What is Codec?
A codec is the software that compresses and decompresses voice audio for transmission over the internet in a VoIP call, balancing sound quality against bandwidth use.
Codec Explained
A codec (coder-decoder) is the algorithm that converts analog voice into compressed digital data for transmission and then decodes it back into audio at the other end of a VoIP call. The choice of codec is a trade-off between audio quality and bandwidth: high-fidelity codecs like Opus and G.722 deliver wideband HD voice but use more data, while efficient codecs like G.729 compress heavily for low-bandwidth connections at some cost to quality, and G.711 offers clear audio with minimal compression but higher bandwidth. WebRTC, which powers browser VoIP, uses the Opus codec by default, giving adaptive, high-quality audio that adjusts to network conditions. Connectoo's browser calling runs over WebRTC with Opus, and the mobile apps handle codec negotiation automatically, so teams get clear VoIP audio without configuring anything. Connectoo calling is VoIP only — WebRTC in the browser plus iOS and Android, numbers in 60+ countries, no SIM-based calling. As with all Connectoo calls, codec quality is just the surface: every call is logged with native WhatsApp chats and SMS in one built-in CRM with auto follow-up workflows, the cross-channel wedge that call-only providers like OpenPhone, JustCall, and Exotel cannot match. Connectoo serves teams of every size.
How Connectoo Implements Codec
Connectoo includes features directly related to codec. Explore these capabilities to see how the concept works in practice within a unified communication CRM.
Related Terms
These glossary terms are closely related to codec. Understanding them together gives you a more complete picture of communication CRM concepts.
Jitter
Jitter is the variation in the arrival time of voice data packets in a VoIP call, which can cause choppy or garbled audio if it is too high.
VoIP
VoIP (Voice over Internet Protocol) is a technology that transmits voice calls over the internet instead of traditional phone lines, letting businesses place and receive calls from a browser or mobile app.
WebRTC
WebRTC (Web Real-Time Communication) is an open standard that lets web browsers make voice and video calls directly, without plugins or separate apps.
SIP (Session Initiation Protocol)
SIP (Session Initiation Protocol) is the signaling protocol used to set up, manage, and end voice and video calls over the internet in VoIP systems.
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