SIP (Session Initiation Protocol)
SIP, or Session Initiation Protocol, is a signaling protocol used to start, modify, route, and end real-time communication sessions over internet protocol networks. In telephony, it coordinates the call while separate media protocols usually carry the actual audio.
How SIP works
SIP participants exchange requests and responses. An INVITE requests a session, provisional and final responses report progress or acceptance, an ACK confirms a successful setup, and a BYE ends an established session. Registration can tell a SIP service where a user or endpoint is currently reachable. SIP URIs identify destinations in a form that resembles an email address.
The session description exchanged during setup helps the endpoints agree on media details such as addresses, ports, and codecs. Once the call is connected, audio commonly travels through RTP rather than inside the SIP messages. This separation is why successful signaling does not automatically prove that two-way audio will work: firewall rules, address translation, codec negotiation, or media routing can still fail.
SIP is also extensible. Methods such as REFER can request a transfer, while provider-specific headers or policies may influence identity and routing. Implementations do not all support every extension in the same way, so a working deployment depends on compatible configuration at both ends.
Why it matters for AI phone calls
SIP is a common bridge between an AI voice application and business telephony. It can route a phone number to an agent, allow an eligible agent to place an outbound call, or connect a call to a PBX or carrier. The configuration typically covers authentication, destination addresses, number formatting, permitted source addresses, codecs, and failure routes.
Troubleshooting should separate signaling from media. A rejected INVITE points toward routing, authentication, or policy. A connected call with silence in one direction points more often toward the media path. Keeping those layers distinct makes interoperability testing more useful.
In practice on ThunderPhone
ThunderPhone supports production numbers through supported direct VoIP connections or manual SIP configuration for any SIP trunk. Once imported numbers are verified, they support inbound and outbound calling. Inbound demo U.S. numbers are available for testing, but they are not intended for production and do not support outbound calling.