Introduce SctpOptions struct as argument to SctpTransport::Start
currently containing the SDP-negotiated parameters
- max-message-size
- sctp-port (local and remote)
This deprecates SctpTransportInternal::Start with the individual
parameters. Replace calls to
Start(local_sctp_port, remote_sctp_port, max_message_size);
with
Start({
.local_port = local_sctp_port,
.remote_port = remote_sctp_port,
.max_message_size = max_message_size
});
BUG=webrtc:402429107
Change-Id: I43c60075e28b211bdc95c2040d3cccd0ac96d125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/380580
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#44130}
diff --git a/api/sctp_transport_interface.h b/api/sctp_transport_interface.h
index 173f3ef..c80fddf 100644
--- a/api/sctp_transport_interface.h
+++ b/api/sctp_transport_interface.h
@@ -88,6 +88,26 @@
virtual void UnregisterObserver() = 0;
};
+// The size of the SCTP association send buffer. 256kB, the usrsctp default.
+constexpr int kSctpSendBufferSize = 256 * 1024;
+
+// SCTP options negotiated in the SDP.
+struct SctpOptions {
+ // https://www.rfc-editor.org/rfc/rfc8841.html#name-sctp-port
+ // `local_port` and `remote_port` are passed along the wire and the
+ // listener and connector must be using the same port. They are not related
+ // to the ports at the IP level. If set to -1 we default to
+ // kSctpDefaultPort.
+ // TODO(bugs.webrtc.org/402429107): make these optional<uint16_t>.
+ int local_port = -1;
+ int remote_port = -1;
+
+ // https://www.rfc-editor.org/rfc/rfc8841.html#name-max-message-size
+ // `max_message_size` sets the maxium message size on the connection.
+ // It must be smaller than or equal to kSctpSendBufferSize.
+ int max_message_size = kSctpSendBufferSize;
+};
+
} // namespace webrtc
#endif // API_SCTP_TRANSPORT_INTERFACE_H_
diff --git a/media/BUILD.gn b/media/BUILD.gn
index ab9316c..fcca48c 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -712,6 +712,7 @@
sources = [ "sctp/sctp_transport_internal.h" ]
deps = [
":media_channel",
+ "../api:libjingle_peerconnection_api",
"../api:priority",
"../api:rtc_error",
"../api/transport:datagram_transport_interface",
@@ -1057,6 +1058,7 @@
"../rtc_base/synchronization:mutex",
"../rtc_base/system:file_wrapper",
"../rtc_base/third_party/sigslot",
+ "../system_wrappers",
"../system_wrappers:field_trial",
"../test:audio_codec_mocks",
"../test:fake_video_codecs",
diff --git a/media/sctp/dcsctp_transport.cc b/media/sctp/dcsctp_transport.cc
index 3422b31..e67a266 100644
--- a/media/sctp/dcsctp_transport.cc
+++ b/media/sctp/dcsctp_transport.cc
@@ -28,6 +28,7 @@
#include "api/environment/environment.h"
#include "api/priority.h"
#include "api/rtc_error.h"
+#include "api/sctp_transport_interface.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/data_channel_transport_interface.h"
@@ -192,20 +193,18 @@
MaybeConnectSocket();
}
-bool DcSctpTransport::Start(int local_sctp_port,
- int remote_sctp_port,
- int max_message_size) {
+bool DcSctpTransport::Start(const SctpOptions& options) {
RTC_DCHECK_RUN_ON(network_thread_);
- RTC_DCHECK(max_message_size > 0);
- RTC_DLOG(LS_INFO) << debug_name_ << "->Start(local=" << local_sctp_port
- << ", remote=" << remote_sctp_port
- << ", max_message_size=" << max_message_size << ")";
+ RTC_DCHECK(options.max_message_size > 0);
+ RTC_DLOG(LS_INFO) << debug_name_ << "->Start(local=" << options.local_port
+ << ", remote=" << options.remote_port
+ << ", max_message_size=" << options.max_message_size << ")";
if (!socket_) {
dcsctp::DcSctpOptions options;
- options.local_port = local_sctp_port;
- options.remote_port = remote_sctp_port;
- options.max_message_size = max_message_size;
+ options.local_port = options.local_port;
+ options.remote_port = options.remote_port;
+ options.max_message_size = options.max_message_size;
options.max_timer_backoff_duration = kMaxTimerBackoffDuration;
// Don't close the connection automatically on too many retransmissions.
