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023be3c977
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}
64 lines
2.1 KiB
C++
64 lines
2.1 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef RTC_TOOLS_DATA_CHANNEL_BENCHMARK_GRPC_SIGNALING_H_
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#define RTC_TOOLS_DATA_CHANNEL_BENCHMARK_GRPC_SIGNALING_H_
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#include <memory>
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#include <string>
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#include "api/jsep.h"
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#include "rtc_tools/data_channel_benchmark/signaling_interface.h"
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namespace webrtc {
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// This class defines a server enabling clients to perform a PeerConnection
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// negotiation directly over gRPC.
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// When a client connects, a callback is run to handle the request.
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class GrpcSignalingServerInterface {
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public:
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virtual ~GrpcSignalingServerInterface() = default;
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// Start listening for connections.
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virtual void Start() = 0;
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// Wait for the gRPC server to terminate.
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virtual void Wait() = 0;
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// Stop the gRPC server instance.
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virtual void Stop() = 0;
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// The port the server is listening on.
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virtual int SelectedPort() = 0;
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// Create a gRPC server listening on |port| that will run |callback| on each
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// request. If |oneshot| is true, it will terminate after serving one request.
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static std::unique_ptr<GrpcSignalingServerInterface> Create(
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std::function<void(webrtc::SignalingInterface*)> callback,
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int port,
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bool oneshot);
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};
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// This class defines a client that can connect to a server and perform a
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// PeerConnection negotiation directly over gRPC.
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class GrpcSignalingClientInterface {
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public:
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virtual ~GrpcSignalingClientInterface() = default;
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// Connect the client to the gRPC server.
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virtual bool Start() = 0;
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virtual webrtc::SignalingInterface* signaling_client() = 0;
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// Create a client to connnect to a server at |server_address|.
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static std::unique_ptr<GrpcSignalingClientInterface> Create(
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const std::string& server_address);
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};
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} // namespace webrtc
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#endif // RTC_TOOLS_DATA_CHANNEL_BENCHMARK_GRPC_SIGNALING_H_
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