The road of WebRTC audio and video development

As early as 2014 through the WebRTC realized the PC client real-time video voice, then the establishment of peer-to WEBRTC with the Libjingle library, using the Peerconnection API implementation. Later in the Remote Desktop, file transfer requires point-to-point connection, the Libjingle library for a period of time, found a few problems:

The 1.libjingle library is built using the XMPP protocol, but our clients do not support the XMPP protocol;

2.libjingle Library is huge, making rewriting difficult.

3. Stability issues.

Then, based on the UDT library to achieve a set of peer-to transfer library, use is also very convenient as long as the two sides agreed to the name of the simultaneous connection to the server can be a successful hole, and later use this library to replace the WEBRTC Libjingle library for transmission.

The implementation details are as follows:

Using WebRTC's voice_engine and video_engine to achieve video voice (because the Voice_engine and Video_engine APIs are simple and easy to maintain), the docking section is as follows:

1 classWebrtc_dllexport voenetwork2 {3  Public:4     //Factory for the voenetwork Sub-api. Increases an internal5     //Reference counter if successful. Returns NULL If the API is not6     //supported or if construction fails.7     Staticvoenetwork* GetInterface (voiceengine*voiceengine);8 9     //releases the Voenetwork Sub-api and decreases an internalTen     //reference counter. Returns the new reference count. This value should One     //Be zero for all sub-api:s before the Voiceengine object can be safely A     //deleted. -     Virtual intRelease () =0; -  the     //installs and enables a user-defined external transport protocol for a -     //specified |channel|. -     Virtual intRegisterexternaltransport ( -         intChannel, transport& Transport) =0; +  -     //removes and disables a user-defined external transport protocol for a +     //specified |channel|. A     Virtual intDeregisterexternaltransport (intChannel) =0; at  -     //The packets received from the network should is passed to this -     //function When external transport is enabled. Note that the data -     //including the Rtp-header must also is given to the voiceengine. -     Virtual intReceivedrtppacket ( -         intChannelConst void* Data, unsignedintLength) =0; in  -     //The packets received from the network should is passed to this to     //function When external transport is enabled. Note that the data +     //including the Rtcp-header must also is given to the voiceengine. -     Virtual intReceivedrtcppacket ( the         intChannelConst void* Data, unsignedintLength) =0; *  $ protected:Panax Notoginseng voenetwork () {} -     Virtual~voenetwork () {} the};

Implement the interface of the transport class sendpacket,sendrtcppacket to send RTP and RTCP packets, call Voenetwork::registerexternaltransport () to register to Voice_ Engine, and then the received packets to voenetwork::receivedrtppacket and Voenetwork::receivedrtcppacket to deal with, the video is similar processing.

The road of WebRTC audio and video development