Maximum Packet Loss and Jitter for Voice Quality
Short answer
B < 1 percent packet loss and < 30 ms jitter
Question
Key Takeaways
The exam tests precise numerical thresholds for voice traffic parameters; the common trap is confusing latency limits (150ms) with jitter or packet loss values.
This question tests the specific ITU-T G.114 and industry-standard thresholds for acceptable VoIP quality, focusing on packet loss and jitter limits. The community consensus confirms that less than 1% packet loss and under 30 ms jitter are required to prevent audible degradation.
Expert Analysis
Why the Answer Is Correct
Option B is correct because industry standards, specifically aligned with Cisco Press materials and ITU recommendations, dictate that voice quality remains high when packet loss is kept below 1% and jitter is under 30 ms. These tight constraints ensure that audio packets arrive consistently and completely, preventing robotic voices or gaps in speech.
Why the Other Options Are Wrong
Options A, C, and D suggest higher tolerances for packet loss or jitter that exceed the thresholds for high-quality real-time communication. For instance, a 10% packet loss (Option A) would result in significant unintelligibility, while 60 ms jitter (Option C) would likely cause buffer underruns and choppy audio without aggressive buffering, which adds latency.
Community Comment Notes
Comment [1] provides a comprehensive breakdown by citing the ITU G.114 standard for latency (≤ 150 ms) alongside the specific jitter (≤ 30 ms) and packet loss (≤ 1%) metrics. Comment [2] reinforces this by referencing page 313 of the Cisco Press CCNP study guide, confirming these exact numbers as the authoritative source for the exam.
Official Reference
https://www.itu.int/rec/T-REC-G.114/en (ITU-T G.114)
This link opens the official vendor documentation in a new tab.
Exam Strategy
Memorize the 'Golden Rules' of VoIP QoS: < 150 ms latency, < 30 ms jitter, < 1% packet loss, and < 150 kbps per call. Distinguish clearly between one-way latency and round-trip time, as exam questions often test the specific directionality mentioned in the prompt.
Community Votes
100% of anonymous learners picked answer B. Votes are pick records left by other test-takers — they are not the verified answer.
Discussion 3 contributions
Compiled from public community threads and reader submissions, lightly edited for clarity. Handles and vote tallies are shown as recorded, and comments reflect the views of individual contributors.
From public community threads 3
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Maleck 👍 3 Answered BAnswer is B. Voice calls, either one-to-one or on a conference connection capability, require the following: ≤ 150 ms of one-way latency from mouth to ear (per the ITU G.114 standard) / ≤ 30 ms jitter / ≤ 1 percent packet loss
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b3532e4 👍 1The correct answer, B. < 1 percent packet loss and < 30 ms jitter, is found on page 313 of the document. It specifies that for voice traffic, a delay of less than 150 ms, jitter of less than 30 ms, and packet loss of less than 1 percent are required to maintain quality(CiscoPress-CCNP-and-CCI…)
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b3532e4 👍 1B. < 1 percent packet loss and < 30 ms jitter
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