options.max_retransmissions = std::nullopt;
@@ -226,15 +225,15 @@
socket_ = socket_factory_->Create(debug_name_, *this,
std::move(packet_observer), options);
} else {
- if (local_sctp_port != socket_->options().local_port ||
- remote_sctp_port != socket_->options().remote_port) {
+ if (options.local_port != socket_->options().local_port ||
+ options.remote_port != socket_->options().remote_port) {
RTC_LOG(LS_ERROR)
- << debug_name_ << "->Start(local=" << local_sctp_port
- << ", remote=" << remote_sctp_port
+ << debug_name_ << "->Start(local=" << options.local_port
+ << ", remote=" << options.remote_port
<< "): Can't change ports on already started transport.";
return false;
}
- socket_->SetMaxMessageSize(max_message_size);
+ socket_->SetMaxMessageSize(options.max_message_size);
}
MaybeConnectSocket();
@@ -722,7 +721,10 @@
RTC_DLOG(LS_INFO) << debug_name_ << " DTLS restart";
dcsctp::DcSctpOptions options = socket_->options();
socket_.reset();
- Start(options.local_port, options.remote_port, options.max_message_size);
+ RTC_DCHECK_LE(options.max_message_size, kSctpSendBufferSize);
+ Start({.local_port = options.local_port,
+ .remote_port = options.remote_port,
+ .max_message_size = static_cast<int>(options.max_message_size)});
}
}
diff --git a/media/sctp/dcsctp_transport.h b/media/sctp/dcsctp_transport.h
index f80214b..3d3b1ec 100644
--- a/media/sctp/dcsctp_transport.h
+++ b/media/sctp/dcsctp_transport.h
@@ -11,19 +11,27 @@
#ifndef MEDIA_SCTP_DCSCTP_TRANSPORT_H_
#define MEDIA_SCTP_DCSCTP_TRANSPORT_H_
+#include <cstddef>
+#include <cstdint>
+#include <functional>
#include <memory>
#include <optional>
#include <string>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
+#include "api/dtls_transport_interface.h"
#include "api/environment/environment.h"
#include "api/priority.h"
+#include "api/rtc_error.h"
+#include "api/sctp_transport_interface.h"
#include "api/task_queue/task_queue_base.h"
+#include "api/transport/data_channel_transport_interface.h"
#include "media/sctp/sctp_transport_internal.h"
-#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_message.h"
#include "net/dcsctp/public/dcsctp_socket.h"
#include "net/dcsctp/public/dcsctp_socket_factory.h"
+#include "net/dcsctp/public/timeout.h"
#include "net/dcsctp/public/types.h"
#include "net/dcsctp/timer/task_queue_timeout.h"
#include "p2p/base/packet_transport_internal.h"
@@ -35,7 +43,6 @@
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
-#include "system_wrappers/include/clock.h"
namespace webrtc {
@@ -56,9 +63,7 @@
void SetOnConnectedCallback(std::function<void()> callback) override;
void SetDataChannelSink(DataChannelSink* sink) override;
void SetDtlsTransport(cricket::DtlsTransportInternal* transport) override;
- bool Start(int local_sctp_port,
- int remote_sctp_port,
- int max_message_size) override;
+ bool Start(const SctpOptions& options) override;
bool OpenStream(int sid, PriorityValue priority) override;
bool ResetStream(int sid) override;
RTCError SendData(int sid,
diff --git a/media/sctp/dcsctp_transport_unittest.cc b/media/sctp/dcsctp_transport_unittest.cc
index a6de226..a609e0f 100644
--- a/media/sctp/dcsctp_transport_unittest.cc
+++ b/media/sctp/dcsctp_transport_unittest.cc
@@ -19,6 +19,7 @@
#include "api/priority.h"
#include "api/rtc_error.h"
#include "api/transport/data_channel_transport_interface.h"
+#include "net/dcsctp/public/dcsctp_message.h"
#include "net/dcsctp/public/dcsctp_options.h"
#include "net/dcsctp/public/dcsctp_socket.h"
#include "net/dcsctp/public/mock_dcsctp_socket.h"
@@ -27,6 +28,7 @@
#include "p2p/dtls/fake_dtls_transport.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/thread.h"
+#include "system_wrappers/include/clock.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -107,8 +109,9 @@
&dcsctp::DcSctpSocketCallbacks::OnConnected));
EXPECT_CALL(peer_a.sink_, OnReadyToSend);
EXPECT_CALL(peer_a.sink_, OnConnected);
-
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
}
// Tests that the close sequence invoked from one end results in the stream to
