How can I send tcp audio packet?

I came from WebRTC to SIP. In the case of webrtc, turn tcp is possible for this purpose, but I can’t find the any solution or configuration in the sip of asterisk.
tcp is used in the strict company firewall which do not allow udp.

Asterisk does not support TURN TCP.

Thanks for the reply.
But enabling the ice flag in the client side like pjsip or Zoiper can use the tcp turn server?
In this case, only turn udp can work?

I also found the option in pjsip.conf

ICE is supported, but any ICE candidates that use TCP will not be considered and Asterisk itself can not be configured for it.

HI jcolp. Thanks for the reply. I got it. Then, I should use the SIP NAT traversal without ICE in the case that no need for Chrome WebRTC Browser <=> Asterisk.