Summary: Kixie's WebRTC client (Kixie PowerCall Dialer) uses the latest technology to decrease the networking, CPU, and memory requirements for placing a call. Using the OPUS and PCMU codecs also ensure that users receive the highest possible level of voice call clarity. A typical Kixie call will use 50kbps to 200kbps based on the number of call legs (ie transferrered calls or call coaching).


How Much Bandwidth Does a Kixie Call Use?

 Typically, a Kixie call uses anywhere between 50 KB/s and 200 KB/s of bandwidth for a single call. The amount stems from which codec we use for you call. Kixie dynamically chooses codec usage based on your computer hardware (CPU performance + GPU performance, if any).


What Does This Mean?

Opus’ adaptability and robustness makes the codec very suitable for VoIP applications running on standalone software or web browsers. 

Based on our carrier's benchmark tests OPUS provides improved call clarity as well as increased CPU performance. Our carrier also has optimised the sampling rates leading to decreases in browser sampling frequency and thus allowing better performance on the browser.

Below are audio samples that compare how Opus and PCMU deal with varying degrees of packet loss. The audio samples with 0% packet loss sets the baseline for comparison. With increases in packet loss, Opus’s superiority stands out. Compared to PCMU at 30% packet loss, Opus truly comes out ahead. Even at such a high level of packet loss, Opus still delivers audio quality that preserves quality and ensures that the dialogue is comprehensible.

Packet Loss Audio Samples
0% Opus
10% Opus
25% Opus
30% Opus


OPUS also decreases jitter, latency, and packet loss. And during poor network connections, your users will experience better voice quality. To give you an example of this, instead of utilizing 100 kbps bandwidth with other WebRTC SDK codecs, Opus only requires 50 kpbs of bandwidth.

Unfortunately, slow-speed connections and congestion are unavoidable, but the VoIP community is constantly working to help solve issues of packet loss with new technology, such as the Opus codec. 

What is Opus?

Opus is an open source audio codec that’s optimized for speech and music transmission over the internet. Audio codecs are software that compress and decompress digital audio signals for transmission. These codecs come in the form of mathematical algorithms and are graded on its ability to retain audio quality while encoding and compressing audio signals.

Compared to other codecs, Opus is highly effective at reducing bandwidth consumption and CPU usage during audio transmission while maintaining high-fidelity audio signals. That’s why Opus is known for its ability to handle a wide variety of VoIP (voice over IP) audio applications including conferencing, CRMs, help desks, and click-to-call applications.

(ex Call Clarity using OPUS)

Chart opus-codec-support-comparison