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AgeCommit message (Expand)Author
2021-05-01Improve call wording, cleanupfiaxh
2021-05-01Fix webcam framerate selectionMarvin W
2021-05-01Correctly handle missing webrtc-audio-processingMarvin W
2021-05-01Echo CancellationMarvin W
2021-04-29Handle non-existant call supportfiaxh
2021-04-29Video optimizationsMarvin W
2021-04-11Handle broken VAPI in older valaMarvin W
2021-04-11Fix custom vapi integrationMarvin W
2021-04-11GStreamer compatMarvin W
2021-04-01Fix bug in legacy SRTP decryptionMarvin W
2021-04-01Remove unnecessary debug codeMarvin W
2021-03-29Migrate to libsrtp2Marvin W
2021-03-29Don't reuse PTs for different media typesMarvin W
2021-03-26Fix device manager usage for GStreamer 1.16Marvin W
2021-03-25Add initial support for DTLS-SRTPfiaxh
2021-03-23Move SRTP implementation into crypto library for reuseMarvin W
2021-03-23Resample audio data for common 48k sample rateMarvin W
2021-03-23Add support for SRTPMarvin W
2021-03-21RTP: Backport gst_caps_copy_nth from GStreamer 1.16Marvin W
2021-03-21Add RTP implementation as pluginMarvin W
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