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).
- make sure register expires is set to non-zero; zero value as registration expiration serves as special value to work without registration
- if there are multiple network interfaces or adresses try binding to single one, using either its IP address if it's fixed or device GUID (as displayed in log); changing NIC priorities from control panel is another option
- check log
Call interrupted immediately after connecting?
- missing audio input or output device (e.g. no microphone plugged to jack with autosensing)
- if using wave file as source: incompatible file format (required wave, S16_LE, either 8ksps or 16ksps depending on codec used) or wave file is missing
Call dropped immediately after being initialized
- see log to check if disconnection cause is local (e.g. second party answered but audio device is missing, no microphone connected) or related to both call parties (e.g. SIP/488 reply received = no compatible codecs selected, SIP/403 = probably wrong user/password)
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
- bind port for SIP is busy, i.e. other application if listening on the same port as tSIP tries to
- change tSIP bind port (or remove binding thus using automatic port)
- or: change bind port in other application - netstat command may help
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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