| /* |
| * 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. |
| */ |
| |
| #ifndef PC_RTP_TRANSPORT_INTERNAL_H_ |
| #define PC_RTP_TRANSPORT_INTERNAL_H_ |
| |
| #include <string> |
| |
| #include "call/rtp_demuxer.h" |
| #include "p2p/base/ice_transport_internal.h" |
| #include "pc/session_description.h" |
| #include "rtc_base/network_route.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| |
| namespace rtc { |
| class CopyOnWriteBuffer; |
| struct PacketOptions; |
| } // namespace rtc |
| |
| namespace webrtc { |
| |
| // This represents the internal interface beneath SrtpTransportInterface; |
| // 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<> { |
| public: |
| virtual ~RtpTransportInternal() = default; |
| |
| virtual void SetRtcpMuxEnabled(bool enable) = 0; |
| |
| // TODO(zstein): Remove PacketTransport setters. Clients should pass these |
| // in to constructors instead and construct a new RtpTransportInternal instead |
| // of updating them. |
| virtual bool rtcp_mux_enabled() const = 0; |
| |
| virtual rtc::PacketTransportInternal* rtp_packet_transport() const = 0; |
| virtual void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) = 0; |
| |
| virtual rtc::PacketTransportInternal* rtcp_packet_transport() const = 0; |
| virtual void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) = 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. |
| sigslot::signal1<bool> SignalReadyToSend; |
| |
| // Called whenever an RTCP packet is received. There is no equivalent signal |
| // for RTP packets because they would be forwarded to the BaseChannel through |
| // the RtpDemuxer callback. |
| sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived; |
| |
| // Called whenever the network route of the P2P layer transport changes. |
| // The argument is an optional network route. |
| sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged; |
| |
| // Called whenever a transport's writable state might change. The argument is |
| // true if the transport is writable, otherwise it is false. |
| sigslot::signal1<bool> SignalWritableState; |
| |
| sigslot::signal1<const rtc::SentPacket&> SignalSentPacket; |
| |
| 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; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_RTP_TRANSPORT_INTERNAL_H_ |