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I want to do one-to-one voice chatting while broadcasting.

I use HLS as a broadcast protocol.

It is because all our CDNs support HTTP delivery.

 

I heard that many implementations don't support AAC audio over WebRTC.

Opus is one of two voice codecs selected as mandatory to implement in WebRTC.

But, Opus inside HLS is not really a standardized payload.

 

1. Does your WebRTC SDK support AAC audio?

2. If no, can I send RTMP(or something for broadcasting) and WebRTC(ono-to-one two-way) from one source(maybe microphone)?

 

My user platforms are android and iOS.

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