Come utilizzare il servizio
di audioconferenza
Il sistema di audioconferenza puo’ essere utilizzato in due modalita’:
1) Collegamento singolo. Si utilizza l’apparecchio per audioconferenza che si ha a
disposizione come un normale telefono vivavoce, chiamando direttamente il
numero della persona con sui si deve avere la conversazione telefonica oppure
facendosi chiamare da questa. E’ bene tener presente che in questa modalita’
potra’ connettersi all’audioconferenza un solo utente e che il costo della chiamata
sara’ uguale a quello di una normale telefonata (urbana o internazionale).
2) Audioconferenza. E' possibile richiedere al CNAF una conference room inviando
una e-mail a [email protected] (preferibile) o chiamando telefonicamente le
seguente persone:
Stefano Zani +39 0516092749
Alessadro Italiano +39 0516092751
Marco Bencivenni +39 0516092869
Potranno essere assegnate due tipi di conference room, una temporanea (per
audioconferenze occasionali) e una permanente (per audioconferenze periodiche).
E’ possibile collegarsi ad una audioconferenza in due modalita’:
1) Utilizzando
un
normale
telefono
e
chiamando
il
numero
06 62288548. Con questa modalita’ il costo della chiamata sara’ pari ad una
telefonata urbana.
2) Utilizzando un cliente software/hardware e configurando in pochi passi un sip/iax
client. In questa modalita’ la chiamata sara’ gratuita.
Le istruzioni per la configurazione di un sip/iax client sono disponibili al seguente
link:
http://www.cnaf.infn.it/main/index.php/Audio-Video_Conferenza
Come configurare un sip client
A soft phone is an IP telephone in software. It can be installed on a personal computer and
function as an IP phone. Soft phones require appropriate audio hardware to be present on
the personal computer they run. This can either be a sound card with speakers or earphones
and a microphone, or, alternatively a USB phone set. Soft phones are inferior to hard phones
but cheaper to obtain, many are available as a free download.
Personally, I suggest to use:
X-Lite for Window
Sjphone for Linux
An Ethernet hard phone is a self contained IP telephone that looks just like a conventional
phone but instead of a conventional phone jack, it has an Ethernet port through which it
communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a
broadband hard phone communicates directly with a VoIP server, VoIP gateway or another
VoIP phone it does not require any personal computer nor any software running on a
personal computer to make or receive VoIP phone calls. It can be used independently, all
that is required is an internet connection. While PC based software solutions are cheaper, a
hard phone is the best solution for IP telephony.
A low impact solution is rapresentad by Analog Telephone Adapters, better noted as ATA.
Read here
Market offers also:
Cordless Hard Phones with IP interface on their base station.
WLAN or WiFi Phones are hard phones with a built-in WiFi transceiver unit instead of an
Ethernet port to connect to a WiFi base station and from there to a remote VoIP server
Other info here
Please note: you are free to use any SIP compliant client, just use the following
parameters:
SIP proxy phone.infn.it port 5060
Login: 2004 Password: 5555
Here you can find some useful numbers
Here some NAT/Firewall infos
Here if you have audio problems: background noise, echo, etc
Alfredo Pagano 14/03/2007
X Lite for Windows
Download and install it
1. Add an account (as Domain use: phone.infn.it)
2. Insert the correct parameters (see the image) click Ok and enable it (Check “Enabled”).
3. Test the environment calling *43 for the echo test.
And remember to MUTE yourself when you are not speaking.
Here you can find some useful numbers.
Here some NAT/Firewall infos.
Here if you have audio problems: background noise, echo, etc
Here for the complete error code list
You can find a complete User Guide here (pdf file)
Alfredo Pagano 14/03/2007
Sjphone for Windows and Linux
Step to step guide to install and configure SJPhone on Windows and/or Linux:
STEP 1
Download the software from http://www.sjlabs.com/sjp.html
STEP 2
After installing, the default soft phone will appear.
For Windows version: Click on the screwdriver symbol
For Linux version: Click on "Phone -> Preferences" Tab
The *Options* window will appear. Enter your name and email address in the 'User
Information' Tab.
STEP 3
Click on the 'Profiles' Tab. In the Profile Tab click on 'New'.
When the 'Create New Profile' window appears, in the 'Profile name:' enter INFN and
in the 'Profile' choose 'Calls through SIP Proxy' Then click OK.
STEP 4
A 'Profile options' windows will appear. Click on the SIP Proxy Tab. Enter in 'Proxy
Domain:' phone.infn.it and enter 5060 in the field to the right side of it. Under 'User
domain:' enter phone.infn.it. In the same SIP Proxy Tab's Advanced options, put a
checkmark on Use separate register. Under this enter in 'Register domain:'
phone.infn.it and enter 5060 in the field to the right side of it. Click OK.
STEP 5
Back in the 'Profiles' Tab you will find an varphonex.com profile. Highlight INFN
and then click on 'Initialize.." A 'Service:INFN' box will appear. Enter in 'Account:'
2004 and in 'Password:' 5555 and click OK.
STEP 6
In the 'Profiles' Tab again highlight INFN profile and then click on 'Use...'
STEP 7
Click on the 'Call Options' tab. Make sure that under 'Incoming Calls' put a check
mark on 'Automatically accept incoming calls'. In 'Outgoing calls' put a
checkmark on 'Use following host address' and make sure you choose the IP
address of your computer.
STEP 8
Click on 'Audio' Tab and make sure that in 'Sound devices' the Playback: is
using your sound device.
Click OK.
STEP 9
Call your conference number!!
And remember to MUTE yourself when you are not speaking.
Here you can find some useful numbers.
Here some NAT/Firewall infos.
Here if you have audio problems: background noise, echo, etc
Alfredo Pagano 14/03/2007
Come configurare un iax client
IAX is the Inter-Asterisk eXchange protocol used by Asterisk, an open source PBX server
from Digium. It is used to enable VoIP connections between Asterisk servers, and between
servers and clients that also use the IAX protocol.
If you want understand more, start to read here
In few words, IAX2 is better than SIP because:
1. IAX2 uses just one UDP port, 4569 (this solve most of the NAT problems)
2. IAX2 supports PKI-style authentication and trunking.
3. IAX2 uses only 2.4k for a single call
I recommend:
IDEFisk: A Windows and Linux softphone from asteriskguru.com
Click to enlarge
Download and lunch it.
Configure it following this screenshot
1. Just lunched
2. Open the Option Menu by Right Clicking on Idefisk
3. Select "Account Options" and fill the camp.
Click to enlarge
Username 2003 Password 5555
And remember to MUTE yourself when you are not speaking.
(Click on Microphone symbol that will become red, click again to Unmute)
You can find a complete function list here (pdf file)