The SIP Interconnect Model

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As Session Initiation Protocol (SIP) becomes more widely interconnected
accepted by the service providers need to define an interconnection
becomes more valuable guideline. This paper takes
account of the SIP and SIP extensions in common use, and
defines a fundamental set of requirements for SIP service providers
(SSP) to apply, in their signaling (FE) or
Signaling Path Border Elements (SEES) for peering.

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[quote="crickdev"][quote="satphoneguy"]what is an IP endpoint? or to be more specific does this include skype, gtalk, msn, yahoo or only SIP?[/quote] IP endpoints are those PC-based ednpoints to which calls can be routed. These include Ribbit Phones, Skype, MSN, and Google Talk. We will add others over time, but these three are the leaders (Yahoo would be next). Calls coming into Ribbit can be answered on any of these endpoints. [quote="satphoneguy"]also do you plan to sell a hosted VoIM/skype switching service that can be accessed using SIP devices?[/quote] Hmmm... I'm having to guess a little bit on the use case you're thinking of. Here's one thing you can do - initiate calls via the Ribbit API/Amphibian web page that ring first on any Ribbtit supported IP endpoint (including Skype) then ring the destination endpoint. Does this help? crick[/quote] crick, i was thinking more in terms of a service that would allow users of hardware SIP devices to place calls to skype and IM services. i known this does not fit in with the mainstream of what ribbit is all about(flash based softphones) but you do seems to have the softswitch infrastructure that makes this possible and there is a demand for such a service since large numbers of people do prefer to use a more traditional telephone like devices. i would imagine that you would not want offer this for free since it would likely be abused by people with no other interest in the ribbit service. on the other hand it could be really attractive as part of say an PSTN termination package with a monthly fee. another question. will calls to skype go out on the users own skype accounts or will they come with a caller ID of ribbits servers? also will incoming from skype to amphibian also be possible? spg

[quote="citiwidebroadband"]Spg I am not sure if your question address to me or not? MC[/quote] questions were intended for the ribbit strategy/business development team. but this is an open forum so all opinions/feedback is certainly welcome. spg

can you elaborate a bit on what is an IP endpoint? or to be more specific does this include skype, gtalk, msn, yahoo or only SIP? i ask because this is one of the most interesting aspects of the ribbit network and would create a double selling point to give both a flash based softphone plus the users get access to the IM networks as a bonus. also do you plan to sell a hosted VoIM/skype switching service that can be accessed using SIP devices? spg

[quote="crickdev"]We get lots of requests for SIP interconnect only. These requests most resemble a request for a SIP softphone built in Flash. The idea behind the requests are that the developer would be able to connect a Flash application directly to a SIP server, Asterisk, or some other SIP-based voice service element. But Flash can't do this natively. What Ribbit has done, is figured out a way to connect voice through Flash/AIR applications to any telephony network (VoIP/SIP, land line, mobile, Google Talk, MSN, Skype). This is exciting stuff and, we believe, a way to add high value to your applications - far greater value than plain old voice! To do this, Ribbit must receive the voice signals at it's servers hosted by [url=http://www.opsource.net]OpSource[/url] in an [url=http://www.equinix.com/prod_serv/ibx/ibxCenters.php]Equinix[/url] facility, manipulate and massage the signals, and then send them on their way. So EVERY CALL using Ribbit technology passes through the Ribbit service network. What does that mean if you're working with Asterisk? It means that you can connect a Ribbit Flash/AIR application [i]indirectly[/i] to your Asterisk server. In either direction, the the voice path must traverse the Ribbit network (your Asterisk server won't connect directly to a Ribbit Flash/AIR application). This goes the same for connecting a Ribbit Flash/AIR application to [u][i]ANY[/i][/u] network. If this works for you, you could use the Ribbit network to connect to your SIP network with our "on-net-only" service. In this case, you route calls to and from your application through the Ribbit network pointed to your SIP address. The price for Ribbit-on-net only service is $5 per logged-in account per month. For example, if there are fifteen accounts on your application, and only seven of them log in during the month, the cost that month is only $35. You would not pay anything for the accounts that do not log in. Usage on-net is unlimited to and from any IP endpoint/SIP network. (*a monthly minimum of $30 is required*) We have some customers who are finding this to be a great model for private, networked, in-web applications. They simply create as many accounts as they need (by e-mail registration, coming available very soon in a new API release) and are billed only for those accounts that log in each month. Remember, traffic always traverses the Ribbit network, so this may not be the ideal solution for campus communication using Asterisk as your calls between offices would trombone to and from Ribbit; however, it could be a great solution for remote workers on Asterisk. This solution is already working for some of our customers - let me know if it works for you! Merry Christmas, Crick[/quote] Crick, Thank you for this detail answer, I got one concern that your comment did not address : How do you prepare for failed server, and if I have 400 businesses using the component like what I did on our (HotSpot) demo site, I do not see your business model will be profitable if all of the callers are making the calls from US to Canada using long distance calling, it will be more pratical if Ribbit passing the request to regional Ribbit Network server, and the regional server will then pass the request to Asterisk using local PSTN Gateway (SIP to Asterisk), this will also work as yur failover network if any one of your servers drop connection. please comment. regards, Michael