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a voip demo base on google webrtc
This project forked from dannyfly/voipclient
a voip demo base on google webrtc
一个基于webrtc的音频通话的android客户端。 (没有server的部分) 此Demo为语音通话的客户端。 需要在两台Android设备上运行此Demo, 并且前提是两台设备连接同一路由器 (因为没有搭配STUN和TURN服务器,因此只能手动输入对方IP,以模拟P2P连接的情景) 然后输入对方的内网IP地址 譬如A输入B的地址:192.168.1.100 B输入A的地址: 192.168.1.101 点击“Start Call”按钮,即可实现双方的语音通话。 Demo上可以选择语音Codec,是否开启回声消除,噪声抑制,自动增益等功能。 Demo主要关注功能的实现与延迟的测量与优化,因此可能在操作与实现上比较粗糙,请多包涵。
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