You might find a few people. :) If you want "the pros" then go to www.freeswitchsolutions.com or email support@freeswitch.org. The guys who wrote FreeSWITCH run that and they have a network of professional FS consultants who can do all sorts of fun things for you! -MC On Tue, Jan 4, 2011 at 11:28 AM, Siobhan Hamilton < siobhan.pluggedin@gmail.com> wrote:
Michael, thanks very much for your response - and as an aside to anyone out there - we are looking to hire someone to build a server with these specs and get it running efficiently for us, if anyone is qualified/interested...
On Tue, Jan 4, 2011 at 2:20 PM, Michael Collins <msc@freeswitch.org>wrote:
Answers inline..
On Tue, Jan 4, 2011 at 7:03 AM, Siobhan Hamilton < siobhan@pluggedin-tech.com> wrote:
My apologies for any duplication; I have tried to post this question several times to no avail (at least from my end)....
Apologies... non-list members are automatically moderated.
My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Freeswitch (the other option being Asterisk). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Freeswitch can do or not.
We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves
Yes. In fact, this is the standard behavior of FS conferences
- Within a conference, give users the ability to mute and/or deaf individual users (which I know can already be done with the "relate" command, so that's solved, pretty much)
Yes, you are correct. The challenge for you will be to create the external process that manages users and permissions, so that certain users can deaf/mute/kick other users and also process the DTMFs that users dial while in the conference.
- Give users the ability to enter a "whisper" mode with another user - where they are holding a private conversation that can only be heard by the two of them
Yes, this can also be handled with the relate command. The challenge will be making sure that the two parties both know that they are in whisper mode and you have to decide if you are going to mix the audio from the rest of the conference into the whispering parties' private chat. In any case this is definitely doable with a little work in an external control script/program.
- Allow users to be in two conferences at once; the user would most likely
have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily
This is an interesting one. I'm sure it can be done, I just don't know the most elegant way of doing it. A brute-force way of doing it would be to let the user be in his own "personal" conference and from there have that personal conference make outbound calls to the other conferences. From there you could use the mute and/or relate commands to control the flow of audio. Again, the big challenge there would be giving the user control over his audio and finding a way to give the user audible indications as to which conference his audio is flowing to, if at all. FreeSWITCH absolutely has the tools to do this. Its conference app is probably the most versatile in the telecom world - OSS or proprietary. Coupled with the event socket you can do al sorts of interesting things, limited only by your imagination and programming skills.
-MC
Could anyone advise me on whether Freeswitch can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen!
Thanks, Siobhan Hamilton
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