I use WebRTC with the JsSip library as a replacement end point for IP phones and softphones.
I tried making incoming and outgoing calls using WebRTC and it went well.
I tried making an originate call action via AMI using WebRTC as a replacement endpoint for the softphone but it failed and response in asterisk CLI is just 'DTLS ECDH Initialized (automatic), faster PFS enabled, if use a Softphone (xlite), and do an action originate call via it, it works fine.
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If using softphone (e.g X-lite), I can control via AMI for example Originate call.
But if using WebRTC can we control via AMI with the same case which is originate call?
Assuming WebRTC replaces softphone.
There is nothing to learn because there is no difference, only in what you pass in. You haven’t provided any logging or further details so it can’t be stated really what is going on.