RTP
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. It is widely used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, and web-based push-to-talk features.
Key Concepts of RTP
Payload Type: Identifies the format of the RTP payload and determines its interpretation by the application.
Sequence Number: Used to detect packet loss and to restore packet sequence.
Timestamp: Used to synchronize and play back media streams.
SSRC (Synchronization Source Identifier): Identifies the source of a stream. Each stream in a session has a unique SSRC.
RTP Core Features
Real-time Transmission: Designed for real-time data transmission, such as audio and video.
Synchronization: Provides mechanisms for synchronizing different media streams.
Scalability: Can be used in both unicast and multicast network configurations.
Extensibility: Can be extended with additional features through RTP profiles and payload formats.
Example: Sending and Receiving RTP Packets Using Python's pylibsrtp Library
Here is an example of sending and receiving RTP packets using Python's pylibsrtp library:
Sender
Receiver
Relevant Switches and Parameters
Common RTP Packet Fields
Version: The version of RTP, typically set to 2.
Payload Type (PT): Indicates the format of the RTP payload.
Sequence Number: Increments by one for each RTP packet sent.
Timestamp: Reflects the sampling instant of the first byte in the RTP data packet.
SSRC: Identifies the source of the RTP stream.
Understanding RTP and its associated features and methods is crucial for implementing and troubleshooting real-time media streaming services.