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KPhoneIS User Manual: How to Connect

Connection Modes: UA Point to Point

UA Point to Point

1st step

select a free port for SIP (default 5060)

2nd step

select -p <port> and -t when you start the KPhoneSI
Example: "kpsi  -u p2p -p 5000 -t -v 3"

The URI is then set to port@dissipate-address:port

3rd


Call a partner using userid@host:port


Connection Modes: Using a proxy agent

Public IP

1st step

Get the settings from your provider

2nd step

open the Identity – Editor

3rd step

Enter the settings, your provider gave you

Field of the Identity – Editor

value

Full Name

userid@proxy or a name of your choice

User Part of SIP URL

UserId (SIP Phone Number)

Host Part of SIP URL

IP address of the Proxy-Agent/Softswitch

Outbound Proxy

IP address (and port) of the proxy-agent, softswitch or SBC

Authentication Username

enter if necessary

Auto Register

set it

4th step

OK-Button

press it



Connection Modes: Using a proxy agent and STUN

Public IP

1st step

Get the settings from your provider

2nd step

open the Identity – Editor

3rd step

Enter the settings, your provider gave you

Field of the Identity – Editor

value

Full Name

userid@proxy or a name of your choice

User Part of SIP URL

UserId (SIP Phone Number)

Host Part of SIP URL

IP address of the Proxy-Agent/Softswitch

Outbound Proxy

IP address (and port) of the proxy-agent, softswitch or SBC

Authentication Username

enter if necessary

Auto Register

set it

OK-Button

press it

4th step

Go to Preferences->Sip Preferences -> Socket

Use Stun Server

yes

Symmetric IP

yes

Symmetric Media

yes

Stun Server

ask your provider

Request Period for STUN server

ask your provider

OK-Button

press it


The "NAT traversal" Problem: A short Introduction

The SIP design assumes, that every User Agent (UA) has a global internet address, thus is freely
reachable from every other UA. Unfortunately, many of us are connected to a LAN and reach
the Net via an edge router utilizing NAT (network address translation). This method works fine
with client/server applications, but makes SIP telephony almost impossible! But there are different
methods to overcome this "NAT traversal" problem.

"NAT traversal" solutions

There are two methods to SIP phone from behind of the NAT

  1. Use a media switch
  2. make the UAs know their translated payload IP-address and port
The second methode may be achieved in different ways:
STUN usage may be necessary  in case 1, too! Ask your provider

In the first case , the SBC/Softswitche (who incoporate teh media switch) sets up a
client/server relationship to both UAs and switchs the payload packages between them.

In the  case 2a ,  the SBC first sets up a client/server relationship to both UAs, but
redirects the media paths to a point to point relationship if possible.

In the cases 1 and 2a, the SBC may be integrated into your proxy agent (then your
proxy and your outbound proxy addresses  will the same) or be a seperate device (your
proxy and your outbound proxy addresses will differ)

In the case 2b, the UA explores its translated payload address using a STUN-server, which
is located in the net. It then fills the appropiate fields in the SIP messages. There are preconditions
for this method:
  1. You must not be connected to a SoftSwitch, SBC or any other device, capable to do media switching for you
    in most times this is the case, if your proxy and your outbound proxy addresses are different
  2. Your router must have a NAT of type "full cone", "half cone" or "port resticted". A so called "symmetric NAT"
  3. won't work
  4. Your firewall must not enforce client/server relationships

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