When it comes to Voice over IP (VoIP), the transport layer protocols
used for audio and video payload are crucial for ensuring a reliable and
high-quality communication experience.
The two main transport layer protocols used for VoIP audio and video
payload are Real-Time Transport Protocol (RTP) and User Datagram Protocol
(UDP). RTP is responsible for the transmission of real-time audio and video
data, while UDP is used for the transport of the RTP packets.
RTP is designed to provide end-to-end delivery of real-time audio and video
data over IP networks. It works on the Application layer of the OSI model
and is responsible for the delivery of the audio and video data from the
source to the destination. RTP also provides a timestamp for each packet to
allow for proper synchronization of the data at the receiver’s end.
Additionally, RTP can be extended with different types of header extensions
to support various media types.
Also, Check: Top 50+ Cisco VoIP IPT and above Level Interview Questions and Answers
UDP, on the other hand, is a connectionless transport protocol that
provides a lightweight way of transmitting data without the overhead of
establishing a connection first, making it ideal for real-time communication
like VoIP. UDP provides a best-effort delivery mechanism, which means that
it does not guarantee delivery of packets or provide any error correction.
However, this lack of error correction is not a significant issue for
real-time communication since the data is constantly changing and can be
discarded if it arrives too late.
In summary, RTP and UDP are the primary transport layer protocols used for
VoIP audio and video payload. RTP is responsible for the delivery of the
real-time data, while UDP is used for the transport of the RTP packets.
Together, they provide a reliable and efficient way of transmitting
high-quality voice and video data over IP networks.