@@ -138,8 +141,12 @@
EXPECT_CALL(peer_b.sink_, OnChannelClosed(1));
}
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
- peer_b.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
+ peer_b.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
peer_a.sctp_transport_->OpenStream(1, kDefaultPriority);
peer_b.sctp_transport_->OpenStream(1, kDefaultPriority);
peer_a.sctp_transport_->ResetStream(1);
@@ -180,8 +187,12 @@
EXPECT_CALL(peer_b.sink_, OnChannelClosed(1));
}
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
- peer_b.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
+ peer_b.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
peer_a.sctp_transport_->OpenStream(1, kDefaultPriority);
peer_b.sctp_transport_->OpenStream(1, kDefaultPriority);
peer_a.sctp_transport_->ResetStream(1);
@@ -217,7 +228,9 @@
EXPECT_CALL(*peer_a.socket_, Send(_, _)).Times(0);
peer_a.sctp_transport_->OpenStream(1, PriorityValue(1337));
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
peer_a.sctp_transport_->OpenStream(2, PriorityValue(3141));
}
@@ -227,7 +240,9 @@
EXPECT_CALL(*peer_a.socket_, Send(_, _)).Times(0);
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
SendDataParams params;
rtc::CopyOnWriteBuffer payload;
@@ -242,7 +257,9 @@
EXPECT_CALL(*peer_a.socket_, Send(_, _)).Times(0);
peer_a.sctp_transport_->OpenStream(1, kDefaultPriority);
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
peer_a.sctp_transport_->ResetStream(1);
SendDataParams params;
@@ -260,7 +277,9 @@
EXPECT_CALL(*peer_a.socket_, options()).WillOnce(ReturnPointee(&options));
peer_a.sctp_transport_->OpenStream(1, kDefaultPriority);
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
SendDataParams params;
rtc::CopyOnWriteBuffer payload;
@@ -276,7 +295,9 @@
.Times(1);
peer_a.sctp_transport_->OpenStream(1, kDefaultPriority);
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
static_cast<dcsctp::DcSctpSocketCallbacks*>(peer_a.sctp_transport_.get())
->OnMessageReceived(
@@ -290,7 +311,9 @@
EXPECT_CALL(peer_a.sink_, OnDataReceived(_, _, _)).Times(0);
peer_a.sctp_transport_->OpenStream(1, kDefaultPriority);
- peer_a.sctp_transport_->Start(5000, 5000, 256 * 1024);
+ peer_a.sctp_transport_->Start({.local_port = 5000,
+ .remote_port = 5000,
+ .max_message_size = 256 * 1024});
static_cast<dcsctp::DcSctpSocketCallbacks*>(peer_a.sctp_transport_.get())
->OnMessageReceived(
diff --git a/media/sctp/sctp_transport_internal.h b/media/sctp/sctp_transport_internal.h
index 24e6fb8..47f62fc 100644
--- a/media/sctp/sctp_transport_internal.h
+++ b/media/sctp/sctp_transport_internal.h
@@ -14,24 +14,21 @@
// TODO(deadbeef): Move SCTP code out of media/, and make it not depend on
// anything in media/.
-#include <memory>
-#include <string>
-#include <vector>
+#include <cstddef>
+#include <cstdint>
+#include <functional>
+#include <optional>
#include "api/priority.h"
#include "api/rtc_error.h"
+#include "api/sctp_transport_interface.h"
#include "api/transport/data_channel_transport_interface.h"
-#include "media/base/media_channel.h"
-#include "p2p/base/packet_transport_internal.h"
#include "p2p/dtls/dtls_transport_internal.h"
#include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/thread.h"
namespace cricket {
// Constants that are important to API users
-// The size of the SCTP association send buffer. 256kB, the usrsctp default.
-constexpr int kSctpSendBufferSize = 256 * 1024;
// The number of outgoing streams that we'll negotiate. Since stream IDs (SIDs)
// are 0-based, the highest usable SID is 1023.
@@ -91,20 +88,20 @@
// completes. This method can be called multiple times, though not if either
// of the ports are changed.
//
- // `local_sctp_port` and `remote_sctp_port` are passed along the wire and the
- // listener and connector must be using the same port. They are not related
- // to the ports at the IP level. If set to -1, we default to
- // kSctpDefaultPort.
- // `max_message_size_` sets the max message size on the connection.
