[asterisk-biz] Ribbit.com ? 1ezphone.com

Tue Mar 4 06:57:24 CST 2008

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

dean at cognation.net 

+1-212-203-4357

+61-2-9016-5642 (Sydney in-dial). 

> -----Original Message-----

> From: asterisk-biz-bounces at lists.digium.com [mailto:asterisk-biz-

> bounces at lists.digium.com] On Behalf Of Tim H. Panton

> Sent: Tuesday, 4 March 2008 3:46 AM

> To: trixter at 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 at 0xdecafbad.com>

> To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-

> biz at 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 at 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

> Belfast +44 28 9099 6461        US +1 516 687 5200

> http://www.trxtel.com the phone company that pays you!

> 

> 

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