blob: ea1f5375ea9e8ec259bd0aa1cc37611b5251ff96 [file] [log] [blame]
* 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 "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<> {
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.
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