WebRTC General VoIP
Singaling Undefined SIP or H.323
Media Transport RTP/RTCP RTP/RTSP
Security SRTP SRTP, H.235
NAT Traversal STUN/TURN/ICE STUN/TURN/ICE
Video Codec VP8 H.263, H.264
Audio Codec G.711, iLBC, iSAC G.7xx, ..
- 출처 : 왜 WebRTC인가?, "HTML5, 현재를 묻다, 미래를 답하다", 2012,12,7
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