RTP (Real-time Transport Protocol) and RTCP (Real-time Transport Control Protocol) are closely related protocols designed to handle the transmission of audio and video data over IP networks. They are commonly used in streaming media systems, VoIP (Voice over IP) services, and other applications where low-latency and real-time communication are essential.

RTP (Real-time Transport Protocol):

Function: RTP is responsible for the end-to-end transport of real-time data. It carries the media data, like audio or video, between the sender and receiver.

Features:

  • Sequence Numbers: Each RTP packet is assigned a sequence number, enabling the receiver to detect packet loss and reorder out-of-sequence packets.
  • Timestamp: RTP packets have timestamps, which allow the receiver to play the media in the correct timing and deal with jitter (variation in packet arrival time).
  • Payload Type Identification: Indicates the format of the contained media, such as PCM audio, MPEG video, etc.
  • SSRC and CSRC: These are identifiers for the synchronization source and contributing sources, useful in mixers and translators in multimedia conferences.

Applications: Used in video conferencing, streaming media, and online games.

RTCP (Real-time Transport Control Protocol):

Function: While RTP carries the media data, RTCP works alongside it to provide feedback about the quality of the data distribution. It doesn’t transport media data but instead provides statistics and control information.

Features:

  • Sender Reports: These provide statistics about the number of RTP packets sent, the number of bytes sent, and more.
  • Receiver Reports: These provide statistics on the data received, including packet loss, jitter, etc.
  • Source Description Items: These provide information about the sources, such as the name, email, phone, and location.
  • Goodbye (BYE) Signal: Indicates the end of participation in a session.
  • Application-defined: Allows application-specific functions.

Bandwidth: RTCP’s traffic is generally kept below 5% of the session bandwidth, to ensure that the control traffic doesn’t overshadow the media traffic.

Use Together:
RTP and RTCP are usually used in tandem. RTP handles the real-time transfer of media data, while RTCP provides control and quality feedback. This combination allows for the optimization of streaming quality in real-time, making adjustments based on network conditions, packet loss, and other factors.

In systems like VoIP, when you experience issues like dropped audio or poor call quality, it’s the RTCP’s feedback mechanism that allows the system to adjust and try to improve the call quality by changing codecs, reducing bitrate, or taking other remedial actions.