Overview
Common types of call quality and audio transmission issues involve one-way audio transmission, incomprehensible audio, and temporary loss of voice signal (aka choppy audio). This article provides information on troubleshooting these issues.
Information
One-Way Audio
The one-way audio issues are typically related to the signaling parameters that are negotiated by the Session Initiation Protocol (SIP). Another possible reason for these issues is the misconfiguration of the firewall.
If the phone system uses a private IP address, confirm that you have correctly configured the system for Network Address Translation (NAT) support. Follow the steps below to check the NAT settings.
- Click on the gear icon.
- Click Network.
- Check the NAT enabled (Kerio Operator is behind a firewall) option.
- Check the Automatically update public IP address option.
- Note down the Audio port range (UDP).
You may need to permit this traffic through the firewall. - If you made any changes, click Apply.
Another NAT setting is configured in the properties of each extension.
- Click on the gear icon.
- Click Extensions.
- Double click an extension.
- Go to the Advanced tab.
- Check the Extension is behind NAT option.
- Click OK.
Many SIP-based applications and some phone devices attempt to identify NAT connectivity using technologies such as Session Traversal Utilities for NAT, or Interactive Connectivity Establishment. If the application or phone is located on the same network as the phone system, these technologies may require further NAT settings. To confirm whether the IP address of a device is accurately identified, refer to the Extensions.
- Locate the extension that may have one-way audio issues, and refer to the registration column.
- If the registered device is located on the same network, confirm that the registration IP address matches the device’s network IP address.
- If the registered device is located on a remote network, confirm that the registration IP address matches the Internet IP address associated with that device.
- If the registration IP address does not match the Internet IP address, refer to the NAT settings previously mentioned.
Incomprehensible Audio
Users may describe the sound as underwater, in a helicopter, or robotic. This type of issue is typically related to the codec or to the encoding of the voice signal. During the setup of a phone call, the SIP protocol determines an appropriate codec. In some situations, the negotiated codec may not be suitable for all devices. To troubleshoot this type of issue, review the call history that is located in the status area of the administration.
- Click on the status icon.
- Click Call History.
- Check the From Codec and To Codec columns.
If these columns are not visible, you can enable them by hovering the cursor above any column and clicking the arrow to open the menu.
Based on the feedback from your callers, you may be able to identify specific codecs that are associated with the incomprehensible calls. In this case, you may consider disabling these codecs. This is configured in the properties of a SIP interface.
- Click on the gear icon.
- Click Call Routing.
- To open the Edit External Interface (SIP) screen, double-click on the SIP interface.
- Select the Codecs tab.
- Under the Selected codecs list, select any codec that may be associated with the poor call quality and click Remove.
- Click OK.
Alternatively, you can disable codecs for specific extensions.
- Click on the gear icon.
- Click Extensions.
- To open the Edit Extensions window, double-click on the extension.
- Go to the Codecs tab.
- Under the Selected codecs list, select the codec that you want to remove.
- Click Remove.
- Click OK.
Choppy Audio
Users may describe this type of audio issue as missing words, missing parts of words, or periodic moments of silence. This type of issue is typically related to networking or bandwidth issues. To confirm the quality of service for each call, follow the steps below.
- Click on the status icon.
- Click Call History.
- Check the From QoS and To QoS columns.
If these columns are not visible, you can enable them by hovering the cursor above any column and clicking the arrow to open the menu.
QoS is reported as the ratio of packets lost to the total number of packets. If you find a high number of packets lost relative to the total packets, it means that your bandwidth or network connection may not be suitable for IP based calls.
- A typical call may consume approximately 64 kilobits per second. Ensure that the network connection has sufficient bandwidth to accommodate the maximum number of anticipated call volume.
- You may also use third-party tools to test the quality of the signal. This is typically measured by the level of interference or jitter.