[Freeswitch-dev] Connect to SIP Trunking

Hi all, I'm new to freeswitch and want to learn SIP development, I have a question: what's protocol is using for the SIP PBX(freeswitch, asterisk, sipx) connect to SIP Trunking? In freeswitch source code, which part source code is for connect to SIP trunking ? Thank you in advance.