我一直被困在同样的问题上.

如果你可以在没有android sip api的情况下制作它,你可以查看rtp api,它为你提供了一些较低级别的工具来制作P2P VOIP应用程序,而无需服务器.

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.

对应的是你必须编写自己的设备发现协议,以便知道audiostream peer使用的端口,如answer中所述.

如果你只打算进行一对一的谈话,但如果你想进行一对一的谈话,那么问题就不那么难了.

对于一对一的会话,会议主持人必须为他想要呼叫的每个远程设备实现n audiostream.每个远程对等体只有一个音频流链接到主机音频流之一.