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bc3eebc722
This is a reland of 487f9a17e4
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
Bug: webrtc:9719
Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29290}
112 lines
3.5 KiB
C++
112 lines
3.5 KiB
C++
/*
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* Copyright 2019 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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#include "pc/sctp_data_channel_transport.h"
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#include "pc/sctp_utils.h"
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namespace webrtc {
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SctpDataChannelTransport::SctpDataChannelTransport(
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cricket::SctpTransportInternal* sctp_transport)
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: sctp_transport_(sctp_transport) {
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sctp_transport_->SignalReadyToSendData.connect(
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this, &SctpDataChannelTransport::OnReadyToSendData);
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sctp_transport_->SignalDataReceived.connect(
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this, &SctpDataChannelTransport::OnDataReceived);
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sctp_transport_->SignalClosingProcedureStartedRemotely.connect(
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this, &SctpDataChannelTransport::OnClosingProcedureStartedRemotely);
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sctp_transport_->SignalClosingProcedureComplete.connect(
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this, &SctpDataChannelTransport::OnClosingProcedureComplete);
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}
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RTCError SctpDataChannelTransport::OpenChannel(int channel_id) {
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sctp_transport_->OpenStream(channel_id);
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return RTCError::OK();
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}
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RTCError SctpDataChannelTransport::SendData(
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int channel_id,
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const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& buffer) {
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// Map webrtc::SendDataParams to cricket::SendDataParams.
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// TODO(mellem): See about unifying these structs.
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cricket::SendDataParams sd_params;
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sd_params.sid = channel_id;
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sd_params.type = ToCricketDataMessageType(params.type);
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sd_params.ordered = params.ordered;
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sd_params.reliable = !(params.max_rtx_count || params.max_rtx_ms);
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sd_params.max_rtx_count = params.max_rtx_count.value_or(-1);
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sd_params.max_rtx_ms = params.max_rtx_ms.value_or(-1);
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cricket::SendDataResult result;
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sctp_transport_->SendData(sd_params, buffer, &result);
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// TODO(mellem): See about changing the interfaces to not require mapping
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// SendDataResult to RTCError and back again.
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switch (result) {
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case cricket::SendDataResult::SDR_SUCCESS:
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return RTCError::OK();
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case cricket::SendDataResult::SDR_BLOCK: {
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// Send buffer is full.
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ready_to_send_ = false;
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return RTCError(RTCErrorType::RESOURCE_EXHAUSTED);
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}
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case cricket::SendDataResult::SDR_ERROR:
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return RTCError(RTCErrorType::NETWORK_ERROR);
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}
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return RTCError(RTCErrorType::NETWORK_ERROR);
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}
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RTCError SctpDataChannelTransport::CloseChannel(int channel_id) {
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sctp_transport_->ResetStream(channel_id);
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return RTCError::OK();
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}
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void SctpDataChannelTransport::SetDataSink(DataChannelSink* sink) {
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sink_ = sink;
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if (sink_ && ready_to_send_) {
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sink_->OnReadyToSend();
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}
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}
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bool SctpDataChannelTransport::IsReadyToSend() const {
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return ready_to_send_;
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}
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void SctpDataChannelTransport::OnReadyToSendData() {
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ready_to_send_ = true;
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if (sink_) {
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sink_->OnReadyToSend();
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}
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}
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void SctpDataChannelTransport::OnDataReceived(
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const cricket::ReceiveDataParams& params,
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const rtc::CopyOnWriteBuffer& buffer) {
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if (sink_) {
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sink_->OnDataReceived(params.sid, ToWebrtcDataMessageType(params.type),
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buffer);
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}
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}
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void SctpDataChannelTransport::OnClosingProcedureStartedRemotely(
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int channel_id) {
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if (sink_) {
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sink_->OnChannelClosing(channel_id);
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}
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}
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void SctpDataChannelTransport::OnClosingProcedureComplete(int channel_id) {
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if (sink_) {
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sink_->OnChannelClosed(channel_id);
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}
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}
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} // namespace webrtc
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