asterisk sporadic rtp problem, need tcp or srtp to have it more solid

Max 300 euros.

Hi I have a webphone (transport = wss)
which uses a turn server (coturn)
to receive calls from asterisk.

If it happens that a client has a 'weak' router,
it may lose some rtp (port setup i guess)
and the client rings
but there is no audio
(mute call).

I'm looking for a way to strenghten the audio
in order to have it retrasmit, i guess
and have no more problems even with those 'weak' routers.

I don't know if it's possible to have rtp over tcp
or if srtp would help.

You are the expert.

The scope of the project is to fix those mute calls.