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+=.