I am working on adding a new feature in a dialer project.
Currently this is how it works.
There are agents who whose physical phone (landline or cellphone) is
connected to the internal router (through a provider like verizon) and
waiting for a call from some of the external customers/debtors.
We have some servers acting as routers and engines. Engines are the servers
talking to the provider's servers through SIP and RTP. Routers are the ones
which manages agents and balances the loads in themselves.
The new requirement is to make agent able to connect to the routers tough
their laptop/computer audio devices (mic/speaker). I m planning to use
libjitsi to achieve this.
Does anyone has more suggestions on this? Does it also require using ICE4?
Any help in this will be great as I am quite new to SIP.