Open Voice Platform

I want to take the time to answer some questions i recently got from some embedded software developers who wanted to know if we would support open sip stacks with our voice platform .The Open Voice platform consists of a number of service offerings, all based on open standards, that allow any application to use the AIM Call Out infrastructure to place calls from any standards-based hardware device, or, for a developer more important, any custom piece of software. This essentially allows you to build your own softphone or to integrate some sort of phone functionality into your line-of-business application. I would encourage everyone to visit my past  blog article where I explain about our voice platform and the RFC's we support currently for this program.
Just a breif overview of SIP Trunking,
A SIP trunk is a service offered by an ITSP (Internet Telephony Service Provider), like AOL, that permits businesses that have a PBX installed to use Voice-over-IP (VoIP) also outside the enterprise network by using the same connection as the Internet connection. SIP Trunking is the next-generation communication platform that gives businesses the ability to SCALE their voice solution gradually to meet their individual business needs, SIMPLIFY the administration of their network by reducing providers, and SAVE money by right-sizing their bloated telecom budget.

               For enterprises wanting to make full use of their installed IP-PBXs and not only communicate over IP within the enterprise, but also outside the enterprise a SIP trunk provided by an ITSP like AOL to connect to the traditional PSTN network is the solution.Unlike traditional telephony, where bundles of physical wires were once delivered from the service provider to a business, a SIP trunk allows any company to replace these traditional fixed PSTN lines with IP connectivity to AOL Voice platform for PSTN termination. SIP trunks can offer significant cost-savings for enterprises, eliminating the need for local PSTN gateways, costly ISDN BRIs or PRIs .

Now for the good news, one of our engineers, Mark Blomsma from our developer network team has written an excellent article on how to write a .NET VoIP application that integrates with AIM Call Out. He used the sip.net framework and the Sipek sip phone, and has written a simple voip application that is interoperable with the AOL Voice network.
                         The Sipek phone is a VoIP engine powered by pjsip.org open SIP stack.Mark had to use Sipek SDK's since sip.net currently does not support any RTP stacks.In a nutshell the SipekSDK wraps a C DLL and offers a managed API that C# developers are able to use.Currently, the Sipek softphone supports a C# wrapper to connect to the pjsip stack.Another nice feature is that this Sipek wrapper can be used in other .NET projects, including Windows Mobile.
                       The code for setting up a connection to AIM Call Out using this sip.net programming stack can be found in  Mark's blog

                        Finally I would encourage all developers using open sip stacks like NIST SIP, minisip, mjsip, opensipstack, osip2/exosip2, PhAPI, PJSIP, resiprocate, Sofia SIP, etc. writing their own application ,to use our open voice platform and provide us your feedback.

                      Interested developers can send me an email at opensip@aol.com if you would like to know more information on our voice platform . I can provide some AIM Call test accounts for people interested in testing your software or hardwarewith our open sip gateway.

                       For more details oh howto get started with AIM Call Out visit http://call-out.aim.com or our developer portal at  http://dev.aol.com/api/aimcall .

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