Well I suppose now is as good a time as any to break cover :-)
If you are interested in a SIP browser based solution check out; www.Surphone.com <http://www.surphone.com/> Yes there are server based and ASP based pricing models, yes it uses Flash - not it doesn't use Adobe FMS. No this isn't related to the work I did with Mexuar, Yes I am consulting with Surphone for the commercialization of their technology and yes I will be selling the technology here in the USA. If you are a USA based client and have an interest send me an email for more details. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-biz- > [EMAIL PROTECTED] On Behalf Of Tim H. Panton > Sent: Tuesday, 4 March 2008 3:46 AM > To: [EMAIL PROTECTED]; Commercial and Business-Oriented Asterisk > Discussion > Subject: Re: [asterisk-biz] Ribbit.com ? 1ezphone.com > > (Sorry about the top posting - It's just the way Zimbra does it) > > There are a couple of things to look out for here > (straying into tech issues): > 1) buffering - TCP tends to get buffered in the kernel to a > much greater extent than udp - so you can easily find yourself > with seconds of latency. > 2) codecs - The only low-latency codec supported by flash > is patented and expensive to license, so the gateway to PSTN > or 'normal' VoIP will always have to carry an aditional cost > of the nelly-moser codec license. > 3) protocols - Flash is using a streaming protocol (RTSP), > which isn't a VoIP protocol, so has not got the VoIP features > we have come to expect. > > All of which is why adobe is (supposed to be) adding SIP > to some future version of Flash. > > - Ok, I admit it, I'm biased, I'm in the Java - IAX camp :-) > > But in general I'm sure that this sort of web-telephony > integration is inevitable. See http://www.phonefromhere.com for our > latest experiment - an iGoogle 'phone home' gadget. > > Tim. > > > > ----- Original Message ----- > From: "Trixter aka Bret McDanel" <[EMAIL PROTECTED]> > To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk- > [EMAIL PROTECTED]> > Sent: 29 February 2008 12:47:21 o'clock (GMT) Europe/London > Subject: Re: [asterisk-biz] Ribbit.com ? 1ezphone.com > > > > > > ----- Original Message ----- > > > From: "Mike Clark" > > > To: [EMAIL PROTECTED], "Commercial and Business-Oriented Asterisk > > > Discussion" > > > Subject: Re: [asterisk-biz] Ribbit.com ? > > > Date: Mon, 17 Dec 2007 17:21:50 -0500 > > > > > 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. > > > > > Flash does not afaik support UDP so the RTP part would be difficult at > best. I am unsure if the really new versions do or not. Granted you > could have a plugin (flash does have the ability to execute programs > that are in a special directory) which really only would need to be a > tcp->udp converter if you wanted, although it could be a full RTP stack > as well instead of doing that in flash. > > Gizmophone has a web component that transmits the audio via HTTPS via > flash. I havent looked at ribbit so I dont know if that is how they are > doing it or not. They also use a plugin to try to limit how many calls > you can do at one time off one box (they did give away free minutes at > one point, they may still do that). > > While the SIP RFC requires TCP support for signalling, the media would > still be udp and still be the problem. And if you want to connect to > asterisk you have to use UDP signalling since asterisk does not yet > officially support TCP, despite the RFCs requirement. > > Personally what I think would be better is a very simple app that can > send events (on/off hook, dnd/presence, dtmf digits, number dialed, etc) > as well as media (just stream it from the mic direct, which is something > that flash has built in). This would connect to some server side > process that will then connect to whatever protocol you prefer for > termination elsewhere. > > On lossy networks you would have a problem of a dropped packet causing a > retransmit, however this may not be that big of a problem in many > environments. If you have any sort of jitter buffer you should be able > to resync the call dynamically so that packet loss does not cause a > growing skew between leg A and leg B. This is probably the biggest > problem to solve, and I do not know how big of a problem it will be for > most users (for some it will be a killer). > > Now if they have java installed as well, flash can do liveconnect calls > to the JRE, but if you are going to go that route, it might be better to > just do it all in java to begin with. > > Now flash recently aquired a key person that was involved in SIP stuff. > The theory (and some statements officially) indicate that the intention > is to build a proper sip stack into flash, but that has yet to be > released. > > There are other bridges that exist to basically do the tcp->udp > translation, which could be run on the users system. Examples include > http://www.transmote.com/flosc/ > > While this is designed to do the open sound protocol, it would not be > difficult to make it do something else, and if you really know action > script you can get around little things like you dont have to do xml > with the xmlsocket, you can bypass the null byte terminator that is > often sent, etc. > > For what is needed to do the tcp->udp bridge it wouldnt be hard to write > that on your own, and then go nuts. > > > -- > Trixter http://www.0xdecafbad.com Bret McDanel > Belfast +44 28 9099 6461 US +1 516 687 5200 > http://www.trxtel.com the phone company that pays you! > > > _______________________________________________ > --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 > > > _______________________________________________ > --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
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