- // It must be smaller than or equal to kSctpSendBufferSize.
- // It can be changed by a secons Start() call.
- //
+ virtual bool Start(const webrtc::SctpOptions& options) = 0;
// TODO(deadbeef): Support calling Start with different local/remote ports
// and create a new association? Not clear if this is something we need to
// support though. See: https://github.com/w3c/webrtc-pc/issues/979
+ [[deprecated("Call with SctpOptions")]]
virtual bool Start(int local_sctp_port,
int remote_sctp_port,
- int max_message_size) = 0;
+ int max_message_size) {
+ return Start({
+ .local_port = local_sctp_port,
+ .remote_port = remote_sctp_port,
+ .max_message_size = max_message_size,
+ });
+ }
// NOTE: Initially there was a "Stop" method here, but it was never used, so
// it was removed.
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index f147c49..20984d0 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -365,6 +365,7 @@
":simulcast_description",
":used_ids",
"../api:field_trials_view",
+ "../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
diff --git a/pc/media_session.cc b/pc/media_session.cc
index ec5c85c..82722a2 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -27,13 +27,13 @@
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
+#include "api/sctp_transport_interface.h"
#include "media/base/codec.h"
#include "media/base/codec_list.h"
#include "media/base/media_constants.h"
#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
-#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/ice_credentials_iterator.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/transport_description.h"
@@ -1267,7 +1267,7 @@
data->set_protocol(secure_transport ? kMediaProtocolUdpDtlsSctp
: kMediaProtocolSctp);
data->set_use_sctpmap(session_options.use_obsolete_sctp_sdp);
- data->set_max_message_size(kSctpSendBufferSize);
+ data->set_max_message_size(webrtc::kSctpSendBufferSize);
auto error = CreateContentOffer(media_description_options, session_options,
RtpHeaderExtensions(), ssrc_generator(),
@@ -1474,10 +1474,11 @@
// we do not implement infinite size messages, reply with
// kSctpSendBufferSize.
if (offer_data_description->max_message_size() <= 0) {
- data_answer->as_sctp()->set_max_message_size(kSctpSendBufferSize);
+ data_answer->as_sctp()->set_max_message_size(webrtc::kSctpSendBufferSize);
} else {
- data_answer->as_sctp()->set_max_message_size(std::min(
- offer_data_description->max_message_size(), kSctpSendBufferSize));
+ data_answer->as_sctp()->set_max_message_size(
+ std::min(offer_data_description->max_message_size(),
+ webrtc::kSctpSendBufferSize));
}
if (!CreateMediaContentAnswer(
offer_data_description, media_description_options, session_options,
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index be8df4a..ab11d66 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -30,6 +30,7 @@
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
+#include "api/sctp_transport_interface.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/fake_payload_type_suggester.h"
#include "call/payload_type.h"
@@ -39,7 +40,6 @@
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "media/base/test_utils.h"
-#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_description_factory.h"
@@ -1656,7 +1656,7 @@
const SctpDataContentDescription* dcd_answer =
dc_answer->media_description()->as_sctp();
EXPECT_FALSE(dc_answer->rejected);
- EXPECT_EQ(kSctpSendBufferSize, dcd_answer->max_message_size());
+ EXPECT_EQ(webrtc::kSctpSendBufferSize, dcd_answer->max_message_size());
}
// Verifies that the order of the media contents in the offer is preserved in
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 90648e2..9f11940 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -2942,20 +2942,17 @@
return observer_;
}
-void PeerConnection::StartSctpTransport(int local_port,
- int remote_port,
- int max_message_size) {
+void PeerConnection::StartSctpTransport(const SctpOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!sctp_mid_s_)
return;
- network_thread()->PostTask(SafeTask(
- network_thread_safety_,
- [this, mid = *sctp_mid_s_, local_port, remote_port, max_message_size] {
+ network_thread()->PostTask(
+ SafeTask(network_thread_safety_, [this, mid = *sctp_mid_s_, options] {
rtc::scoped_refptr<SctpTransport> sctp_transport =
transport_controller_n()->GetSctpTransport(mid);
if (sctp_transport)
- sctp_transport->Start(local_port, remote_port, max_message_size);
+ sctp_transport->Start(options);
}));
}
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 0667f77..39d7b2c 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -412,9 +412,7 @@
// Asynchronously calls SctpTransport::Start() on the network thread for
// `sctp_mid()` if set. Called as part of setting the local description.
