VoIP+IAX Program Theory for OM

Jeremiah Flerchinger jeremiah.flerchinger at gmail.com
Fri Feb 22 06:45:50 CET 2008

I think there would be more compatibility across devices & networks with 
a SIP based client than an IAX.  Asterisk supports both, although I not 
sure of the benefits of using one versus the other. 

I previously suggested looking into LinPhone since it is a GTK solution 
that is the basis for PhoneGaim & several other solutions.  I'm not sure 
I like the idea of Mozilla's Zap if it requires all the overhead of 
running Mozilla.

Right now I use the Gizmo Project server (without the Gizmo client) for 
handling of all my SIP calls.  I use GrandCentral to get a local POTS 
number (and a number local to my parents), voicemail I can check from 
the internet or any phone, and forward calls to both my cellphone & my 
SIP phone for free.  If someone has a better setup that doesn't require 
a land-line & still allows POTS calls, I'm happy to hear it.

I don't see running my own Asterisk server being that much more 
beneficial to my needs.  My dream is to have a freerunner that would 
default to SIP when attached to a wifi network and switch to the GSM 
network when wifi is not available.

Meadhbh S. Hamrick wrote:
> Hey Kyle...
> Yes.. this is a basic "convergence" function. I worked at Divitas 
> Networks last year trying to make this happen on WinMoblie and Symbian 
> phones. Sadly, I was unsuccessful in getting them to work on a Linux 
> Solution, so I had to do all my Linux based VoIP experimentation on my 
> own time, though I've been working exclusively in the SIP/SRTP 
> solution space.
> There's a lot of flexibility in this kind of solution.
> For one, you don't really need a POTS phone number connected to your 
> VoIP system, you can use it as a complete net-bearer system using IAX 
> or SIP.
> You could then add an IPCSP (IP Communications Service Provider) to 
> link the POTS phone number to your asterisk box. This is sort of what 
> Skype does with Skype-In / Skype-Out, but their network is closed and 
> you have to wait for them to port their client to various platforms :-(.
> And if you wanted to have a lot of fun, you could even roam between 
> VoIP and cell. This is essentially what the carriers do with UMA, and 
> it's what Divitas did, though uptake on their product has been pretty 
> slow.
> Several folks are in this game.... I've been looking at Mozilla's ZAP 
> project recently, and "under the hood" on the client I can't imagine 
> rewriting a media framework given that gStreamer seems to work quite 
> nicely.
> -cheers!
> -M.
> On Feb 21, 2008, at 9:33 AM, Kyle Bassett wrote:
>> Hey guys,
>> I've been contemplating writing an IAX client for OM which would be 
>> capable of the following:
>> Prerequisites:
>> -user has dedicated VoIP phone number routed to an Asterisk server[1]
>> ---OR a compatible VoIP provider that supports fallback[2] calling
>> -user has smartphone+OM with some form of internet access (wifi/bt 
>> internet/ethernet)
>> ---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)
>> [1] asterisk.org
>> [2] 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 established
>> 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!
>> -Kyle
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> -- 
> Meadhbh S. Hamrick (It's pronounced "Maeve")
> mhamrick at cryptonomicon.net
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