The test works by randomly dropping small bursts of packets until enough NACKs have been sent back by the receiver. Retransmitted packets are never dropped in order to assure that all packets are eventually delivered. When enough NACK packets have been received and all dropped packets retransmitted, the test waits for the receiving side to send a number of RTCP packets without NACK lists to assure that the receiving side stops sending NACKs once packets have been retransmitted. BUG=2043 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1934004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4482 4adac7df-926f-26a2-2b94-8c16560cd09d
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.