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. > > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
