Frequently asked questions

Q: I launch MicroSIP but nothing happens.

A: Check for MicroSIP icon in system tray.

Q: How to setup account?

A: Right click on MicroSIP icon in system tray (near clock:).

Q: How to add contact?

A: Right click on blank white area in Conacts tab.

Q: How to specify different SIP port?

A: If you use SIP proxy - append ":port" to proxy only. If not, append ":port" to "SIP server" AND "Domain". Format: "proxy:port" OR ("server:port" AND "domain:port").

Q: How to achieve best voice quality?

A: Voice quality depends on audio codec that was selected in negotiation for current call session. In extended mode MicroSIP will show you, what codec was selected for session.

Codecs by quality:

High quality: Opus@24kHz, speex@16,32kHz, SILK@16,24kHz, G.722@16kHz
Medium quality: SILK@12kHz, G.711@8kHz (PCMU and PCMA), AMR-WB@16kHz
Enhanced quality: AMR, iLBC@8kHz
Low quality: GSM@8kHz, G.723@8kHz, G.729@8kHz, speex@8kHz, SILK@8kHz, GSM

Codecs without compression: Linear PCM@8,16,44kHz

Notice 1. VoIP provider can route your voice session to external destination through low-quality audio codec. In this case you cannot achieve high quality. Key to quality lays in hands of your VoIP provider.
Notice 2. VoIP provider can limit set of allowed codecs.
Notice 3. For incoming calls use force codec option in MicroSIP settings.

Finally try Speex@16kHz between two MicroSIPs. You can call by local IP, to exclude SIP server restrictions. You'll know what means high quality.

Sound latency caused by set of dynamic buffers on the path of audio. Average value - 200 ms (one way). There is no way to reduce latency significantly.

Q: I use MicroSIP without registration on SIP server. How to specify address of my SIP gateway?

A: You can fill "Domain" in account page OR enter number in format <number>@<gateway>.

Q: How to set up MicroSIP for point to point without a SIP server between 2 laptops?

A: Minimum what need to do - install microisp. Now you can make and receive calls. To make call enter number in format: "sip:192.168.1.33" or just "192.168.1.33", where "192.168.1.33" - IP address of callee. If you want make IP-to-IP calls simultaneously with active SIP account, additionaly you must enable local account in Settings.

Q: How to do an attended transfer in Microsip?

A: Disable single call mode in the settings. Make or receive a first call. Make a second call. After that, the Attented Transfer button in the second window will be enabled.

Please note that a PBX may not support call transfer using standard SIP commands.

Call transfer (blind/attended) is also possible using feature codes (see PBX documentation), and in this case it can also work in the single call mode.