A user in location X dials an extension at location Y.
The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio.
What is the cause of this issue?
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A. B. C. D.B.
The most likely cause of choppy or clipped audio during a call that travels through a QoS-enabled WAN network is a codec mismatch. When the two endpoints of a call use different codecs, the audio quality can be negatively impacted, resulting in choppy or clipped audio. This can occur when a call is routed over a WAN network, as different locations may have different codec configurations.
Codec mismatch can occur if the two endpoints of a call use different codecs or if the same codec is configured with different settings, such as different bit rates or packetization periods. When a call is established, the two endpoints negotiate the codec that they will use for the call. If the two endpoints cannot agree on a common codec, the call will not be established, but if a mismatch occurs, the audio quality may be degraded.
Missing Call Admission Control can also cause audio quality issues by allowing too many calls to be established across a WAN link, which can lead to network congestion and packet loss. However, this is less likely to be the cause of choppy or clipped audio than a codec mismatch.
Ptime mismatch can also cause audio quality issues. Ptime is the length of time for each audio packet in a VoIP call. If the ptime value is set differently on the two endpoints, this can result in choppy or clipped audio. However, this is less common than a codec mismatch, and it is usually not the default configuration.
A phone class of service issue could also cause audio quality problems. For example, if the phone is configured with a low priority class of service, it may not be allocated sufficient bandwidth for the call, resulting in degraded audio quality. However, this is unlikely to be the cause of choppy or clipped audio in a QoS-enabled WAN network.