RTP stands for Real-time Transport Protocol.

What is the RTP protocol?

RTP is defined in IETF RFC 3550, with many additional RFCs referring to it and adding more functionality to it.

RTP is designed for sending and receiving real time media. To that end, it is implemented on top of UDP and focuses on low latency delivery.

RTP’s packet header is 12 bytes (or more) at length, keeping the overhead of packetization on top of the actual data being sent to a minimum.

RTP packets include a timestamp and a sequence number in them to deal with network issues such as jitter and out of order packets.

RTP is almost always paired with RTCP. The secured version of RTP is called SRTP.

What is the purpose of RTP in VoIP and WebRTC?

RTP is a very popular transport protocol that is used by most of the VoIP protocols and solutions today. VoIP services that don’t make direct use of RTP usually have their own proprietary implementation that operates in similar ways.

The popularity of RTP stems from its wide support of media codecs, its use of UDP and its low overhead.

WebRTC does NOT use RTP. That is because RTP isn’t secured. WebRTC uses SRTP instead which can be viewed as the secure variant of RTP.

[vimeo https://vimeo.com/198382107 w=590]

About WebRTC Glossary

The WebRTC Glossary is an ongoing project where users can learn more about WebRTC related terms. It is maintained by Tsahi Levent-Levi of BlogGeek.me.

Looking to learn more about WebRTC? 

Check my WebRTC training courses