[Linphone-users] linphone prefs

[adam]

hi,

Im moving onto documenting the linphone preferences. I was wondering if 
anyone had a minute to check this terminology and the accuracy of the 
statements (only for Network 
Preferences):


Network preferences
The "Network" tab of the "Preferences" allows you to determine how 
Linphone should send data over the internet.



* Global option
Essentially there are not enough IP Addresses for all the types of 
objects that would like to have one (usually computers but also handhelds 
personal organisers, mobile phones etc). An IP Address is the group of 
numbers that identifies your computer on the internet. It looks something 
like this : "192.168.0.123". Because there are not enough combinations of 
these number groups to account for every device that needs to connect to 
the internet a new way of addressing is being developed, this is called 
"IPv6".  An IPv6 address might look something like this : 
"E2D6:0000:0000:0000:01B4:7BD8:D0A3:1220"

The option that Linphone offers here is to use an IPv6 address. To 
select the option means that you have an IPv6 address. If you don't know 
anything about this then chances are you are using the IPv4 IP addresses 
and you do not need to select this item. 



*Nat traversal options (experimental)
Most probably you will not need to worry about these settings. If you 
can connect to other Linphones or softphones no problem then just leave 
the default settings "No firewall" checked.

If you are in a private network, which is quite possibly the case if 
you connect through an office network, internet cafe, or you have a hub or 
router at home, then you may have some difficulties using Linphone to 
conenct to other Linphone that are not in the same network. This is 
because either Firewalls may block SIP traffic (the method Linphone uses 
to transfer voice data over the internet) or there is a device known as a 
NAT router between your computer and the internet. NAT is "Network Address 
Translation". A "NAT router" sits between the public Internet and a local 
("private") network. The NAT router correctly forwards requests from 
computers on a local network to computers on the internet and vice versa.

 However this can cause probles for SIP which is the method 
("protocol") that Linphone (and other softphones) use to make a connection 
with another softphone over the public Internet. If you are having 
problems calling other softphones when you know that the address you are 
calling exists then it could be because your computer is connected to the 
internet via a NAT. In this case you need to tell Linphone that this is 
the case. Linphone then offers two ways for this problem to be solved.
    
 - STUN
The first is to use a STUN server as an intermediatry. STUN stands for 
Simple Traversal of UDP through NATs (Network Address Translation).  
Essentially this means that the Linphone will communicate with a STUN 
server and the STUN server then tells Linphone the public IP address of 
your NAT router. It also reports to Linphone which port was opened by the 
NAT device for incoming traffic. Linphone then uses this information for 
connecting with a VoIP server or other softphones. If you know the address 
of a STUN server (and there are some free ones on the internet you can 
use) then enter the address of the STUN server in the first field.

A STUN server address looks like a URL. An example might be : 
"stun.myserver.com". In which case you would enter this information in the 
field and click the circle to the left which selects this option.

Note : the above settings are for the purposes of illustration only, 
there is no such existing STUN server address.

 - Firewall Address
If you know the IP address of your NAT router you can enter this into 
the second field.

In this case the address of the NAT router might be "203.11.2.37" 
(this is a fictional address), and hence I would enter this information 
and click on the circle to the left to ensure this option is enabled.

Note : the above settings are for the purposes of illustration only, 
there is no such existing NAT router address. 



* RTP properties
RTP stands for "Real Time Protocol". This set of rules ("protocol") is 
what Linphone uses to transfer audio data over the internet.
    
 - RTP port 
Because a computer must handle many different types of data transfer 
over the internet and many different sets of rules (protocols) for 
different types of data, the computer assigns a "port" for each type of 
data. Ports are identified by a number between 0 and 65535. Web browser 
traffic, for example, uses port 80. Linphone uses by default port 7078 for 
RTP traffic. Don't change this unless you have a need to and know what you 
are doing.

 - Jitter
Inorder to wnsure a nice smooth delivery of audio (voice), Linphone 
needs to hold some data in a "buffer". This means that it stores some data 
so that it can release it nice and smoothly, giving you nice smooth audio. 
If yopu are having problems with the audio 'breaking up' or 'stuttering' 
then you may wish to increase the time set for "jitter comemsation". Do 
that by sliding the marker to the right (you will see the number to the 
right of the slider increase).


* SIP info vs RTP rfc2833
DTMF is short for "dual-tone multi-frequency". This is the way that a 
telephone communicates with a telephone exchange or switching center. DTMF 
is familar to most people as "touch-tone", where different tones co-relate 
to different numbers on a phones key pad. These tones are actually being 
sent down the telephone line when you press them and they communicate with 
the telephone switching centers so your call gets sent to the right place. 
It is a kind of audio signalling process for telephoney. The system is 
known as "dual-tone" because each key is represented by two tones. 

Incidently "touch-tone" is a trademark and is just one version of DTMF.

The option "Use SIP INFO message instead of RTP for DTMF transmitting" 
determines which technique Linphone should use for carrying the DTMF 
information. If you choose "SIP INFO" (by checking the box) the DTMF 
information is actually represented numerically and not by a tone. If you 
choose RTP (the default) then tones are used and transported using the 
Real Time Protocol. Generally SIP INFO is not used by softphones. So you 
should probably leave this item in its default.








-- 



Adam Hyde
~/.nl

selected projects
http://www.xs4all.nl/~adam

the streaming suitcase
http://www.streamingsuitcase.com

simpel
http://www.simpel.cc

r a d i o q u a l i a
http://www.radioqualia.net

Free as in 'media'
email : address@hidden
phone : (+31) 6 16 009 046 (Netherlands mobile)
sip : address@hidden
skype : esetqualia