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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}
221 lines
7 KiB
C++
221 lines
7 KiB
C++
/*
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* Copyright 2013 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_utils.h"
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#include <stddef.h>
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#include <stdint.h>
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#include "rtc_base/byte_buffer.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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// Format defined at
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// http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-01#section
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static const uint8_t DATA_CHANNEL_OPEN_MESSAGE_TYPE = 0x03;
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static const uint8_t DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE = 0x02;
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enum DataChannelOpenMessageChannelType {
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DCOMCT_ORDERED_RELIABLE = 0x00,
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DCOMCT_ORDERED_PARTIAL_RTXS = 0x01,
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DCOMCT_ORDERED_PARTIAL_TIME = 0x02,
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DCOMCT_UNORDERED_RELIABLE = 0x80,
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DCOMCT_UNORDERED_PARTIAL_RTXS = 0x81,
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DCOMCT_UNORDERED_PARTIAL_TIME = 0x82,
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};
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bool IsOpenMessage(const rtc::CopyOnWriteBuffer& payload) {
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// Format defined at
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// http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
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if (payload.size() < 1) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message type.";
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return false;
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}
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uint8_t message_type = payload[0];
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return message_type == DATA_CHANNEL_OPEN_MESSAGE_TYPE;
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}
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bool ParseDataChannelOpenMessage(const rtc::CopyOnWriteBuffer& payload,
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std::string* label,
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DataChannelInit* config) {
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// Format defined at
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// http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
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rtc::ByteBufferReader buffer(payload.data<char>(), payload.size());
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uint8_t message_type;
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if (!buffer.ReadUInt8(&message_type)) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message type.";
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return false;
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}
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if (message_type != DATA_CHANNEL_OPEN_MESSAGE_TYPE) {
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RTC_LOG(LS_WARNING) << "Data Channel OPEN message of unexpected type: "
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<< message_type;
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return false;
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}
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uint8_t channel_type;
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if (!buffer.ReadUInt8(&channel_type)) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message channel type.";
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return false;
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}
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uint16_t priority;
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if (!buffer.ReadUInt16(&priority)) {
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RTC_LOG(LS_WARNING)
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<< "Could not read OPEN message reliabilility prioirty.";
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return false;
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}
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uint32_t reliability_param;
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if (!buffer.ReadUInt32(&reliability_param)) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message reliabilility param.";
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return false;
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}
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uint16_t label_length;
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if (!buffer.ReadUInt16(&label_length)) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message label length.";
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return false;
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}
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uint16_t protocol_length;
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if (!buffer.ReadUInt16(&protocol_length)) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message protocol length.";
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return false;
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}
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if (!buffer.ReadString(label, (size_t)label_length)) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message label";
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return false;
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}
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if (!buffer.ReadString(&config->protocol, protocol_length)) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN message protocol.";
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return false;
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}
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config->ordered = true;
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switch (channel_type) {
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case DCOMCT_UNORDERED_RELIABLE:
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case DCOMCT_UNORDERED_PARTIAL_RTXS:
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case DCOMCT_UNORDERED_PARTIAL_TIME:
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config->ordered = false;
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}
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config->maxRetransmits = absl::nullopt;
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config->maxRetransmitTime = absl::nullopt;
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switch (channel_type) {
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case DCOMCT_ORDERED_PARTIAL_RTXS:
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case DCOMCT_UNORDERED_PARTIAL_RTXS:
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config->maxRetransmits = reliability_param;
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break;
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case DCOMCT_ORDERED_PARTIAL_TIME:
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case DCOMCT_UNORDERED_PARTIAL_TIME:
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config->maxRetransmitTime = reliability_param;
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break;
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}
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return true;
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}
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bool ParseDataChannelOpenAckMessage(const rtc::CopyOnWriteBuffer& payload) {
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if (payload.size() < 1) {
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RTC_LOG(LS_WARNING) << "Could not read OPEN_ACK message type.";
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return false;
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}
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uint8_t message_type = payload[0];
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if (message_type != DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE) {
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RTC_LOG(LS_WARNING) << "Data Channel OPEN_ACK message of unexpected type: "
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<< message_type;
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return false;
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}
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return true;
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}
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bool WriteDataChannelOpenMessage(const std::string& label,
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const DataChannelInit& config,
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rtc::CopyOnWriteBuffer* payload) {
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// Format defined at
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// http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-09#section-5.1
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uint8_t channel_type = 0;
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uint32_t reliability_param = 0;
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uint16_t priority = 0;
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if (config.ordered) {
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if (config.maxRetransmits) {
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channel_type = DCOMCT_ORDERED_PARTIAL_RTXS;
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reliability_param = *config.maxRetransmits;
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} else if (config.maxRetransmitTime) {
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channel_type = DCOMCT_ORDERED_PARTIAL_TIME;
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reliability_param = *config.maxRetransmitTime;
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} else {
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channel_type = DCOMCT_ORDERED_RELIABLE;
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}
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} else {
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if (config.maxRetransmits) {
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channel_type = DCOMCT_UNORDERED_PARTIAL_RTXS;
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reliability_param = *config.maxRetransmits;
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} else if (config.maxRetransmitTime) {
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channel_type = DCOMCT_UNORDERED_PARTIAL_TIME;
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reliability_param = *config.maxRetransmitTime;
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} else {
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channel_type = DCOMCT_UNORDERED_RELIABLE;
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}
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}
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rtc::ByteBufferWriter buffer(NULL,
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20 + label.length() + config.protocol.length(),
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rtc::ByteBuffer::ORDER_NETWORK);
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// TODO(tommi): Add error handling and check resulting length.
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buffer.WriteUInt8(DATA_CHANNEL_OPEN_MESSAGE_TYPE);
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buffer.WriteUInt8(channel_type);
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buffer.WriteUInt16(priority);
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buffer.WriteUInt32(reliability_param);
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buffer.WriteUInt16(static_cast<uint16_t>(label.length()));
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buffer.WriteUInt16(static_cast<uint16_t>(config.protocol.length()));
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buffer.WriteString(label);
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buffer.WriteString(config.protocol);
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payload->SetData(buffer.Data(), buffer.Length());
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return true;
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}
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void WriteDataChannelOpenAckMessage(rtc::CopyOnWriteBuffer* payload) {
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uint8_t data = DATA_CHANNEL_OPEN_ACK_MESSAGE_TYPE;
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payload->SetData(&data, sizeof(data));
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}
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cricket::DataMessageType ToCricketDataMessageType(DataMessageType type) {
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switch (type) {
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case DataMessageType::kText:
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return cricket::DMT_TEXT;
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case DataMessageType::kBinary:
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return cricket::DMT_BINARY;
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case DataMessageType::kControl:
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return cricket::DMT_CONTROL;
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default:
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return cricket::DMT_NONE;
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}
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return cricket::DMT_NONE;
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}
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DataMessageType ToWebrtcDataMessageType(cricket::DataMessageType type) {
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switch (type) {
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case cricket::DMT_TEXT:
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return DataMessageType::kText;
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case cricket::DMT_BINARY:
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return DataMessageType::kBinary;
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case cricket::DMT_CONTROL:
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return DataMessageType::kControl;
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case cricket::DMT_NONE:
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default:
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RTC_NOTREACHED();
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}
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return DataMessageType::kControl;
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}
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} // namespace webrtc
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