G.722 Codec: High-Quality VoIP Audio Explained

When you hear a VoIP call that sounds almost like the person is in the room, you’re likely hearing G.722, a wideband audio codec that captures voice frequencies from 50 Hz to 7 kHz, delivering near-CD quality sound. Also known as 7 kHz audio codec, it’s the reason your business calls don’t sound muffled or robotic anymore.

G.722 isn’t just an upgrade—it’s a shift in what voice communication can be. Unlike older codecs like G.711, which only handles 300 Hz to 3.4 kHz, G.722 captures the full range of human speech, including subtle tone changes and consonants like "s" and "th" that get lost in narrowband systems. This matters because clarity isn’t just nice to have; it reduces misunderstandings in customer service, legal calls, and remote team huddles. It works best when paired with sufficient bandwidth and a stable network, which is why it’s often used alongside UDP, the preferred protocol for real-time voice traffic because it avoids delays caused by TCP’s error-checking and jitter buffers, tools that smooth out network delays to keep audio flowing without breaks.

You’ll find G.722 in enterprise VoIP systems, video conferencing platforms, and cloud phone services that prioritize call quality over bandwidth savings. It uses about 64 kbps per call—more than G.729’s 8 kbps, but far less than G.711’s 80 kbps—making it a smart middle ground. If your internet connection can handle it, G.722 gives you the clearest voice possible without needing expensive hardware. Most modern VoIP phones and softphones support it out of the box, and it’s especially useful for teams that handle frequent international calls, customer support, or remote collaboration where tone and nuance matter. The posts below cover everything from how G.722 compares to other codecs, to real-world bandwidth needs, network setups that maximize its performance, and why some businesses still avoid it despite its advantages.