Free online OPUS to GSM converter

1. Source format: Opus

Opus is a modern, highly efficient audio codec designed for speech and music alike. It is widely used in VoIP, web audio, streaming, gaming voice chat, and messaging apps because it delivers excellent quality at low bitrates and handles both narrowband speech and full-band music well. Opus files are commonly stored in an .opus container, often based on Ogg.

2. Target format: GSM

GSM audio usually refers to the GSM 06.10 speech codec stored in a .gsm file. It is an older, low-bitrate format intended primarily for speech, not music. Its main advantages are very small file size and broad support in legacy telephony and voice-processing systems. Typical use cases include voicemail archives, telephony workflows, and compatibility with older audio tools.

3. How to perform the conversion

The most reliable way to convert Opus to GSM is with FFmpeg, which supports both formats and gives you control over sample rate and channel layout.

Step 1: Install FFmpeg on your computer if it is not already available.

Step 2: Open a terminal or command prompt in the folder containing your Opus file.

Step 3: Run a conversion command such as:

ffmpeg -i input.opus -ar 8000 -ac 1 -c:a gsm output.gsm

This command reads input.opus, converts it to 8 kHz mono audio, and encodes it as GSM in output.gsm.

Step 4: Test the resulting file in your target application or player to confirm compatibility.

If you prefer a graphical workflow, you can also use desktop audio editors or converters that support FFmpeg-based export. In many cases, the process is File → Open, then File → Export or Save As, followed by choosing a GSM-compatible output format if available.

You can also convert the file on 101convert.com for free, without installing any software.

4. Notes on quality, compatibility, and limitations

Quality loss is significant when converting from Opus to GSM. Opus is designed for much higher fidelity, while GSM is a narrowband speech codec, so music, stereo content, and high-frequency detail will be heavily degraded.

Mono and 8 kHz sampling are typically required for GSM. If the source is stereo or higher sample rate, FFmpeg will downmix and resample during conversion. This is normal, but it means the output is suitable mainly for speech.

Not every player or editor supports .gsm files. If compatibility is important, verify that your destination system expects raw GSM 06.10 audio rather than a different telephony container.

For voice recordings, the conversion is usually fine; for music or general listening, GSM is not a good target format because the result will sound noticeably compressed and limited.