My First FreeSWITCH Lab After Getting Comfortable with Asterisk

Trying FreeSWITCH after spending time with Asterisk was useful because it exposed which parts of my VoIP knowledge were conceptual and which parts were tied to one product. Registration, SIP dialogs, SDP and RTP behaved according to the same protocols, but the configuration layout, command-line tools and terminology were different enough that copied Asterisk habits did not carry over directly.

The first goal was deliberately small: register two SIP endpoints and complete an internal call. FreeSWITCH already had a directory and dialplan structure, so I focused on understanding where user credentials lived, which SIP profile accepted the registration, and how a dialed extension moved into the dialplan. The fs_cli console became the equivalent of having a live window into the server, while Sofia commands exposed SIP profile and registration state.

What stood out was the idea of SIP profiles as distinct listening and routing contexts. Instead of assuming there was one global SIP service, I had to pay attention to which profile an endpoint reached, which address and port that profile bound to, and which context the call entered. That reinforced a lesson from Asterisk: signaling behavior depends heavily on where a request enters the system, not only on the number being dialed.

The media path also gave me a useful comparison. SDP still told the endpoints where RTP should go, and the same NAT and firewall mistakes produced the same classes of failure. When a call answered with bad audio, switching PBX software did not change the troubleshooting method. I still checked the offer and answer, the media addresses, the negotiated codec and the actual UDP flow. The product changed; the protocol evidence did not.

The lab was valuable because it removed some accidental certainty. I had begun to associate 'how VoIP works' with 'how Asterisk is configured'. FreeSWITCH broke that association. A PBX is an implementation of signaling, routing and media behavior, not the definition of those protocols. Learning a second platform made the common pieces easier to see and prepared me for SIP proxies where call routing can happen without behaving like a traditional PBX at all.