Asterisk: The Future of Telephony, 2nd Edition

rtp.conf

The rtp.conf file controls the Real-time Transport Protocol (RTP) ports that Asterisk uses to generate and receive RTP traffic. The RTP protocol is used by SIP, H.323, MGCP, and possibly other protocols to carry media between endpoints.

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 ...