How to Implement Conference Call Feature Using JsSIP with FusionPBX/Freeswitch

[Subhan00]

Hi everyone,

I’m currently developing a SIP phone using the JsSIP library and would appreciate some guidance on implementing a conference call feature.

So far, I’ve successfully implemented the basic call controls such as mute, unmute, hold, unhold, and call transfer. I’m now looking to add support for conference calling.

I’m using FusionPBX with FreeSWITCH as the backend SIP server. Has anyone implemented conference calling with this setup using JsSIP? I’m specifically looking for advice or best practices on how to:

Merge multiple calls into a single conference

Manage audio streams appropriately

Handle SIP signaling and media negotiation

Use FreeSWITCH’s conference modules (if applicable)

Any examples, references, or tips would be really helpful!

Thanks in advance!

[ibc]

Don’t try to fast-track answers by name-dropping the library authors. It’s not clever, it’s just annoying. So don't do that again, please.

BTW nobody is gonna give you full consultancy services for free.

[Subhan00]

Got it — sorry if tagging came across the wrong way, that wasn’t my intention (removing it now). I really appreciate all the work that’s gone into JsSIP and was just hoping to learn from anyone who’s worked on a similar setup. If anyone has any pointers or direction on implementing conferencing, I’d be super grateful. Thanks!