mirror of https://github.com/sipwise/rtpengine.git
This became necessary because of the way Asterisk handles Sequence numbers when changing SSRC. They continue to increment a single sequence number even though the SSRC is different, on switching back this causes the packet_index function to interpret this as many lost packets. The previous function had dead-spots that would not adjust the packet_index at all if the difference fell in these ranges. These gaps always resulted in behavior contra what would happen in webrtc clients.pull/110/head
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