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Create a server using: ./data_channel_benchmark --server --port 12345 Start the flow of data from the server to a client using: ./data_channel_benchmark --port 12345 --transfer_size 100 The throughput is reported on the server console. The negotiation does not require a 3rd party server and is done over a gRPC transport. No TURN server is configured, so both peers need to be reachable using STUN only. Bug: webrtc:13288 Change-Id: Iac9a96cf390ab465ea45a46bf0b40950c56dfceb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36206} |
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| .. | ||
| audioproc_f | ||
| converter | ||
| data_channel_benchmark | ||
| frame_analyzer | ||
| network_tester | ||
| psnr_ssim_analyzer | ||
| py_event_log_analyzer | ||
| rtc_event_log_visualizer | ||
| rtp_generator | ||
| testing | ||
| unpack_aecdump | ||
| author_line_count.sh | ||
| BUILD.gn | ||
| class_usage.sh | ||
| compare_videos.py | ||
| DEPS | ||
| header_usage.sh | ||
| metrics_plotter.py | ||
| OWNERS | ||
| sanitizers_unittest.cc | ||
| video_file_reader.cc | ||
| video_file_reader.h | ||
| video_file_reader_unittest.cc | ||
| video_file_writer.cc | ||
| video_file_writer.h | ||
| video_file_writer_unittest.cc | ||
| video_replay.cc | ||