- From: Christopher Allan Webber <cwebber@dustycloud.org>
- Date: Sat, 21 Oct 2017 18:25:35 -0500
- To: Manu Sporny <msporny@digitalbazaar.com>
- Cc: public-credentials@w3.org
Manu Sporny writes: > On 10/21/2017 01:09 PM, Kim Hamilton Duffy wrote: >> I wanted to ask for a volunteer to look into improving our collective >> dial-in experience on the Tuesday calls. The concerns: - phone line >> has limited number of connections, and costs money for Digital >> Bazaar. We'd like to avoid this. - linphone: variety of issues -- >> possibly just user experience, but not working for a number of us: >> dropped connections, inability to use >> >> This could be resolved by suggestions for, per David's request, >> "idiot-proof" sip clients. Can anyone volunteer to look into this? I >> personally stopped using Linphone due to dropped connections and >> haven't had time to follow up. > > We could try to enable WebRTC on our conference system, but would need > someone to help us out with the Web-side of this. Someone like the > editor of the WebRTC spec *cough*Dan Burnett*cough* might know a thing > or two about this. WebRTC is super cool but I've found that WebRTC is also a big challenge sometimes to get people to be able to connect to... If you're willing to go VOIP-only, I still recommend Mumble. Again, it's what we use in the SocialCG group, and it has worked out very well for us so far. Clients for all major operating systems, and barely any at all maintenance work to get it going. You won't have a way for people to dial in via traditional phone lines though. I'd be willing to test-host a single CCG call if it would help for giving a trial run. But I know you all have put a lot of effort into your Asterisk setup so I totally get if you'd prefer not to! - Chris
Received on Sunday, 22 October 2017 13:03:17 UTC