Assuming that your VOIP phone is on remote site and you are connected to firewall through VPN(IKEv1 or IKEv2) connection. Your end terminal is able to reach SIP server on some port 5060,5061 or any other port and successfully registers itself with SIP server.
Then, I think you do not need to explicitly open port for SIP and RTP messages as ASA will automatically create necessary pinholes.