| #ifndef RTCWEB_H |
| #define RTCWEB_H |
| |
| |
| class StateNotifier |
| { |
| |
| public: |
| |
| // Called when the state of the session changes. |
| // INIT->SENT_OFFER->RECEIVED_ANSWER->INPROGRESS->TERMINATED |
| virtual void onStateChange(int newState, char * stateInfo)=0; |
| }; |
| |
| |
| |
| class Session |
| { |
| public: |
| |
| static Session * create(char* id, StateNotifier & obj); |
| |
| |
| // generates a session description |
| virtual int generateLocalDescription(char * desc, int maxLen) = 0; |
| |
| // configures the local media options |
| virtual int setLocalDescription(char * desc, int maxLenDesc, char * type, int maxLenType) = 0; |
| |
| // configures the remote media options |
| virtual int setRemoteDescription(char * desc, int maxLenDesc, char * type, int maxLenType) = 0; |
| |
| // Starts or stops sending/receiving media. |
| virtual int enable(bool enable) = 0; |
| |
| // Mutes or unmutes the sending of media. |
| virtual int mute(char * media, int maxLen, bool mute) = 0; |
| |
| // Sends a DTMF tone (for use telephony situations) |
| virtual int sendDTMF(int event) = 0; |
| |
| // Adds an additional stream to the session (for multi-user) |
| virtual int addStream(char * media, int maxLen, int source) = 0; |
| |
| // Removes a stream from the session. |
| virtual int removeStream(char * media, int maxLen, int source) = 0; |
| |
| // Gets a URL for a given stream that can be used by |
| // <video> or another playout destination. The default |
| // stream can be obtained by passing 0. |
| virtual int getStreamURL(char * media, int maxLen, int source) = 0; |
| |
| }; |
| |
| |
| |
| #endif // RTCWEB_H |