Use UDP to transport.
Used in internet telephony, Voice over IP (VoIP) and video telecommunication. It can be used for one-on-one calls (unicast) or in one-to-many conferences (multicast).
It favors the quick delivery of packets over ensuring all data is received.
- RTP packets include:
- a sequence number, used to detect lost packets;
- payload identification, which describes the specific media codec;
- frame indication, which marks the beginning and end of each IP frame;
- source identification, which identifies the originator of the frame;
- intramedia synchronization, which uses timestamps to detect different delay jitter within a single stream and compensate for it.
RTP Control Protocol (RTCP)
Use UDP
Used in conjunction with RTP to send information back to the sender about the media stream.
primarily used for the client to send quality of service (QoS) data, such as jitter, packet loss and round-trip time (RTT). The server may use this information to switch to a different codec or stream quality. This data can also be used for control signaling or to collect information about the participants when many are connected to the stream.
- 服务质量的监视与反馈;
- 媒体间的同步;
- 播组中成员的标识。