Audio delay in Grandstream IP Phone

Audio delay issue in Grandstream IP Phones GRP260x and IP PBX UCM

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Audio delay in Grandstream IP Phone

Description

In an IP phone system, audio delay can occur due to network latency, jitter, and packet loss. In addition, audio codecs and processing that are used in IP phone systems can also contribute to audio delay.

To reduce audio delay in IP phones, network administrators can use Quality of Service (QoS) techniques to prioritize audio traffic, use low-latency codecs, and ensure that the network has sufficient bandwidth to handle the audio traffic.

Phone Settings

This article suggests disabling Enable Audio RED with FEC option under Account > Codec Settings to prevent the audio delay issue.

Figure 1 shows the analysis of RTP stream using different codecs
Figure 2 Disable Audio RED with FEC to prevent audio delay in GRP260X phones

UCM Settings

1. Disable the Jitter Buffer in the VoIP trunk’s Basic Settings. Excessive use of a jitter buffer can lead to delays. Consider toggling the jitter buffer methods and monitor the call quality for improvements.

Disable Jitter Buffer to avoid delay in Grandstrem UCM
Figure 3 Disable Jitter Buffer in the VoIP trunk to avoid delay in all calls

2. Disable Jitter Buffer and Audio FEC under the UCM’s Extensions > Media Setting

Disable Audio FEC and Jitter Buffer in UCM extensions
Figure 4 Disable Audio FEC and Jitter Buffer in UCM extensions’ Media Settings

3. If you encounter audio issues on remote phones registered through UCMRC, verify that STUN is properly configured in the phone’s NAT Traversal settings.