If you have a better solution or just suggestion to this workaround, let me know Preconditions: List item Download and extract swig (swigwin 4.0.0) Download and install JDK Download and install Python (Python 3.7.3 x64) and set at custom installation: tick 'Add Python to environment variables'. Is there a CLI command to show the codec in an active chanpjsip connection? If I use pjsip show channel I get a nice listing of many useful values for the active channel, but there doesn't seem to be an easy way of see&hellip. GitHub Gist: instantly share code, notes, and snippets.
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: [email protected], [email protected],32kHz, [email protected],24kHz, [email protected]
Medium quality: [email protected], [email protected] (PCMU and PCMA), [email protected]
Enhanced quality: AMR, [email protected]
Low quality: [email protected], [email protected], [email protected], [email protected], SIL[email protected], GSM
Codecs without compression: Linear [email protected],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 [email protected] 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.
PJMEDIA_CODEC_VPXFunctions
pjmedia_codec_vpx_vid_init(pjmedia_vid_codec_mgr *mgr, pj_pool_factory *pf)¶Initialize and register VPX codec factory.
PJ_SUCCESS on success.
mgr: The video codec manager instance where this codec will be registered to. Specify NULL to use default instance (in that case, an instance of video codec manager must have been created beforehand).pf: Pool factory.
pjmedia_codec_vpx_vid_deinit(void)¶
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Unregister VPX video codecs factory from the video codec manager and deinitialize the codec library.
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PJ_SUCCESS on success.