It’s day two of the WebRTC Conference & Expo and, as expected, there are plenty of new WebRTC announcements being made. For example, Requestec has stated that its Zenon WebRTC-enabled gateway is now able to mix Opus with other audio codecs to form a single, low latence, low bandwidth stream that can connect participants all around the world in a clear conversation free from delays and loss of volume.
Opus is the wideband audio codec of choice for WebRTC communications. Totally open, highly versatile and royalty free, the codec is currently unmatched for transmitting audio over the Internet. Aside from Opus, WebRTC audio also relies on the legacy narrowband G.711 codec which doesn’t provide the same quality audio experience as Opus.
“As far as we are aware, competitors in this space are only mixing WebRTC G.711, which at 8kHz is not good enough for many applications including language learning or business-critical communications,” said Marek Zwiefka-Sibley, Requestec's head of Business Development, in a statement.
Requestec’s Zenon gateway goes one step beyond merely supporting Opus by mixing the WebRTC codec with Flash’s Speex codec and SIP into a single stream. This offers conference participants the best possible experience as they only need bandwidth for a single download connection.
By comparison, scalable video coding (SVC) technology sends a separate audio stream for each conference member, which can be an issue in low-bandwidth or congested wireless environments.
“While we look forward to SVC coming to WebRTC with VP9, we don’t believe that the SVC architecture is right for many deployments,” said Requestec CTO Ben Weekes. “It makes sense for mass-market free services where the service provider’s CPU is the bottleneck, but for premium services such as education and health where customer satisfaction at any bandwidth is the priority, then mixing the codecs into a single stream provides the best experience.”
Edited by
Ryan Sartor
