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1. Use more aggressive resyncronization algorithm. 2. Respect the marker bit in RTP packets and do immediate resync on such packets without increasing the jitter delay. This makes the jitter buffer work well with UAs that use silence suppression. Tested with Cisco and X-Lite. 3. The AmPlayoutBuffer has been modified to allow the audio data be put directly in the audio buffer by real timestamp from RTP. This change can possibly break the adaptive playout buffer so it has been put in "#ifdef USE_ADAPTIVE_JB" blocks. The last change is required due to the fact that internal audio packet size (in timestamp units) can differ from RTP packet size and so the internal audio packet can be overlapped by several RTP packets or vice versa. While the original playout buffer is generally ok but the problem is in the loss concealments that can break the buffer as the concealment algorithm can refuse to return any data and so breaking the timestamp monotony and thus causing the voice corruption. Sponsored by: Digifonica Canada Limited Work done by: Sippy Software, Inc. git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@234 8eb893ce-cfd4-0310-b710-fb5ebe64c474sayer/1.4-spce2.6
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