Creating an Audio/Video conferencing system with VOIP or WebRTC
Create an audio/video communications system which functions on desktop or mobile apps. Handle calls through WebRTC or VOIP using a SIP server. Achieve maximum Audio/Video quality by utilizing codecs that offer excellent quality/bandwidth tradeoffs.
Depending on your infrastructure stack, you can support VP8/H.264 and VP9/H.265 video calling, and G722/G729/G711 audio codecs for audio calling. The system can be further customized to handle other audio/video codecs if necessary.
Using open VPN tunneling, you will provide easier and faster connections to your users even in restricted areas and bypass server blocks by an ISP.
Monetize calls by integrating with backend software like SoftSwitch.
Make a video call or conference with multiple participants. Video can be recorded for future reference.
Make audio calls with clear audio quality and fast connections, either one on one or in a group. Audio can be recorded for future reference.
Provide instant messaging and archive file sharing features within your app.
Android/iOS or browser based client, with WebRTC Media server or SIP server to handle calls.
Utilize VPN tunneling to work around ISP level blocks when needed to provide stable connection.
Monetize your app in various ways using ads or with call charges. Rates can vary by geographical region.
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