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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I am are pleased to inform everyone that we are announcing support for the Asterisk open source telephony platform with our open sip gateway .I just finished testing a few Softphones and IP Phones behind my Asterisk server in our lab and so wanted to pass on the configuration information to everyone.To all the users , I do want to mention that there is a limitation of maximum number of simultaneous calls as per our fair usage policy.
Note: This configuation was tested with the complete source code for the most current Asterisk release 1.4.19.1

Asterisk Setup documentation for AIM Call Out can be found in our voice portal :  http://voice.aol.com/devices/asterisk-14

There are several videos on you-tube to help users get started on this open source telephony platform that can be used at home or small business.Visit my youtube channel for related Asterisk videos : http://www.youtube.com/VoIPalicious

If you are using Free PBX for your business , visit this link  from their article this week:  http://nerdvittles.com/index.php?p=215

For more details on howto get started with AIM CallOut visit  http://call-out.aim.com

We are currently working to provide support for an inbound calling service to compliment  AIM Call Out , so stay tuned for that announcement.Let me know if anyone has any issues in setting up Asterisk server in your infrastructure.Drop me a line at opensip@aol.com if you have any questions or comments.

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We are going to be offering free unlimited phone calls worldwide to this Mother's Day weekend, May 10th - 11th with AIM Call Out http://call-out.aim.com.Users can make as many calls as they want to more than 200 countries, using a headset & a microphone or by using your regular landline / Mobile phone or you can also use any SIP device including IP phones ,Telephone Adapters and wifi phones.
The list of devices supported by Aim Call Out can be found here http://voice.aol.com/SIP-Client-List
Users must set up an AIM Call Out account and add at least $5 in non-refundable call credit to get started. Users can then make as many calls as they want to more than 200 countries and will still have $5 to spend at the end of the free calls weekend.
WHEN: Users can make free calls between 6 a.m. EDT on Saturday, May 10th and 4 a.m. EDT on Monday, May 12th.
HOW: Users must have a free AIM screenname and use version 6.5 or higher of the AIM service which can be downloaded for free .Once users launch AIM they just click on the telephone icon at the base of the AIM Buddy List to get started.For more information on this offer pls visit http://call-out.aim.com/mothers-day 

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Nothing is better than instant messaging with your friends but sometimes when you need to get caught up, talking is easier than typing. So use the AIM Talk feature and have a voice conversation for free - hands-free. Forget long distance charges, talk to anyone around the world with AIM Talk free. Try it out with any of your AIM buddies.
Here’s how to do it:
1. Open an instant messaging window and choose your friend’s screen name just like you would to message them.
2. Click the “talk” button at the bottom of the IM window.
3. AIM will send the request to your friend and when they click “OK” you can start talking! It’s that easy.
        You’ll need a microphone and speakers, or a PC headset to hear your friend and let them hear you. It’s a PC-to-PC call and it’s free!
        When you want to call a friend’s home or mobile phone, check out our PC-to-phone service, AIM Call Out. It’s a pay-as-you go long distance service with a great voice quality at very low cost. You can dial your friend’s phone directly from AIM or use your regular phone and make calls with AIM Call webconnect feature.
     Read my last blog of how to make calls using your regular phone with the Webconnect feature.

Check out http://call-out.aim.com for more information.

If you have any questions about your free AIM Talk sessions, check the AIM troubleshooting information at:http://www.aim.com/help_faq/using/win/aimtalk.adp?aolp=

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Its the delivery of Voice over IP via a mobile handset. Mobile VoIP is delivered by a third party service provider over a WiFi or 3G network cellular network that a mobile device is connected to. Mobile VoIP service providers typically require a user to download software onto their mobile device in order to gain access to their service.

There are some regular Mobile phones like the "Nokie E series" that has a built in SIP stack which can work with our sip gateway.The only important thing that you have to remember when setting up this Mobile, make sure auto registration is turned off when making calls.

There are also wifi phones like the "Linksys WIP330" which lets you make voip calls over the wireless-g network connected to the Internet to make high quality low-cost VoIP calls through AIM CallOut. The handset features peer-to-peer dialing, speed dial, 3-way conferencing, call waiting, call transfer, and call forward, mute, hold and selectable ringtones.Check out the review from Tom Keatings blog from a while back
http://blog.tmcnet.com/blog/tom-keating/voip/linksys-wip330-wirelessg-ip-phone-review.asp

Few days ago , I got a call from this wifi phone from one of my co-worker & call quality was really good.I have to warn everyone that it may take a bit of time to get used to all the settings on this phone.Now you can make affordable long distance and international calls to Landline or Mobile phone using our Open SIP Gateway "SIP.AOL.COM" using this phone.

I am still busy testing lot of SIP devices along with our QA team members this week. Hopefully this weekend , I will  provide some screenshots on howto setup AIM CallOut with this wifi phone in here.

Tags: Linksys wip 330, wifi phones, Mobile VOIP, SIP Mobile phones, Nokie E series, Nokie Mobile Phones

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Great news for Asterisk , FreeSwitch , sipX ,CallWeaver ,YATE ,Trixbox users !

Our platform is compatible with open source telephony platforms.

Stay tuned for my instructions of how to setup Opensouce telephony softwares with our Voice Gateway.

I am thinking of  testing the popular Asterisk PBX Software in our labs sometime soon.

Let me know in the comments tab below if you would like to see your open source applicated tested in our lab .

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