davidwaf added a comment - 23/Jun/11 3:08 PM Would like to enlist as a tester

Chris Hemmings added a comment - 23/Jun/11 3:38 PM Put me down for QC, any web UI stuff, I can help out. One thing that would be a great feature is to be able to authenticate that the client has things setup correctly to make a 'call'. To make sure they have their speakers/headphones working correctly, one could use a captcha type of system by speaking a word and asking the client to confirm what the word is by selecting something on screen. To confirm that the user has a working mic setup would be trickier. I guess you would have to go down the asr route and get them to say a specified word before the call connects. A question. Can you initiate a call from fs to a client that has a movie open but is not currently in a call? If so, an example Flex/AIR application could be kinda cool. I'm thinking of an integrated Ticket/Support system of some sort. /2cents

Kees Varekamp added a comment - 23/Jun/11 6:09 PM I can help out with QC and docs. No flash experience but can look into that too if necessary. my ideal rtmp client: - no ui - a js api - authentication - potentially a way to do rpc: js -> rtmp client -> fs events -> rtmp -> js client

Vadim F. added a comment - 24/Jun/11 1:25 PM No flash\flex experience but can help with testing.

Mitch Capper added a comment - 25/Jun/11 2:09 AM I would also suggest going the way of a minimal flash broker between mod_rtmp and javascript. The less actual logic in flash means the easier for anyone to change the javascript to have the client work how they like without having to recompile the flash. A proper flash lib could also be used by someone who wants a flash client and didnt want to use javascript but I would guess most customers would prefer the javascript client (as per conference.freeswitch style).

Kerem Erciyes added a comment - 25/Jun/11 7:02 AM As always I am happy with QC and testing, also can commit some developer time to create a Click2Call button that uses mod_rtmp as a backend.

Chris Hemmings added a comment - 26/Jun/11 2:37 PM With Mitch on this. Minimal flash broker is good, Lots of exposed ActionScript methods is good. JS is good. Buying Flash development software is expensive!

Anthony Minessale II added a comment - 27/Jun/11 10:56 AM We have some broker code we are preparing for release. What I am looking for here are people willing to work on coding some JS html we can add to tree. I do not just want to add the broker code and have everyone still working on it independently. Once we gather up some devs to commit to the project we can go forward.

Chris Hemmings added a comment - 27/Jun/11 11:32 AM Put me down for HTML/JS UI development.

Venky added a comment - 28/Jun/11 1:49 PM We could build something like http://phono.com/ 1. Does broking with minal flash and mostly js, 2. provides an sdk on which further apps can be built. Count me in for this for development.

Jack Loranger added a comment - 29/Jun/11 10:29 AM I am working on windows 2003 platform,Visual Studio 2008 can do QC and testing. HTML and javascript not much actionscript.

Seven Du added a comment - 30/Jun/11 7:50 PM Take me in. I have some experience on Flex( http://wiki.freeswitch.org/wiki/FsAir ) and i'm happy with js. And I do like to write code and test :)

rentmycoder added a comment - 02/Jul/11 3:25 AM I am also very interested in this project. I have great HTML/JS and Flex experience. I would be happy to develop great stuff!:)

Anthony Minessale II added a comment - 05/Jul/11 11:55 AM I checked the beginning demo code into clients/flex who wants to get started? We can hand out commit and whatnot. Visit IRC and/or email consulting@freeswitch.org with your existing jira account and desired login for GIT We'll start up #freeswitch-webclient on irc as well

Anthony Minessale II added a comment - 07/Jul/11 10:20 AM As we work on this let's keep in mind that we should try to keep the engine separated from the logic because if webrtc takes off it may obselete the flash portion of this and it can just be future proofed now with sound design.

rentmycoder added a comment - 07/Jul/11 2:06 PM I'm not against the separation, it's a good idea, and correct me if i'm wrong but, but webrtc has nothing to do with rtmp... I think it should implemented in FS in a defferent endpoint like mod_peerconnection and a new client like peerconnection_client. https://sites.google.com/site/webrtc/reference/getting-started Or do you suggest we should use the same Javascript API-s what webrtc uses?

