How Do I Know Whether a FreeSWITCH Problem Is SIP, Dialplan or RTP?
Use a diagnostic ladder to decide whether a broken call is a registration, INVITE, routing, bridge, RTP or timeout problem before changing configuration.
FreeSWITCH learning series · Part 14 of 30
Use a diagnostic ladder to decide whether a broken call is a registration, INVITE, routing, bridge, RTP or timeout problem before changing configuration.
What this question really means in a working FreeSWITCH system
Use the last layer that demonstrably worked. No REGISTER response is transport/profile. INVITE arrives but no matching route is dialplan. A bridge generates an outbound INVITE that is rejected is trunk/signaling. Both legs answer but audio fails is media/RTP. A call dies around a repeatable timer points toward session/NAT/signaling refresh behavior. This ladder prevents configuration thrashing.
Start with the mental model
A failed call is easier to diagnose when you ask three questions in order: did SIP establish the intended call legs, did the dialplan choose the intended route, and did RTP flow both ways? This prevents configuration roulette.
| Layer | Typical failure | Best first question |
|---|---|---|
| SIP registration | Phone absent | Did REGISTER reach the correct profile? |
| SIP call setup | Immediate reject/timeout | Who returned the response? |
| Dialplan | Wrong/no route | What context and destination were evaluated? |
| B-leg | Destination unavailable | Was the correct user/gateway dialed? |
| RTP | Silence/one-way audio | Where did SDP tell each side to send media? |
Work through it from zero
Record source, destination, timestamp, ringing/answer behaviour and what each party heard.
Check FreeSWITCH is healthy and the intended Sofia profile is active.
Follow REGISTER or INVITE response sequences and note who generated the failure.
Capture actual context, destination and matched extension.
Verify user contact or gateway and the B-leg created.
Only after answer, compare SDP and packet flow in both directions.
What beginners usually confuse
- Making several changes between tests.
- Calling a carrier problem “FreeSWITCH is down”.
- Treating a 200 OK as proof of two-way media.
How to know you are actually finished
- Use one diagnostic worksheet per incident.
- Sanitize traces before sharing.
- Restore known-good configuration if the failure began with a recent controlled change.
How to use this in a real implementation
For this FreeSWITCH task, use a bounded test built around this objective: use a diagnostic ladder to decide whether a broken call is a registration, invite, routing, bridge, rtp or timeo…. Observe one state change at a time, save the evidence, and only then move to the next layer.
- Capture the exact runtime evidence related to Name one exact symptom before changing the next layer.
- Keep a known-good test number/endpoint and repeat the same call after each configuration change.
- Record SIP response codes, context/destination decisions and media observations separately; they answer different questions.
- If you cannot explain which file/module owns the behavior, stop and locate that ownership before editing more configuration.
Continue from here
After this article: use the next link that matches the unresolved part of use a diagnostic ladder to decide whether a broken call is a registration, invite, routing, bridge, rtp or timeo…. Start the FreeSWITCH learning series · Continue into real-time AI media streaming
Questions a careful reader usually asks next
Should I change several FreeSWITCH files at once?
Not while learning or troubleshooting. Prove name one exact symptom first, then change one layer and repeat the same test so you know what caused the new behavior.
Is a successful CLI command enough proof?
No. For use a diagnostic ladder to decide whether a broken call is a registration, invite, routing, bridge, rtp or timeo…, confirm the live SIP, dialplan or media behavior that the command was intended to affect. A parser or CLI success only proves the command was accepted, not that the call path now behaves correctly.
Where should business logic live?
For use a diagnostic ladder to decide whether a broken call is a registration, invite, routing, bridge, rtp or timeo…, keep low-level signaling/media truth in FreeSWITCH, while customer, campaign and business state stays in the application layer unless the telephony engine genuinely needs it for call execution.
References and further reading
Protocol and configuration facts for How Do I Know Whether A FreeSWITCH Problem Is Sip Dialplan Or Rtp are grounded in the current FreeSWITCH Users Manual and, where Asterisk is compared, Asterisk's official documentation. Community tutorials are included only as credited learning aids.
- FreeSWITCH Users Manual — authoritative project documentation for the FreeSWITCH behavior/configuration referenced above.
- FreeSWITCH Getting Started — authoritative project documentation for the FreeSWITCH behavior/configuration referenced above.
- FreeSWITCH — SIP Profiles with Sofia — authoritative project documentation for the FreeSWITCH behavior/configuration referenced above.
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