Since version 0.1.67 (2019.05) "Troubleshooting" function is part of "Help" menu and contains simple checks for most of the problems listed below. This would be recommended first step in case of any problems or for any new installation - reminding of trivial things like missing microphone or missing audio devices with remote desktop (forcing to switch audio modules to "Null" or wave file).

No registration?

Call interrupted immediately after connecting?

Call dropped immediately after being initialized

Call confirmed but no audio in one or both directions

Often caused by ACK message not being received or RTP stream being directed to wrong address.
In general tSIP is not well equipped for NAT traversing (e.g. ICE is not available) as this kind of problems is not expected with commercial VoIP operators (as their devices are supposed to be accessible from public addresses, be correctly configured and most of the times are prepared for situations when second party advertises local or incorrect public adresses) or for communication withing single network.
In not too rare cases router - in particular ALG = Application Level Gateway feature - may be responsible. It is supposed to help with NAT traversing but oversimplified (e.g. based on text substitution) or plainly incorrect implementations may often block communication. If setting related to this function cannot be found in router configuration but its presence is suspected then changing SIP listen port(s) to other than 5060 and 5061 might help.

Failed to init application

Possible crash related to AGC / audio preprocessor

Rare crash (sqrt domain error) was reported when using AGC on Linux/Wine. I don't know if this is specific to Wine or rather some audio signal pattern and I wasn't able to reproduce this. While I haven't removed AGC from application (leaving it for test purposes) I'm not recommending using this option as it's usability is very debatable (Windows 7 has built-in AGC enabled by default anyway, also AGC can often degrade audio quality by amplifying noise).


2018.01. I've received report with problem with one (German?) cloud-based virtual PABX operator. This operator allows to register multiple phones/softphones to single account but apparently after tSIP was registering other devices from same account were losing registration / not received incoming calls. Seemingly this was fixed by filling "Contact" in tSIP account settings with same value as user name (if not filled tSIP fills Contact with semi-random value which may look weird but shouldn't cause problems) and setting local port (binding) to 5060 (despite that softphone was behind NAT that was changing source port anyway).

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