Which protocol is responsible for carrying real-time voice traffic in Cisco collaboration networks? Learn why RTP is essential for VoIP media delivery—key knowledge for the Cisco Certified Field Technician (CCT) exam.
Table of Contents
Question
Which protocol is responsible for carrying real-time voice traffic in Cisco collaboration networks?
A. RTP
B. SIP
C. H.323
D. FTP
E. ICMP
Answer
A. RTP
Explanation
RTP (Real-time Transport Protocol) is responsible for delivering voice and video streams in VoIP communications.
RTP (Real-time Transport Protocol) is responsible for carrying real-time voice traffic in Cisco collaboration networks. RTP is designed to deliver audio and video streams over IP networks, providing essential features such as payload type identification, sequence numbering, timestamping, and delivery monitoring. These features enable the reliable transport and reconstruction of real-time media, even if packets arrive out of order or experience minor losses.
While protocols like SIP and H.323 handle call setup, signaling, and teardown, RTP is specifically used to transport the actual voice (and video) media between endpoints during a call.
RTP is the protocol that transports real-time audio and video streams in VoIP communications, ensuring media is delivered efficiently and with minimal delay across the network.
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