I use Anveo Direct's Iristel route for termination via an Asterisk server and an OBi1032 IP phone.
Some time ago I set my phone's peer to use G.722 for testing purposes. This caused all G.711 calls to be transcoded. It didn't seem to cause much of a load on the Asterisk server nor cause any audio problems that I noticed, so I left it.
Today I called a supplier and thought, "dang that sounds clear!" Checked the logs, and discovered that for the first time ever since I started doing VoIP in 2008, I unintentionally called another VoIP user with G.722 capability. The audio came from a RingCentral IP.
I hope the next time this happens is far sooner than nine years from now.
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