Mayuresh <mayuresh%acm.org@localhost> writes: > On Tue, Jul 28, 2020 at 08:26:44AM -0400, Greg Troxel wrote: >> https://desktop.jitsi.org/Main/Download.html >> >> Perhaps you can file a bug with upstream that they don't have a source >> download and that their github repo doesn't have tags, and maybe even >> that it appears unmaintained. > > I noticed, NetBSDers have logged it already. Then it would be helfpul to put a link the bug report in the wip package. And maybe even fix the package. > A few weeks back I had tried tinkering with asterisk and WebRTC with jssip > but could not get a good reliable solution. It was difficult to figure out > whose fault (i.e. varying quality of audio/video) was - browser, jssip > layer, asterisk? Someone had advised jitsi that time on the thread. > > Looking for others' experiences with any of the above or any other stacks, > preferably involving NetBSD server. I can't tell what you are even trying to do. For SIP clients, there is (not in pkgsrc) linphone baresip For PBX, there is asterisk. For SIP router, there is freeswitch and kamailo.
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