Well I suppose now is as good a time as any to break cover J
If you are interested in a SIP browser based solution check out;
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
If you are a
Regards,
Dean Collins
Cognation Pty Ltd
dean@cognation.net
> -----Original Message-----
> From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-
> bounces@lists.digium.com] On Behalf Of Tim H. Panton
> Sent: Tuesday, 4 March 2008 3:46 AM
> To: trixter@0xdecafbad.com; 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" <trixter@0xdecafbad.com>
> To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-
> biz@lists.digium.com>
> 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@mattruby.com, "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
>
> http://www.trxtel.com the phone company that pays you!
>
>
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