Anthony Minessale II added a comment - 07/Jul/11 3:55 PM Yes I am only talking about the UI side. The only thing webrtc has to do with rtmp is that it will make it obsolete when its in all the browsers. It will supply more RTP like communication and direct access to sound devices and the UI will be more or less the same.

rentmycoder added a comment - 08/Jul/11 12:23 AM ok, I'll check the webrtc js API and see what would be more usefull, to use it as it is, or to create a genral one ont top of it... have you looked after it already? maybe we could add an experimental mod_webrtc for testing and generalisig purposes... for the first sight I see one difference on the UI side, in flash you can only graphically change access to mic/speaker for the given site, using general flash dialogs (and these dialogs has a minimum canvas size to be displayable) so flash content should be displayed at least in this case on the UI, however it can shown in a separate window using a different flash object... I see a lot of similarities on the server side regarding access restrictions which are very important if we are talking about system security and fraud prevention... -restrict telephony functions and events on a per profile or per user basis (dial,hangup,transfer,hold,api,etc...) -login system should have a mode where the authentication does not happen against SIP credetials, but separate username/password is defined in the user directory. this should be enabled or forced on a per profile and per user basis. These restrictions should apply for boths mod_rtmp and mod_webrtc, ur using different paramater prefixes? What do you think?

Anthony Minessale II added a comment - 15/Jul/11 3:23 AM the username/pass in the directory is not sip specific it's designed to be centralized but you could check for specific params that apply to the web phone user/pass and enforce them if they are present otherwise fall to the defaults. Pretty much you can do anything you want that's why the directory is an xml object. The restrictions make sense as well.

Anthony Minessale II added a comment - 16/Jul/11 2:56 PM Does everybody have an id for git etc? Has anybody done anything yet?

rentmycoder added a comment - 20/Jul/11 3:47 PM I've started experimenting with the code... and it works quite great, stable and well planned... For me the most important thing to improve to be able to use this mudule in production is: SECURITY... One thing is to restructure the clint side to move the rtmp callbacks into a new class, as is commented in the source code, but this is obvoius... But what is more important how to restrict features and access on the server side... Just one sample: How could I allow one user to access my default context, all other ones could allow only the public context to disallow them to dial out on my gateways to variuos high cost foreign phone numbers... I can use only one context in the profile,,, I suggest we could restrict the following thing on a per user basis: -rtmp login enabled/disabled -authentication: username/password different than sip credentials -registration required or not -which profile to use What do you think? Could we use RTMPS to secure the connection? RTMPS which is RTMPT over a secure SSL connection using HTTPS. The other important question: compatibility can we use http tunneling to overcome firewall issues, like rtmpt? RTMPT which is encapsulated within HTTP requests to traverse firewalls. Thanks, John

Stephen Dame added a comment - 20/Jul/11 5:04 PM Hello lm new to list and fs groups, i have the mod_ftmp demo running on my test server, interested in working on the project and helping to get the configs documented. i host and develop customizations around Bigbluebutton, which uses flex, freeswitch and red5. We have 30 freeswitch servers in production and growing daily. Interested in building a conference monitor similar to the one used on the weekly fs conf calls. regards, Stephen

bluepuma added a comment - 28/Jul/11 11:22 AM Everyone seems to be going video: from websites (Google+, Facebook, AOL.com/av, etc.) to SIP clients (Jitsi, Ekiga, Bria, EyeBeam, etc.). I hope a potential mod_rtmp Flash-client will include video functionality as well.

Jack Loranger added a comment - 23/Aug/11 12:58 PM Just got it to connect with the latest git and it looks like it has great potential. My vision includes auto filling the user and passcode and call destination to make a one click into a conference. I also like the interactive conference display on conference.freeswitch.org and would like to add photos to the display.

Jack Loranger added a comment - 02/Sep/11 5:31 PM Inside MapKeyPad.txt is a javascript function that can be placed in your <script> in freeswitch.html to send DTMF when 0-9 , # and * are pressed on your keyboard. The DTMF for # is sent when the Enter key (keycode 13) is pressed.

Ricardo Viteri added a comment - 13/Nov/11 5:54 AM Please add me as a tester.

Jack Loranger added a comment - 09/Feb/12 12:20 AM The BetterKeyPadHandler.txt file contains a simple JavaScript snippet that will enable the webpage to recognize key-presses and send dtmf for 0-9 and # * . it also maps the ENTER key to # , which can be easily excluded.