[asterisk-biz] Ribbit.com ?

[Zoa]

I've looked at using flash for sip before, some people seem to actually 
do it with actionscript 3, but i have no clue how.
I never used actionscript before, but from what google tells me, adobe 
is still working on SIP support in flash and i think actionscript is not 
powerful enough to do SIP/RTP ?

http://www.voisen.org/projects was the most interesting link i found.

I suppose the best way would be to use red5 to force it to connect to 
asterisk, and use the flash streaming protocol to send the audio to 
red5/asterisk for media conversion. (this new jack thing in svn can 
maybe help with that?)

Zoa

Trixter aka Bret McDanel wrote:
> On Mon, 2007-12-17 at 17:21 -0500, Mike Clark wrote:
>   
>> Ribbit has a totally different model as they are a full blown ITSP and 
>> have provided a Flex/Actionscript API to their Flash phone component at 
>> no charge to developers. I have an app ready to roll as soon as they are 
>> completely live.
>>
>> I would love to see a similar type API to a Flash SIP or IAX2 component 
>> where I could access my own Asterisk or Freeswitch server.
>>
>> Mike Clark
>>
>>     
>
> How is audio transport done?  Gizmocall.com has had a flash client for a
> while, which streams tcp port 443 (it is ssl data my guess is that its
> straight https).  They have a plugin which I think is just for
> authentication.  To deal with any potential loss that may occur they
> probably send a modified rtp style packet with timestamps so you can
> drop audio to sync up with wall clock when it comes back.  Flash
> reportedly has issues doing udp, as in it doesnt do it (I am not a flash
> monkey so I dont know for sure, but that is what others have said).
>
> If that is the case at least to FreeSWITCH you can stream mp3s and
> bridge that to another call leg.  This means it is currently in its
> present form compatible with standard flash and a trivial client could
> be fashioned.  Ming.sourceforge.net lets you code flash apps instead of
> using a gui to create them, and may make development easier for some.
>
> There exists the ability to bridge text IM to jabber or sip (to from,
> mix/match) and can even do a text based ivr if you wanted.  XML-rpc
> gives you control over the system, etc.  It wouldnt be that hard to use
> this as the base to compete with ribbit if you wanted.  
>
> OpenMRCP (another OSTAG-Open Source Telephony Advancement Group -
> release) lets you interface to ASR/TTS that way if you wanted.  There
> are native cepstral and lumenvox modules that dont require that but it
> gives you a choice.
>
> There exists RTP failover (someone pulls the power cord the call stays
> up, including state in the application, the customer never knows),
> clustering and other things that make this a not too shabby solution.
>
> I am unsure on the capabilities of asterisk in this regard it may do
> some or all of it.  It may be just as trivial to integrate this into
> asterisk.
>
>