Is there currently a way to setup a VOIP provider over a standard protocol like SIP-T or through PRI, etc, and add ability to use the telephone systems for the voice portion of a call or group call?
we have looked into the possibility of adding phone dial in by the means of a webrtc to sip gateway for customer requests in the past and would be open to do so.
If you need this feature for a customer project please get in touch with our sales team.
Some general remarks:
- even having the standby ability to dial into a room through sip will effectively break end to end encryption since you then need to decrypt the webrtc connection for the phone to join
- if its for the “microphone and camera are so hard on a pc/laptop” then you can already use Meet on a (smart)phone. If you are concerned about the amount of data transmitted you can switch your end into audio only mode.
the idea is to have a local meeting server that is an alternative to solutions like webex, zoom, etc. STUN/TURN services present locally. SIP-T device present locally - potentially on the TURN server.
This is a technical discussion, not a sales conversation.
We recently added support for “bridge” type connections to our signalling server (will be used for a custom SFU). This same type of connection could be technically used to connect SIP systems. But unless there is a customer project that would make this viable to invest development capacities on this, this is nothing that we will be working on in the foreseeable future.