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RTP alone isn't going to supply every single doable attribute, And that's why other protocols are employed by WebRTC. Some of the much more noteworthy items RTP isn't going to contain:
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Dependability: SRTP’s encryption and authentication characteristics improve the dependability of communication channels, lowering the potential risk of facts breaches.
When RTP is meant to be used in latency-critical situations, it isn't going to inherently offer you any attributes that guarantee QoS. In its place, it only provides the information needed to let QoS to get applied somewhere else while in the stack.
Just before inspecting RTP's use in WebRTC contexts, It can be beneficial to possess a basic notion of what RTP does and will not offer. RTP is an information transportation protocol, whose mission is to maneuver information involving two endpoints as effectively as is possible under recent circumstances.
RTP is made to have a multitude of multimedia formats, which permits the event of new formats devoid of revising the RTP normal. To this finish, the information expected by a certain software on the protocol isn't A part of the generic RTP header.
Authentic-time multimedia streaming apps call for timely delivery of data and sometimes can tolerate some packet loss to attain this objective. Such as, lack of a packet in an audio software may perhaps end in loss of a fraction of the second of audio data, that may be built unnoticeable with acceptable mistake concealment algorithms.[five] The Transmission Handle Protocol (TCP), Though standardized for RTP use,[six] will not be Commonly Utilized in RTP apps mainly because TCP favors dependability around timeliness.
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