blob: ec712163da23d20e02e22e352817ca34cb2095fa [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 "absl/types/optional.h"
#include "api/ortc/packet_transport_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
namespace webrtc {
class RtpTransportAdapter;
struct RtpTransportParameters final {
RtcpParameters rtcp;
// Enabled periodic sending of keep-alive packets, that help prevent timeouts
// on the network level, such as NAT bindings. See RFC6263 section 4.6.
RtpKeepAliveConfig keepalive;
bool operator==(const RtpTransportParameters& o) const {
return rtcp == o.rtcp && keepalive == o.keepalive;
bool operator!=(const RtpTransportParameters& o) const {
return !(*this == o);
// Base class for different types of RTP transports that can be created by an
// OrtcFactory. Used by RtpSenders/RtpReceivers.
// This is not present in the standard ORTC API, but exists here for a few
// reasons. Firstly, it allows different types of RTP transports to be used:
// DTLS-SRTP (which is required for the web), but also SDES-SRTP and
// unencrypted RTP. It also simplifies the handling of RTCP muxing, and
// provides a better API point for it.
// Note that Edge's implementation of ORTC provides a similar API point, called
// RTCSrtpSdesTransport:
class RtpTransportInterface {
virtual ~RtpTransportInterface() {}
// Returns packet transport that's used to send RTP packets.
virtual PacketTransportInterface* GetRtpPacketTransport() const = 0;
// Returns separate packet transport that's used to send RTCP packets. If
// RTCP multiplexing is being used, returns null.
virtual PacketTransportInterface* GetRtcpPacketTransport() const = 0;
// Set/get RTP/RTCP transport params. Can be used to enable RTCP muxing or
// reduced-size RTCP if initially not enabled.
// Changing |mux| from "true" to "false" is not allowed, and changing the
// CNAME is currently unsupported.
// RTP keep-alive settings need to be set before before an RtpSender has
// started sending, altering the payload type or timeout interval after this
// point is not supported. The parameters must also match across all RTP
// transports for a given RTP transport controller.
virtual RTCError SetParameters(const RtpTransportParameters& parameters) = 0;
// Returns last set or constructed-with parameters. If |cname| was empty in
// construction, the generated CNAME will be present in the returned
// parameters (see above).
virtual RtpTransportParameters GetParameters() const = 0;
// Only for internal use. Returns a pointer to an internal interface, for use
// by the implementation.
virtual RtpTransportAdapter* GetInternal() = 0;
// Classes that can use this internal interface.
friend class OrtcFactory;
friend class OrtcRtpSenderAdapter;
friend class OrtcRtpReceiverAdapter;
friend class RtpTransportControllerAdapter;
friend class RtpTransportAdapter;
} // namespace webrtc