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352-001 · Question #426

How should you compensate for jitter on an IP network so it carries real-time VolP traffic with acceptable voice transmission quality?

The correct answer is B. Set up a playout buffer to play back the voice stream. A playout (jitter) buffer compensates for network jitter by collecting and reordering incoming VoIP packets before delivering them at a steady playback rate.

Design Considerations

Question

How should you compensate for jitter on an IP network so it carries real-time VolP traffic with acceptable voice transmission quality?

Options

  • ASet up VAD to replace gaps on speech with comfort noise.
  • BSet up a playout buffer to play back the voice stream.
  • CDeploy RSVP for dynamic VolP packet classification.
  • DChange CODEC from G.729 to G.711.

How the community answered

(18 responses)
  • A
    6% (1)
  • B
    89% (16)
  • C
    6% (1)

Why each option

A playout (jitter) buffer compensates for network jitter by collecting and reordering incoming VoIP packets before delivering them at a steady playback rate.

ASet up VAD to replace gaps on speech with comfort noise.

VAD (Voice Activity Detection) detects silence and replaces it with comfort noise to reduce bandwidth consumption; it does not address packet timing variation caused by jitter.

BSet up a playout buffer to play back the voice stream.Correct

A playout buffer, also called a de-jitter buffer, absorbs variable packet arrival times by holding incoming RTP audio packets for a configurable delay before rendering them to the listener. This smooths out the irregular spacing caused by network jitter, ensuring the voice stream is played back at a constant, uninterrupted rate. It is the standard mechanism for making real-time VoIP traffic tolerable over best-effort IP networks.

CDeploy RSVP for dynamic VolP packet classification.

RSVP is a resource reservation signaling protocol used to request QoS guarantees in the network, but it does not buffer or reorder already-received packets at the endpoint to compensate for jitter.

DChange CODEC from G.729 to G.711.

Switching from G.729 to G.711 changes the codec bitrate and compression, affecting bandwidth and quality, but does nothing to mitigate the timing irregularities introduced by jitter.

Concept tested: Jitter buffer compensation for VoIP real-time traffic

Source: https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/14569-jitter.html

Topics

#jitter compensation#playout buffer#VoIP QoS#real-time traffic

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