RTP (Real Time Transport Protocol) is used to carry the signaling traffic that sets up a Voice over IP call.

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Multiple Choice

RTP (Real Time Transport Protocol) is used to carry the signaling traffic that sets up a Voice over IP call.

RTP is designed to carry the real-time media itself—audio or video—once a call has been established. Signaling traffic, which sets up, manages, and terminates the call, is handled by separate signaling protocols such as SIP or H.323. Those signaling messages negotiate capabilities, determine media formats, and establish the session, after which RTP streams transport the actual audio or video between endpoints. RTP includes timing, sequencing, and payload type information for the media, while RTCP provides feedback about quality. So, the statement is false: signaling traffic does not ride over RTP; signaling is separate, and RTP carries the real-time media.

RTP is designed to carry the real-time media itself—audio or video—once a call has been established. Signaling traffic, which sets up, manages, and terminates the call, is handled by separate signaling protocols such as SIP or H.323. Those signaling messages negotiate capabilities, determine media formats, and establish the session, after which RTP streams transport the actual audio or video between endpoints. RTP includes timing, sequencing, and payload type information for the media, while RTCP provides feedback about quality. So, the statement is false: signaling traffic does not ride over RTP; signaling is separate, and RTP carries the real-time media.

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