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
AudioStream
s. The
AudioGroup
mixes all the
AudioStream
s 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
AudioStream
s.
|
AudioGroup |
An AudioGroup is an audio hub for the speaker, the microphone, and
AudioStream
s.
|
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). |