webrtc/rtc_tools/data_channel_benchmark/peer_connection_signaling.proto
Florent Castelli 023be3c977 Data Channel Benchmarking tool
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}
2022-03-15 16:18:16 +00:00

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Protocol Buffer

syntax = "proto3";
package webrtc.GrpcSignaling;
service PeerConnectionSignaling {
rpc Connect(stream SignalingMessage) returns (stream SignalingMessage) {}
}
message SignalingMessage {
oneof Content {
SessionDescription description = 1;
IceCandidate candidate = 2;
}
}
message SessionDescription {
enum SessionDescriptionType {
OFFER = 0;
ANSWER = 1;
}
SessionDescriptionType type = 1;
string content = 2;
}
message IceCandidate {
string mid = 1;
int32 mline_index = 2;
string description = 3;
}