[从头开始搭建视频会议系统]webrtc添加端到端加密函数的方法:

webrtc添加端到端加密函数的方法:

一：实现自己的加密类。

继承接口:

class HCCrypto :public webrtc::FrameEncryptorInterface {
public:
    void AddRef() const;

    int Encrypt(cricket::MediaType media_type,
                uint32_t ssrc,
                rtc::ArrayView<const uint8_t> additional_data,
                rtc::ArrayView<const uint8_t> frame,
                rtc::ArrayView<uint8_t> encrypted_frame,
                size_t* bytes_written)  ;

    size_t GetMaxCiphertextByteSize(cricket::MediaType media_type,
                                    size_t frame_size);

    rtc::RefCountReleaseStatus Release() const ;
};

 

二.添加到对应的senders

  auto senders = peer_connection_->GetSenders();
  for(auto  sender: senders ){
       rtc::scoped_refptr<HCCrypto> frame_encryptor( new  HCCrypto());
//        sender->
      sender->SetFrameEncryptor(frame_encryptor);
  }

ps:注意添加的时机。需要是在添加完track之后 要不然 。代码里面有判断加入不进去。

void RtpSenderBase::SetFrameEncryptor(
    rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
  frame_encryptor_ = std::move(frame_encryptor);
  // Special Case: Set the frame encryptor to any value on any existing channel.
  if (media_channel_ && ssrc_ && !stopped_) {
    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
      media_channel_->SetFrameEncryptor(ssrc_, frame_encryptor_);
    });
  }
}

 

三。注意看头文件的注释.非常重要.

// FrameEncryptorInterface allows users to provide a custom encryption
// implementation to encrypt all outgoing audio and video frames. The user must
// also provide a FrameDecryptorInterface to be able to decrypt the frames on
// the receiving device. Note this is an additional layer of encryption in
// addition to the standard SRTP mechanism and is not intended to be used
// without it. Implementations of this interface will have the same lifetime as
// the RTPSenders it is attached to. Additional data may be null.
// Note: This interface is not ready for production use.
class FrameEncryptorInterface : public rtc::RefCountInterface {
 public:
  ~FrameEncryptorInterface() override {}

  // Attempts to encrypt the provided frame. You may assume the encrypted_frame
  // will match the size returned by GetMaxCiphertextByteSize for a give frame.
  // You may assume that the frames will arrive in order if SRTP is enabled.
  // The ssrc will simply identify which stream the frame is travelling on. You
  // must set bytes_written to the number of bytes you wrote in the
  // encrypted_frame. 0 must be returned if successful all other numbers can be
  // selected by the implementer to represent error codes.
  virtual int Encrypt(cricket::MediaType media_type,
                      uint32_t ssrc,
                      rtc::ArrayView<const uint8_t> additional_data,
                      rtc::ArrayView<const uint8_t> frame,
                      rtc::ArrayView<uint8_t> encrypted_frame,
                      size_t* bytes_written) = 0;

  // Returns the total required length in bytes for the output of the
  // encryption. This can be larger than the actual number of bytes you need but
  // must never be smaller as it informs the size of the encrypted_frame buffer.
  virtual size_t GetMaxCiphertextByteSize(cricket::MediaType media_type,
                                          size_t frame_size) = 0;
};

 

ps:修改webrtc的代码测试是否添加成功.

  rtc::Buffer encrypted_video_payload;
  if (frame_encryptor_ != nullptr) {
    std::cout<<"++++++++++有加密++++++++++++++++++"<<std::endl;

下次把国密加密添加进来.