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:
AudioStreamspecifies a remote endpoint and consists of network mapping and a configured
AudioGrouprepresents the local endpoint for one or more
AudioGroupmixes 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
Note: To use the RTP APIs, you must request the
RECORD_AUDIO permissions in your manifest file.
|AudioCodec||This class defines a collection of audio codecs to be used with
|AudioGroup||An AudioGroup is an audio hub for the speaker, the microphone, and
|AudioStream||An AudioStream is a
|RtpStream||RtpStream represents the base class of streams which send and receive network packets with media payloads over Real-time Transport Protocol (RTP).|