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Basic Asterisk/VOIP implementation questions

Posted on 2006-06-26
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Last Modified: 2012-05-05
Hi. Basically I'm interested in using asterisk to support users of a website to both conference in with eachother and also leave "voice blogs" in certain users boxes on the site. some trendy people have referred to this as "mobcasting". In a nutshell this is what I'm trying to accomplish.

My question is, in addition to a high bandwidth dedicated server (which I have already, 3 GZ 1 GZ RAM, freebsd, 3.3 TB/mo. on an OC3 connection), would I need any type of hardware to interface to only RECEIVE calls onto the site? What other types of telephony fees would I be looking at for a starter system? I found this: http://cgi.ebay.com/IAX-Native-FXS-for-Digium-Asterisk-VoIP-PBX-Beats-IAXy_W0QQitemZ130001205393QQihZ003QQcategoryZ61841QQrdZ1QQcmdZViewItem
is compatible with asterisk.

what would be the max. number of simultaneous sip connections given my hardware configuration?

Recommend an interface for Asterisk?? as there are many, I'm fairly technical but wouldnt mind investing in something that can make me quickly productive. Have never used Asterisk so am not aware of how difficult it is to install..

thanks

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Question by:dprasad
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by:grblades
ID: 16990604
In order to receive calls and to make calls between users the only interface you need on the asterisk box is a network interface.
You will only need additional interfaces if you want to link it to the regular telephone network.

The item you found on ebay allows you to plug it into the network somewhere and connect a standard analogue phone and link it to asterisk. Basically it allows you to use an analogue phone instead of a software client. The benefit of IAX rather than SIP is that it has no problems with firewalls so works better if it is located behind routers/firewalls.

To see how many calls you can simultaneously handle see http://www.voip-info.org/wiki/view/Asterisk+dimensioning
Basically if both parties are using the same codec (audio compression method) then asterisk will use very little cpu. If it has to convert then it will use a lot more. So you could force everyone to use the same codec and this will help you cope with significantly more simultaneous calls.
A bit of an estimate but I would say you should be able to have around 10,000 connected users (registered phones) with 1000 active calls. Users recording messages will use a bit more resources.
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by:dprasad
ID: 16995820
wow 1000 is a lot, 1000 at the same time?

Do you know of a configurable java applet client that uses IAX? looking for something that would require minimal user configuration, something like some opf the sip client I see, that install directly into your browser then put a little icon in your system tray or something like that. Not anything you have to download, unzip, install etc..
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grblades earned 500 total points
ID: 16999238
Yes once the call is established the two phones send the voice directly between them so there is very little server load associated. It only needs to keep track of which calls are active. It is only when different codecs are used and asterisk has to decode and reencode the audio between clients that the server load is much higher.

http://moziax.mozdev.org/index.html which is a firefox plugin
http://www.hem.za.org/jiaxclient/
http://www.dorstel.de/iaxphone/index_en.html

Diax is not bad for a software client. It does not require installing. You just give the user a directory containing the files and they run the exe.
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by:dprasad
ID: 17001319
thanks for your help grblades. I will be asking more questions in this space. got asterisk installed on my freebsd box yesterday.
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