VoIP+IAX Program Theory for OM
schmidtm524 at googlemail.com
Thu Feb 21 19:05:41 CET 2008
thanks for the ideas. we had a discussion about the same way you describe to
bring in VOIP and Video Chat into the serverless messenger
This is a pgp encryptred tunnel from one node (e.g. on your Open Moko Phone)
to your node at home,
If at home then an asterisk IAX server is installed, you can route the Call
from RS-Node to RS-Node to the normal phone line.
Or: the other way round, the Asterisk/IAX server at home could voip or gsm
to the cellular phone.
The idea of using the serverless messenger for this situation is, that all
communication is encrypted
as well from mobil-me to mobil-friend, as well as from mobil-me to home-me.
So what does this sentence exactly mean: "I've been contemplating writing an
IAX client for OM "?
The code is already working or you want to write?
Wouldit be possible to plug that protocol into the serverless messenger
So that the communication layer of this is used? then a third-party free
voip line is guaranteed.
That means, if there is a flat data account, then users can talk from mobile
to mobile secure, or phone out from their node at home.
Please have a look to integrate that in a messaging application, which is
pgp secure. Thanks
On Thu, Feb 21, 2008 at 6:33 PM, Kyle Bassett <kylebassett at gmail.com> wrote:
> Hey guys,
> I've been contemplating writing an IAX client for OM which would be
> capable of the following:
> -user has dedicated VoIP phone number routed to an Asterisk server
> ---OR a compatible VoIP provider that supports fallback calling
> -user has smartphone+OM with some form of internet access (wifi/bt
> ---in addition to regular GSM/CDMA service on the smartphone
> -[optional] user has regular GSM/CDMA cell phone+service
> Usage situation:
> The user exchanges the VoIP number with all contacts. When someone
> attempts to contact the user, via dialing the VoIP number, the asterisk
> server answers the call and checks to see if the user is available over
> VoIP. If the user's smartphone is on and connected to the internet, the OM
> IAX client should connect to the asterisk server automatically (depending on
> the user's settings, etc.) If the phone is available over VoIP, asterisk
> attempts to ring the user over VoIP for a specified time. If the user does
> not answer or a connection problem persisted, then the asterisk server can
> forward the call to the user's regular (OM or third party) cell line.
> Asterisk is very flexible and many permutations of this example can be
> accomplished, ie. calling all the numbers at once, and forwarding the call
> to the first to pickup.
> There are many benefits to this system:
> --User has complete control over the call routing and voicemail system
> --User can prevent the usage of regular cell airtime by using VoIP as much
> as possible
> --User can give one phone number to all contacts and have asterisk decide
> how to handle the call (routing not just to the cell phones, but to home
> lines, etc.)
> --During the debugging process with OM+GTA0x, users can carry both phones
> and still use just one number throughout the day
> ----Call comes in->asterisk tries OM[VoIP]-> tries OM[GSM] -> tries
> regular third party cell phone (can also ring all numbers at once)
>  asterisk.org
>  fallback calling is a service that allows VoIP users to enter a number
> (regular landline/cell) as a fallback in case the VoIP call cannot be
> I have tried to remain as general as possible, that way this post won't
> become outdated with specifics to any specific hardware. In reality, we
> want OM on as many phones as possible. ;-)
> Please provide any feedback or ideas!
> OpenMoko community mailing list
> community at lists.openmoko.org
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