WebRtc是一款非常优秀的实时音视频通信框架，采用RFC方案SDP offer/answer的方式建立网络会话。

1. 发起Setup Call

1).call and create local

CreatePeerConnection()
GetUserMedia() => AddLocalStream()/SetLocalRender()

2).send offer

createOffer() => SetLocalSdp(offer) => SendOffer()

3).recv answer

RecvAnswer() => SetRemoteSdp(answer)
OnRemoteStream() => SetRemoteRender()

2. 应答Answer Call

1).recv call

CreatePeerConnection()

2).recv offer

RecvOffer() => SetRemoteSdp()
GetUserMedia() => AddLocalStream()/SetLocalRender()

3).send answer

createAnswer() => SetLocalSdp(answer) => SendAnswer()
OnRemoteStream() => SetRemoteRender()

3. stun网络

下面将就网络的基本类型作简单描述，以便理解stun的作用，假设内网主机I:i, 防火墙F:f, 外网主机S1:s1和S2:s2.

1) full cone: I:i <=> F:f, 任意外网主机均可通过F:f发给I:i

2) restricted cone: I:i <=> F:f <=> S1, I:i <=> F:f <=> S2, 只有S1和S2可以通过F:f发给I:i

3) port restricted cone: I:i <=> F:f <=> S1:s1, I:i <=> F:f <=> S2:s2, 只有S1:s1和S2:s2可以通过F:f发数据给I:i

4) symmetric NAT: I:i <=> F:f1 <=> S1:s1, I:i <=> F:f2 <=> S2:s2

基本处理流程如下(来着mozilla):

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相关webrtc源代码封装库和浏览器插件实现，可以参照librtc和RTCHub。