Asterisk Voice Delay

Unlike standard telephony, where there is no perceivable delay between voice transmission and receipt, at least on local calls, VOIP introduces a number of extra layers than can increase this delay.

Here are a few things you can look at to reduce this delay:

  • Ensure all codecs in use are non compressed. Any compression will add additional delays.
  • Use the same codecs on each end. Conversion means extra processing, adding delays.
  • Turn off all jitterbuffers. These introduce a small delay to compensate for fluctuations in latency.
  • Use the smallest packet size possible. Usually this is 30 or 20ms but it may be possible to reduce it, depending on the phone hardware.
  • Ensure the media is not being passed through the PBX, so the phone is sending audio directly to the other endpoint. For sip use -+canrevite=no+=.

Unlike standard telephony, where there is no perceivable delay between voice transmission and receipt, at least on local calls, VOIP introduces a number of extra layers than can increase this delay.

Here are a few things you can look at to reduce this delay:

  • Ensure all codecs in use are non compressed. Any compression will add additional delays.
  • Use the same codecs on each end. Conversion means extra processing, adding delays.
  • Turn off all jitterbuffers. These introduce a small delay to compensate for fluctuations in latency.
  • Use the smallest packet size possible. Usually this is 30 or 20ms but it may be possible to reduce it, depending on the phone hardware.
  • Ensure the media is not being passed through the PBX, so the phone is sending audio directly to the other endpoint. For sip use -+canrevite=no+=.

Created by: stan, Last modification: Mon 06 of Dec, 2010 (11:05 UTC)
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