A codec (coder-decoder) is the algorithm that compresses raw audio or video for transmission and reconstructs it on the receiving side. Raw media is far too large to send over a network in real time, so the codec's job is to shrink it dramatically while preserving enough fidelity for the clinical conversation, then rebuild a watchable, audible signal at the other end. The compromise a codec strikes — among picture or sound quality, the CPU it burns, and how widely it is supported — is the core engineering trade-off.
WebRTC's working set is well established. For audio, Opus is the standard workhorse. For video, the common choices are VP8, VP9, H.264, and increasingly AV1, the newer codec that compresses more efficiently at the cost of more processing. A crucial practical factor is hardware support: when a device can encode or decode a codec in dedicated silicon rather than software, battery drain and heat drop sharply and older or cheaper phones can keep up; without it, the same codec can overwhelm a modest device.
For a telemedicine product the clinical wins rarely come from exotic codec choices. They come from selecting widely supported profiles so that the patient on a years-old phone or an uncommon browser can still connect reliably, and from letting WebRTC's bandwidth adaptation reduce quality gracefully rather than dropping the call. The common mistake is chasing a newer codec like AV1 for its compression headline and then discovering that a meaningful share of your patient population lacks hardware support, turning an efficiency win into stutter, heat, and battery complaints on exactly the low-end devices that needed help most.
