Some personal understanding about WEBRTC

Some personal understanding about WEBRTC
Just participated in the sound network presided over the first WEBRTC conference in Beijing, coupled with reading "hundred asked Freeswtich" written by Daniel, to it has more understanding, record for later review:
1, simple understanding, WEBRTC is a way to achieve web-to-business dial audio and video telephony technology;
2, provide audio and video engine, sound and video codec technology, support G.711,opus, video support VP8,VP9 (depending on browser implementation);
3, open source version does not do audio effects optimization, such as echo suppression, noise reduction;
4, no signaling, developers need to implement or choose the existing protocol such as SIP, the recommended way is to find open source client such as Jssip;
5, only support peer-to, will bring through the problem, multi-party calls do not support, want to implement the function of the developer to implement their own server, the recommended way is to find a soft switch server to implement Media agent, but also to solve the problem of crossing such as FreeSWITCH;
6, support a better browser is Google, Firefox, is currently mainly Google and some audio and video service providers in the push, such as sound network, has not been supported by Microsoft and Apple;

7, love her not easy, start compiling there is a certain difficulty, Chinese books scarce, it is said that "WebRTC authoritative guide" will be published in the Chinese version of the sound network in the near future;
8, the above is not a problem, and finally this is believed to be the most painful point of pain, mobile development needs to do a variety of models to adapt their work.