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Re: [Linphone-users] linphone prefs


From: Simon Morlat
Subject: Re: [Linphone-users] linphone prefs
Date: Tue, 10 Oct 2006 21:19:51 +0200
User-agent: KMail/1.9.4

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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