Hi all, New PJSIP user here. I have a project where I have to create a gateway that will interface a group of intercoms to a SIP server as SIP clients. As a new user of PJSIP, I've been exploring its capabilities and believe it's the right tool for this project. I'm eager to share my idea and would greatly appreciate your constructive feedback. * I have a group of intercoms where I will use (PJ)sockets to connect to and stream audio packets to and from. * I have one piece of hardware running Linux/PJSIP. This hardware will: * Make multiple concurrent SIP registrations to an SIP Server (one for each intercom) - I have modified the simple_pjsua.c sample and have this part working as a test. * Create 4 NULL Sound ports (null_port.c) - here is where I will write code to interface PJSIP to the intercom devices. * Have these four ports connected to a conference bridge ready to be used When a call comes in via the SIP server, the app grabs a free NULL sound port, passes the address dialled information to it, and establishes a connection between the SIP server and the intercom device. The conference bridge is then used to connect the incoming call to the NULL sound port. * There will be a concurrent channel limit based on the number of NULL sound ports I can instantiate. * There will also be a limit on how many concurrent SIP registrations a single PJSIP instance can maintain. I am about to write a NULL sound port for my next test, so if anyone has any pointers or warnings, they would be greatly appreciated. Kindest Regards Billy