The need for quality audio & video

It’s safe to say that if you ever notice anything about voice or video, it’s usually not a positive. People will go crazy on poor media quality. Pops, clicks, hisses, artifacts, echo, delay, too soft, too loud, freezes or sync issues. We’ve all experienced these. We hate the introduction of these items because it detracts from what we’re trying to do. Communicate.

The mobile industry has long known what you probably have been a first hand witness to. Namely, the direct correlation between call quality and the length of the call. We’ve laughed at the rigid telco industry for their inflexibility as they bolted equipment to the floor. 100 years later, they’re still working to get voice quality on mark, but dismissing poor radio reception, voice does mostly work.

The various WebRTC working groups and bug trackers are hard at work on extremely boring items like echo cancellation, packet loss concealment, forward error correction, user media discovery. Tedious items in miniature that collectively add up to impact the experience. The noble goal though – simply not to be noticed by the end user.

This effort solely for P2P connections, what about those folks busy trying to create a service. Something that sits in the middle, disrupting the P2P connection. Existing and new conference platforms, recordings — all wanting to inject themselves into the transaction. My teeth clench as they glide over the topic of media quality.

GIPS (Global IP Sound) was for many the ‘gold’ standard for VoIP audio quality and this inclusion into WebRTC was a godsend. However, WebRTC still doesn’t deal terribly well with misbehaving Internet connectivity. The prioritization between a voice & video connection still not worked out. There remains some road ahead for WebRTC to develop.

The short is if you’re going to be mixing either the audio or video elements of WebRTC (beyond mesh networking) you’d be well heeded not to pass over the aspect of media quality, because that is indeed the product on offer.