Early Media SRTP issue

I have a sip trunk and I receive a call to a webrtc sip peer.

Why early media doesn’t work with SRTP?

When the function sip_get_rtp_peer is called inside ast_channel_early_bridge it has an if where prohibits the Glue.

if (p->srtp) {
	res = AST_RTP_GLUE_RESULT_FORBID;
}

Asterisk doesn’t forward through the information required to do such a thing. To do so would require major work.