- void StartSctpTransport(int local_port,
- int remote_port,
- int max_message_size) override;
+ void StartSctpTransport(const SctpOptions& options) override;
// Returns the CryptoOptions for this PeerConnection. This will always
// return the RTCConfiguration.crypto_options if set and will only default
diff --git a/pc/peer_connection_internal.h b/pc/peer_connection_internal.h
index 95c4268..ce69fbd 100644
--- a/pc/peer_connection_internal.h
+++ b/pc/peer_connection_internal.h
@@ -31,6 +31,7 @@
#include "api/rtc_error.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
+#include "api/sctp_transport_interface.h"
#include "call/call.h"
#include "call/payload_type_picker.h"
#include "p2p/base/port.h"
@@ -124,9 +125,15 @@
bool fire_callback = true) = 0;
// Asynchronously calls SctpTransport::Start() on the network thread for
// `sctp_mid()` if set. Called as part of setting the local description.
+ virtual void StartSctpTransport(const SctpOptions& options) = 0;
+ [[deprecated("Call with SctpOptions")]]
virtual void StartSctpTransport(int local_port,
int remote_port,
- int max_message_size) = 0;
+ int max_message_size) {
+ return StartSctpTransport({.local_port = local_port,
+ .remote_port = remote_port,
+ .max_message_size = max_message_size});
+ }
// Asynchronously adds a remote candidate on the network thread.
virtual void AddRemoteCandidate(absl::string_view mid,
diff --git a/pc/sctp_transport.cc b/pc/sctp_transport.cc
index f26d7ca..78f50ff 100644
--- a/pc/sctp_transport.cc
+++ b/pc/sctp_transport.cc
@@ -11,14 +11,25 @@
#include "pc/sctp_transport.h"
#include <algorithm>
+#include <cstddef>
+#include <memory>
#include <optional>
#include <utility>
#include "api/dtls_transport_interface.h"
#include "api/priority.h"
+#include "api/rtc_error.h"
+#include "api/scoped_refptr.h"
+#include "api/sctp_transport_interface.h"
#include "api/sequence_checker.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "media/sctp/sctp_transport_internal.h"
+#include "p2p/dtls/dtls_transport_internal.h"
+#include "pc/dtls_transport.h"
#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"
+#include "rtc_base/thread.h"
namespace webrtc {
@@ -143,14 +154,13 @@
UpdateInformation(SctpTransportState::kClosed);
}
-void SctpTransport::Start(int local_port,
- int remote_port,
- int max_message_size) {
+void SctpTransport::Start(const SctpOptions& options) {
RTC_DCHECK_RUN_ON(owner_thread_);
- info_ = SctpTransportInformation(info_.state(), info_.dtls_transport(),
- max_message_size, info_.MaxChannels());
+ info_ =
+ SctpTransportInformation(info_.state(), info_.dtls_transport(),
+ options.max_message_size, info_.MaxChannels());
- if (!internal()->Start(local_port, remote_port, max_message_size)) {
+ if (!internal()->Start(options)) {
RTC_LOG(LS_ERROR) << "Failed to push down SCTP parameters, closing.";
UpdateInformation(SctpTransportState::kClosed);
}
diff --git a/pc/sctp_transport.h b/pc/sctp_transport.h
index bee64a8..5b7f828 100644
--- a/pc/sctp_transport.h
+++ b/pc/sctp_transport.h
@@ -63,7 +63,7 @@
void Clear();
// Initialize the cricket::SctpTransport. This can be called from
// the signaling thread.
