bdrtc
bdrtc
video frame maybe dropped prior to encode, [according to webrtc code](https://chromium.googlesource.com/external/webrtc/+/master/api/video_codecs/video_encoder.h#65) the capture video frame maybe dropped by encoder's internal rate limiter or MediaOptimizations , but no stats about this...
It seems there is no field describe packet loss rate of specfic audio/video inbound rtp stream, the fractionLost field exist in [RTCRemoteInboundRtpStreamStats](https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats), but it‘s about outbound loss rate. packet loss...
**Background** There is one use case: use webrtc in receive only mode/non-sender(such as online gaming), in this case, client will not send SR, there is no way to calculate round-trip...
1.webrtc support audio nack, but its disabled default, currently user can only active this via munging SDP. most webrtc platform enable audio nack via munging sdp（include google meet also:>）. 2....
Ipv6 is enabled by default in some browsers and there are ipv6 candidates generated by default during setup peerconnection call, but not all webrtc based system support ipv6 now, ipv6...
**Background** we know webrtc provide getStats api for developer debug call, but sometimes this approach is insufficient from a developer perspective when debug online problem, webrtc native have a lot...
**Background** The feature request is to change how browsers complete their ICE Gathering. Currently, after a short period after establishing the call, we stop gathering new candidates and move to...
There is no standard WebRTC API for managing the DTLS-SRTP cipher priority order currently, and the media stream are encrypted using AES-128 by default, for enhanced security, we suggest add...
Opus DTX([Discontinuous Transmission](https://www.rfc-editor.org/rfc/rfc6716#page-13) )is an extension of the Opus Audio codec, when enabled, it will encode silence at a lower bitrate by reducing the number of frames sent over the...
Very excited to see that the one way media scene has been added to the latest webrtc nv use case. Regarding the scenarios inside, I have a few questions. 1....