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Alternative setup
Description: This additional setup is included for students setting up the
connection of a SIP trunk towards the server that acts as telephony provider in
the training course, with the virtual machine created in their computers. The
procedure solves a NAT Traversal's potential problem when editing the file
sip_nat.conf
Goal: NAT Traversal potential troubleshooting
Instructions:
NAT (Network Address Translation), is a series of procedures used by routers
for connecting two networks, it basically consists in changing IP addresses and
ports when the connection goes through a router or a firewall. NAT Traversal
are techniques that establish and maintain connections in networks by using
TCP/IP or UDP protocols which go through gateways.
This process, which is very common in all networks because most of them has
a public IP that maintains the connection and several IPs in their private
network which need this process for allowing a connection between internal
users and external networks, is not exempted from potential problems,
especially in voice over IP.
It is not the purpose of this course to deepen on NAT or NAT Traversal,
however we will detail a useful solution for the case in this study. If you
purchase a SIP trunk from a telephony provider, you might face a similar
problem, where SIP connection is established but voice packages do not arrive
properly, and this causes the telephone to sound but we cannot hear anything.
We press the "Show Filter" button, and write sip_nat.conf in the File box,
then we press "Filter".
There are two points to consider, the "externhost" parameter is the public IP
assigned to your machine. In order to know which such IP is you may enter
any IP deployment web page, for example http://whatismyipaddress.com/,
then copy the result and place it in such field. The "localnet" parameter is
the LOCAL network's address in which your Elastix server is located.
When we finish the edition we press on "Save", and then we press "Reload
Asterisk", so changes are implemented.
Let's try by making some test calls through the SIP trunk that we configured,
and we should be able to resolve any related problem.