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Hi, I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space