| /* |
| * 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 API_ORTC_RTPTRANSPORTINTERFACE_H_ |
| #define API_ORTC_RTPTRANSPORTINTERFACE_H_ |
| |
| #include <string> |
| |
| #include "absl/types/optional.h" |
| #include "api/ortc/packettransportinterface.h" |
| #include "api/rtcerror.h" |
| #include "api/rtp_headers.h" |
| #include "api/rtpparameters.h" |
| #include "common_types.h" // NOLINT(build/include) |
| |
| 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: |
| // https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx |
| class RtpTransportInterface { |
| public: |
| 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; |
| |
| protected: |
| // 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 |
| |
| #endif // API_ORTC_RTPTRANSPORTINTERFACE_H_ |