackers or viruses can penetrate your network and launch DOS (denial of service) attacks or similar. This can cause the network to be flooded thus creating heavy packet losses for you VoIP system, which will lead to bad audio quality or trouble with call setup. This is often avoided by making sure your network is updated and also hardened for cyber security.

Double talk

Latency

Too mush delay in the network (from user A to user B) will cause people to start talking at the same time. Roundtrip latency should not exceed 300ms.

Codec

For ICX-AlphaCom the best codec to use is G.722.

If using 3rd party devices:

make sure that they support G722
A SIP user in AlphaPro should be defined with G722 codec