Wondering why Cisco Webex calls have poor audio quality? Learn how high network latency and jitter impact VoIP performance and what steps to take to resolve these issues, essential for Cisco Certified Field Technician (CCT) exam success.
Table of Contents
Question
A user reports poor audio quality on Cisco Webex calls. What is the most likely cause?
A. DNS misconfiguration
B. High network latency and jitter
C. Outdated phone firmware
D. Incorrect VLAN assignment
E. Firewall blocking SIP traffic
Answer
B. High network latency and jitter
Explanation
Voice over IP (VoIP) quality is highly sensitive to latency and jitter. Network performance should be checked to ensure stable and low-latency communication.
High network latency and jitter are the most probable reasons for poor audio quality on Cisco Webex calls. VoIP technologies, including Webex, transmit voice as data packets over IP networks. When there is excessive delay (latency) or variability in packet arrival times (jitter), the result is choppy, distorted, or delayed audio, making conversations difficult to follow.
- Latency refers to the time it takes for voice packets to travel from sender to receiver. When latency is high, users experience noticeable delays, overlapping speech, or echoes.
- Jitter is the variation in packet arrival times. High jitter causes audio to sound robotic, choppy, or results in missing words and audio gaps.
Both latency and jitter can be caused by network congestion, misconfigured Quality of Service (QoS), insufficient bandwidth, or unstable Wi-Fi connections. Webex and other VoIP systems use jitter buffers and advanced codecs to mitigate these effects, but persistent high latency and jitter will still degrade call quality.
Other options such as DNS misconfiguration, outdated firmware, incorrect VLAN assignment, or firewall issues can cause connectivity problems or prevent calls from being established, but they are not the primary causes of poor audio quality during an active call.
Maintaining low latency (below 150 ms) and jitter (below 30 ms) is critical for clear VoIP audio. High values in either metric directly result in poor sound quality on Webex and other VoIP calls.
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