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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <string>
#include <utility>
#include "call/rtp_demuxer.h"
#include "p2p/base/ice_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/network_route.h"
#include "rtc_base/ssl_stream_adapter.h"
namespace rtc {
class CopyOnWriteBuffer;
struct PacketOptions;
} // namespace rtc
namespace webrtc {
// This class is an internal interface; it is not accessible to API consumers
// but is accessible to internal classes in order to send and receive RTP and
// RTCP packets belonging to a single RTP session. Additional convenience and
// configuration methods are also provided.
class RtpTransportInternal : public sigslot::has_slots<> {
virtual ~RtpTransportInternal() = default;
virtual void SetRtcpMuxEnabled(bool enable) = 0;
virtual const std::string& transport_name() const = 0;
// Sets socket options on the underlying RTP or RTCP transports.
virtual int SetRtpOption(rtc::Socket::Option opt, int value) = 0;
virtual int SetRtcpOption(rtc::Socket::Option opt, int value) = 0;
virtual bool rtcp_mux_enabled() const = 0;
virtual bool IsReadyToSend() const = 0;
// Called whenever a transport's ready-to-send state changes. The argument
// is true if all used transports are ready to send. This is more specific
// than just "writable"; it means the last send didn't return ENOTCONN.
void SubscribeReadyToSend(const void* tag,
absl::AnyInvocable<void(bool)> callback) {
callback_list_ready_to_send_.AddReceiver(tag, std::move(callback));
void UnsubscribeReadyToSend(const void* tag) {
// Called whenever an RTCP packet is received. There is no equivalent signal
// for demuxable RTP packets because they would be forwarded to the
// BaseChannel through the RtpDemuxer callback.
void SubscribeRtcpPacketReceived(
const void* tag,
absl::AnyInvocable<void(rtc::CopyOnWriteBuffer*, int64_t)> callback) {
callback_list_rtcp_packet_received_.AddReceiver(tag, std::move(callback));
// There doesn't seem to be a need to unsubscribe from this signal.
// Called whenever a RTP packet that can not be demuxed by the transport is
// received.
void SetUnDemuxableRtpPacketReceivedHandler(
absl::AnyInvocable<void(RtpPacketReceived&)> callback) {
callback_undemuxable_rtp_packet_received_ = std::move(callback);
// Called whenever the network route of the P2P layer transport changes.
// The argument is an optional network route.
void SubscribeNetworkRouteChanged(
const void* tag,
absl::AnyInvocable<void(absl::optional<rtc::NetworkRoute>)> callback) {
callback_list_network_route_changed_.AddReceiver(tag, std::move(callback));
void UnsubscribeNetworkRouteChanged(const void* tag) {
// Called whenever a transport's writable state might change. The argument is
// true if the transport is writable, otherwise it is false.
void SubscribeWritableState(const void* tag,
absl::AnyInvocable<void(bool)> callback) {
callback_list_writable_state_.AddReceiver(tag, std::move(callback));
void UnsubscribeWritableState(const void* tag) {
void SubscribeSentPacket(
const void* tag,
absl::AnyInvocable<void(const rtc::SentPacket&)> callback) {
callback_list_sent_packet_.AddReceiver(tag, std::move(callback));
void UnsubscribeSentPacket(const void* tag) {
virtual bool IsWritable(bool rtcp) const = 0;
// TODO(zhihuang): Pass the `packet` by copy so that the original data
// wouldn't be modified.
virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) = 0;
virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) = 0;
// This method updates the RTP header extension map so that the RTP transport
// can parse the received packets and identify the MID. This is called by the
// BaseChannel when setting the content description.
// TODO(zhihuang): Merging and replacing following methods handling header
// extensions with SetParameters:
// UpdateRtpHeaderExtensionMap,
// UpdateSendEncryptedHeaderExtensionIds,
// UpdateRecvEncryptedHeaderExtensionIds,
// CacheRtpAbsSendTimeHeaderExtension,
virtual void UpdateRtpHeaderExtensionMap(
const cricket::RtpHeaderExtensions& header_extensions) = 0;
virtual bool IsSrtpActive() const = 0;
virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
RtpPacketSinkInterface* sink) = 0;
virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
void SendReadyToSend(bool arg) { callback_list_ready_to_send_.Send(arg); }
void SendRtcpPacketReceived(rtc::CopyOnWriteBuffer* buffer,
int64_t packet_time_us) {
callback_list_rtcp_packet_received_.Send(buffer, packet_time_us);
void NotifyUnDemuxableRtpPacketReceived(RtpPacketReceived& packet) {
void SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute> route) {
void SendWritableState(bool state) {
void SendSentPacket(const rtc::SentPacket& packet) {
CallbackList<bool> callback_list_ready_to_send_;
CallbackList<rtc::CopyOnWriteBuffer*, int64_t>
callback_undemuxable_rtp_packet_received_ =
[](RtpPacketReceived& packet) {};
CallbackList<bool> callback_list_writable_state_;
CallbackList<const rtc::SentPacket&> callback_list_sent_packet_;
} // namespace webrtc