AAC or MP3 are not the best choice of codecs for ultra-low bandwidth transmissions. HE-AAC v2 may be somewhat usable, but in your case I'd use a proper speech codec with higher efficiency.

Opus is the best option. It is available in FFmpeg through libopus. In fact, Opus is not just made for speech; it offers hybrid encoding for both speech and music.

Example:

ffmpeg -i <input> -c:a libopus -ac 1 -ar 16000 -b:a 8K -vbr constrained out.opus

Here, -ac sets the output to mono, -ar sets the sampling rate to 16 kHz, and -b:a sets the bitrate to 8 kBit/s. The constrained variable bitrate mode is used here. In principle, it's not strictly necessary to downsample and downmix to mono with ffmpeg, as that is something libopus will do on its own to reach the specified bitrate target.

Some further recommendations are given here. Note that with Opus, 6–8 kBit/s is usable range for (mono, lower sample rate) speech, but not for music.

You'll find an interesting comparison of different codecs and their bitrate/quality curve on the Opus website:

I should add that this figure is an indication only; it's compiled from different test results and anecdotal knowledge.

AAC or MP3 are not the best choice of codecs for ultra-low bandwidth transmissions. Use a proper speech codec with higher efficiency.

Opus is the best option. It is available in FFmpeg through libopus. In fact, Opus is not just made for speech; it offers hybrid encoding for both speech and music.

Example:

ffmpeg -i <input> -c:a libopus -ac 1 -ar 16000 -b:a 8K -vbr constrained out.opus

Here, -ac sets the output to mono, -ar sets the sampling rate to 16 kHz, and -b:a sets the bitrate to 8 kBit/s. The constrained variable bitrate mode is used here. In principle, it's not strictly necessary to downsample and downmix to mono with ffmpeg, as that is something libopus will do on its own to reach the specified bitrate target.

Some further recommendations are given here. Note that with Opus, 6–8 kBit/s is usable range for (mono, lower sample rate) speech, but not for music.

You'll find an interesting comparison of different codecs and their bitrate/quality curve on the Opus website:

I should add that this figure is an indication only; it's compiled from different test results and anecdotal knowledge.

AAC is not the best codec for ultra-low bandwidth transmissions. Use a proper speech codec with higher efficiency such as Opus, which is available in FFmpeg through libopus.

Example:

ffmpeg -i <input> -c:a libopus -ac 1 -ar 16000 -b:a 8K -vbr constrained out.opus

Here, -ac sets the output to mono, -ar sets the sampling rate to 16 kHz, and -b:a sets the bitrate to 8 kBit/s. The constrained variable bitrate mode is used here.

Some further recommendations are given here. Note that with Opus, 6–8 kBit/s is usable range for (mono, lower sample rate) speech, but not for music.

AAC or MP3 are not the best choice of codecs for ultra-low bandwidth transmissions. HE-AAC v2 may be somewhat usable, but in your case I'd use a proper speech codec with higher efficiency.

Opus is the best option. It is available in FFmpeg through libopus. In fact, Opus is not just made for speech; it offers hybrid encoding for both speech and music.

Example:

ffmpeg -i <input> -c:a libopus -ac 1 -ar 16000 -b:a 8K -vbr constrained out.opus

Here, -ac sets the output to mono, -ar sets the sampling rate to 16 kHz, and -b:a sets the bitrate to 8 kBit/s. The constrained variable bitrate mode is used here. In principle, it's not strictly necessary to downsample and downmix to mono with ffmpeg, as that is something libopus will do on its own to reach the specified bitrate target.

Some further recommendations are given here. Note that with Opus, 6–8 kBit/s is usable range for (mono, lower sample rate) speech, but not for music.

You'll find an interesting comparison of different codecs and their bitrate/quality curve on the Opus website:

I should add that this figure is an indication only; it's compiled from different test results and anecdotal knowledge.