When your VoIP call sounds fuzzy or cuts out, the issue might not be your internet—it could be what’s happening to the audio in the background. Transcoding, the process of converting audio from one codec to another, often to bridge incompatible systems. Also known as codec conversion, it’s common in call centers, SIP trunks, and cloud PBXs that connect different providers. Transrating, a simpler process that only changes the bitrate of the same codec without altering the encoding format. Also known as bitrate adjustment, it’s used when you need to save bandwidth but keep the original audio structure intact. These two processes sound similar, but they’re not interchangeable—and mixing them up can hurt your call quality or spike your bandwidth costs.
Transcoding is like translating a book from English to Spanish: you’re rewriting the whole thing. It takes more processing power, adds delay, and can introduce artifacts—especially if you’re converting between lossy codecs like G.729 and Opus. That’s why many VoIP systems avoid it unless absolutely necessary. Transrating, on the other hand, is like adjusting the font size in that same book. The words stay the same, but you’re using less paper (or bandwidth). It’s faster, cheaper, and safer for real-time calls. Most modern VoIP networks use transrating to adapt to poor connections, while transcoding is reserved for connecting legacy systems or incompatible endpoints.
Here’s what you’ll find in the posts below: real-world examples of how transrating keeps calls clear during network hiccups, why transcoding can cause echo or dropped syllables, and how to spot which one your provider is using. You’ll also learn how codec packetization intervals, MOS scores, and PESQ metrics tie into these processes. Whether you’re managing a call center, setting up a remote team, or just trying to stop your calls from sounding like a robot, understanding transrating vs transcoding gives you real control over your audio quality.