Episode Details

SIP doesn't carry audio — it negotiates sessions, and RTP does the rest. That separation is what makes one-way paging possible: a SIP speaker is just an endpoint with a phone number, no mic, no dial pad, that answers an INVITE and plays whatever arrives. This episode unpacks the fork between unicast paging (thirty speakers, thirty simultaneous calls) and multicast paging (one SIP call, one RTP stream, five hundred listeners), plus the Asterisk MulticastRTP dial strings, FreeSWITCH-based open source approaches, and the layer 2 versus layer 3 networking that decides whether any of it works across VLANs. Episode #515219 — open it directly at myweirdprompts.com/515219