The way that low level network I/O works (and this a low level failure), is that the domain name is looked up to get an IP address, and the IP address is then used for the low level operations. For UDP, the IP address can be specified for each request, or can be specified at the beginning of a session. I can’t remember which Asterisk. uses. For TCP it can only be specified at the beginning of a session. Translating a domain name to an IP address is an expensive operation and not something you are going to do 50 times a second, even when using UDP for media.
Note if there is no audio in the first minute, it probably means there never was a valid route to the IP address agreed as the media destination.
I’m not sure if destination unreachable responses from routers or the end system will also produce this error, given that they are asynchronous.
Incidentally, although I haven’t used WebRTC, the general advice on this forum is that you should not even think of using WebRTC unless you have a thorough understanding of the underlying technologies. WebRTC is neither stable in time, not suitable as a plug and play technology for new users. It is fairly clear that you have only a sketchy knowledge of even basic TCP/IP.