Directmedia

In Asterisk, SIP signaling to establish a call always passes through the Asterisk server. After the call is established, the RTP audio can also pass through Asterisk or can flow directly between the endpoints in the call.

Both options have advantages and disadvantages.

• When RTP flows directly between endpoints, Asterisk does not need to process that audio traffic. This can reduce server load when there are many simultaneous calls.

• When RTP flows directly between endpoints, Asterisk cannot detect in-call feature codes after the call is established. Features that depend on DTMF or dialed codes, such as transfers, forwarding, do not disturb, and call recording control, may not work as expected.

Therefore, if call transfers or other features that depend on in-call codes must work, the voice traffic must pass through Asterisk.

This behavior is controlled by the directmedia option in the SIP configuration. With directmedia=yes, RTP can flow directly between the endpoints. With directmedia=no, RTP remains bridged through Asterisk.

In MagnusBilling, this option is important when analyzing online calls, re-invites, audio IPs, call recording, transfers, and other features that depend on Asterisk staying in the media path.