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We didn't aggregate at all in previous versions and there are apparently various RTP implementations that don't handle aggregation well at all. As part of this also document that for RTSP it is recommended to keep it set to "none" while for WebRTC it should be set to "zero-latency". Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/692> |
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