Page 510 - Asterisk™: The Future of Telephony

The default rtp.conf file uses the RTP port range of 10,000 through 20,000. However,
this is far more ports than you’re likely to need, and many network administrators may
not be comfortable opening up such a large range in their firewalls. You can limit the
RTP port range by changing the upper and lower bound limits within the rtp.conf file.

For every bidirectional SIP call between two endpoints, five ports are generally used:
port 5060 for SIP signaling, one port for the data stream and one port for the Real-Time
Control Protocol (RTCP) in one direction, and an additional two ports for the data
stream and RTCP in the opposite direction.
UDP datagrams contain a 16-bit field for a Cyclic Redundancy Check (CRC), which is
used to verify the integrity of the datagram header and its data. It uses polynomial
division to create the 16-bit checksum from the 64-bit header. This value is then placed
into the 16-bit CRC field of the datagram, which the remote end can then use to verify
the integrity of the received datagram.
Setting rtpchecksums=no requests that the OS not do UDP checksum creating/checking
for the sockets used by RTP. If you add this option to the sample rtp.conf file, it will
look like this:
[general]
rtpstart=10000
rtpend=20000
rtpchecksums=no

say.conf

The say.conf file is used to configure spoken language grammar rules for a number of
applications, such as SayNumber(). If you’re looking to use Asterisk in a language that
isn’t currently supported, you can script support through the configuation options in
this file.

sip.conf

The sip.conf file defines all the SIP protocol options for Asterisk. The authentication
for endpoints, such as SIP phones and service providers, is also configured in this file.
Asterisk uses the sip.conf file to determine which calls you are willing to accept and
where those calls should go in relation to your dialplan. Many SIP-related options are
configured in sip.conf, which was covered in depth in Appendix A.

sip_notify.conf

Asterisk has the ability to remotely notify a SIP phone to recheck its configuration files
or reboot by sending it a specially formatted, manufacturer-specific NOTIFY message

482 | Appendix D: Configuration Files