Provides APIs for RTP (Real-time Transport Protocol), allowing applications to manage on-demand or interactive data streaming. In particular, apps that provide VOIP, push-to-talk, conferencing, and audio streaming can use these APIs to initiate sessions and transmit or receive data streams over any available network.
To support audio conferencing and similar usages, you need to instantiate two classes as endpoints for the stream:
AudioStream specifies a remote endpoint and consists of network mapping
and a configured AudioCodec.AudioGroup represents the local endpoint for one or more AudioStreams. The AudioGroup mixes all the AudioStreams and optionally interacts with the device speaker and the microphone at
the same time.The simplest usage involves a single remote endpoint and local endpoint. For more complex usages,
refer to the limitations described for AudioGroup.
Note: To use the RTP APIs, you must request the INTERNET and RECORD_AUDIO permissions in your manifest file.
| AudioCodec | This class defines a collection of audio codecs to be used with
AudioStreams. |
| AudioGroup | An AudioGroup is an audio hub for the speaker, the microphone, and
AudioStreams. |
| AudioStream | An AudioStream is a RtpStream which carrys audio payloads over
Real-time Transport Protocol (RTP). |
| RtpStream | RtpStream represents the base class of streams which send and receive network packets with media payloads over Real-time Transport Protocol (RTP). |