Re: [Linphone-users] linphone prefs

[adam]

thanks! I have added those comments to the manual. In the next days I will send 
to the list the rest of the Preference information I am writing.

thanks again for the corrections :)

adam

..on Tue, Oct 10, 2006 at 09:19:51PM +0200, Simon Morlat 
wrote:
> Hi,
> 
> This seems perfect to me.
> Just a few remarks: 
> - SIP is the signaling protocol, it does not carry voice, it makes phone 
> agree 
> on how to setup the voice and perhaps video streams, and signal when the call 
> terminates.
> - changing the RTP port makes no problem: it's dynamic, there is no risk at 
> doing it. However it must always be an even number (7078, not 7079).
> 
> Simon
> 
> >
> >
> > 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.

-- 



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