You have 2 options for the backend:
a) home server - more complicated, depending on your technical specifications. What functionality do you need on the server? For example, Asterisk has an audio mixer for a conference call, while it lacks the presence of a user (you need to integrate with Openfire ). Kamailio / Openser support presence through SIMPLE , but they lack support for an audio mixer (you need to integrate with Asterisk or SEMS ). You can also use an all-in-one device, such as Sipwise , but this requires extensive knowledge if you want to configure something other than the default configuration,
A good role is that most servers require very little effort to make sip sound calls, so if you need it, it's best to set up your own server.
The advantages of this approach are that you have full control over the service, you can view backend logs, you can test use cases that you cannot use for public service. Disadvantages - this requires a significant resource for configuration, configuration and support. This is subjective and depends on the functionality that you need from the server.
b) public service is the easiest way. Depending on the capabilities of the service, it may be free or have a monthly fee. Most public SIP services allow audio calls, for everything else it depends on the service (calls to PSTN, video calls, conference calls / video calls, etc.). I would recommend sip2sip.info , but you can also easily find others. Pros, you can start using it immediately (after registration), and it doesn’t matter how you manage the service. Disadvantages - you do not have control over the service, you cannot see backend logs (which are vital if you are developing a multifunctional SIP software client).