Unverified50% confidenceFactExact time
RTP is responsible for carrying audio and video streams and reassembling them in order, RTCP is responsible for monitoring jitter, packet loss, and round-trip delay, SIP is responsible for signaling to complete the state machine management of call setup, negotiation, and hang-up, and WebRTC encapsulates these capabilities into the browser and solves NAT traversal through ICE/STUN/TURN.
1
Sources
50%
Confidence
Long-term
Relevance
10/12/2026
First Seen
Sources
Related Entities
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