Session Initiation Protocol, or SIP, is a standardized Voice over IP (VoIP) protocol. This protocol relies heavily on the RTP, which uses UDP ports in the TCP/IP stack. It presents addresses in much the same format as email, as user@domain.
We configure this protocol by editing /etc/asterisk/sip.conf
. This file has a number of settings in a [general]
section, followed our definitions of users.
There is a whole host of options that we can set. These options include:
context
: Sets the default context for calls coming into the server. These calls can be from our users, or if we are connected to the Internet, they can be from anywhere. Just to be on the safe side, we should not set this to be a context that can call long distance. The default is "default".realm
: Sets the realm of the server. As we discussed earlier, the calls are addressed like email, in that the format isuser@domain
. This variable is how we set the domain part. This could be your host name or a domain...