- void Start(int local_port, int remote_port, int max_message_size);
+ void Start(const SctpOptions& options);
// TODO(https://bugs.webrtc.org/10629): Move functions that need
// internal() to be functions on the SctpTransport interface,
diff --git a/pc/sctp_transport_unittest.cc b/pc/sctp_transport_unittest.cc
index d17ad54..0631e35 100644
--- a/pc/sctp_transport_unittest.cc
+++ b/pc/sctp_transport_unittest.cc
@@ -53,9 +53,7 @@
}
void SetDataChannelSink(DataChannelSink* sink) override {}
void SetDtlsTransport(cricket::DtlsTransportInternal* transport) override {}
- bool Start(int local_port, int remote_port, int max_message_size) override {
- return true;
- }
+ bool Start(const SctpOptions& options) override { return true; }
bool OpenStream(int sid, PriorityValue priority) override { return true; }
bool ResetStream(int sid) override { return true; }
RTCError SendData(int sid,
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index cb896c9..96a1cb9 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -5056,9 +5056,9 @@
std::min(local_sctp_description->max_message_size(),
remote_sctp_description->max_message_size());
}
- pc_->StartSctpTransport(local_sctp_description->port(),
- remote_sctp_description->port(),
- max_message_size);
+ pc_->StartSctpTransport({.local_port = local_sctp_description->port(),
+ .remote_port = remote_sctp_description->port(),
+ .max_message_size = max_message_size});
}
}
diff --git a/pc/test/fake_peer_connection_base.h b/pc/test/fake_peer_connection_base.h
index 2022b04..0138dca 100644
--- a/pc/test/fake_peer_connection_base.h
+++ b/pc/test/fake_peer_connection_base.h
@@ -389,9 +389,7 @@
bool fire_callback = true) override {
return RTCError(RTCErrorType::INTERNAL_ERROR, "");
}
- void StartSctpTransport(int local_port,
- int remote_port,
- int max_message_size) override {}
+ void StartSctpTransport(const SctpOptions& options) override {}
void AddRemoteCandidate(absl::string_view mid,
const cricket::Candidate& candidate) override {}
diff --git a/pc/test/mock_peer_connection_internal.h b/pc/test/mock_peer_connection_internal.h
index 785510f..25d9d82 100644
--- a/pc/test/mock_peer_connection_internal.h
+++ b/pc/test/mock_peer_connection_internal.h
@@ -309,7 +309,7 @@
const RtpTransceiverInit&,
bool),
(override));
- MOCK_METHOD(void, StartSctpTransport, (int, int, int), (override));
+ MOCK_METHOD(void, StartSctpTransport, (const SctpOptions&), (override));
MOCK_METHOD(void,
AddRemoteCandidate,
(absl::string_view, const cricket::Candidate&),
diff --git a/test/pc/sctp/BUILD.gn b/test/pc/sctp/BUILD.gn
index 24004d6..e30fd82 100644
--- a/test/pc/sctp/BUILD.gn
+++ b/test/pc/sctp/BUILD.gn
@@ -12,9 +12,14 @@
visibility = [ "*" ]
sources = [ "fake_sctp_transport.h" ]
deps = [
+ "../../../api:libjingle_peerconnection_api",
"../../../api:priority",
+ "../../../api:rtc_error",
"../../../api/environment",
+ "../../../api/transport:datagram_transport_interface",
"../../../api/transport:sctp_transport_factory_interface",
"../../../media:rtc_data_sctp_transport_internal",
+ "../../../rtc_base:checks",
+ "../../../rtc_base:copy_on_write_buffer",
]
}
diff --git a/test/pc/sctp/fake_sctp_transport.h b/test/pc/sctp/fake_sctp_transport.h
index 3f71d44..a22460f 100644
--- a/test/pc/sctp/fake_sctp_transport.h
+++ b/test/pc/sctp/fake_sctp_transport.h
@@ -11,12 +11,20 @@
#ifndef TEST_PC_SCTP_FAKE_SCTP_TRANSPORT_H_
#define TEST_PC_SCTP_FAKE_SCTP_TRANSPORT_H_
+#include <cstddef>
+#include <functional>
#include <memory>
+#include <optional>
#include "api/environment/environment.h"
#include "api/priority.h"
+#include "api/rtc_error.h"
+#include "api/sctp_transport_interface.h"
+#include "api/transport/data_channel_transport_interface.h"
#include "api/transport/sctp_transport_factory_interface.h"
#include "media/sctp/sctp_transport_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
// Used for tests in this file to verify that PeerConnection responds to signals
// from the SctpTransport correctly, and calls Start with the correct
@@ -26,10 +34,10 @@
void SetOnConnectedCallback(std::function<void()> callback) override {}
void SetDataChannelSink(webrtc::DataChannelSink* sink) override {}
void SetDtlsTransport(cricket::DtlsTransportInternal* transport) override {}
- bool Start(int local_port, int remote_port, int max_message_size) override {
- local_port_.emplace(local_port);
- remote_port_.emplace(remote_port);
- max_message_size_ = max_message_size;
+ bool Start(const webrtc::SctpOptions& options) override {
+ local_port_.emplace(options.local_port);
+ remote_port_.emplace(options.remote_port);
+ max_message_size_ = options.max_message_size;
return true;
}
bool OpenStream(int sid, webrtc::PriorityValue priority) override {
