Video calls connect us in the home office and across national borders. Functions such as screen sharing, video recording, conferencing or dialling into the telephone network can be implemented by us in high quality and integrated into industry solutions.
The WebRTC technology
WebRTC is an internet standard that provides real-time communication via audio/video directly in the browser or in mobile apps. It was standardised by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). The first implementation was published in 2011 and has since become stable and hardened in the market.
Today, various WebRTC implementations exist. The standard is supported by Google, Apple, Microsoft, Mozilla and Opera and can therefore run in every modern browser. The most popular implementation is the one from Google, which was published as an open source library. WebRTC is being continuously developed and new functions are being added regularly.
New dynamics
WebRTC is considered disruptive as it sparks a new dynamic in the previously rather sluggish telecommunications market. The technology enables new use cases for VoIP solutions and reduces implementation efforts. Various obstacles have been removed. For example, thanks to WebRTC, important codecs such as H.264, VP8 and Opus are freely available and the media data transmitted via the internet is strongly encrypted (DTLS-SRTP).
Opportunities
Software start-ups and SMEs can now also offer new types of communication solutions that were previously reserved for large telecommunications companies. The telecommunications companies, for their part, are challenged more. The opportunities for growth and the spirit of innovation have risen sharply.
Your contact
Sebastian Schmid
Sebastian has experience with WebRTC since 2013 and has developed one of the first mobile WebRTC apps ever: GoVideo with 60,000 downloads.
Testimonials
Functional components
Direct call
Efficient direct communication via audio/video between two participants (peer-to-peer) using powerful codecs such as Opus, VP8 / VP9 and H.264.
Conference
Conference room with a large number of participants. Scalable with Selective Forwarding Unit (SFU). Transmission adapts dynamically to the available network bandwidth thanks to simulcast.
Recording
Record and save conversations and conferences in the highest quality. Re-upload partial segments after network interruptions.
Data channel
High-performance data channel for transferring files and other data enables a variety of novel use cases.
Screen sharing
Share the screen or selected application windows.
Spatial audio
With 3D or immersive audio, the user is even more immersed in the soundscape.
Simulcast
Each participant receives the optimal video resolution that is supported by their device and allowed by their bandwidth. The dedicated control and management of media streams allows the application to be scaled to a large number of participants.
Security
WebRTC implements the highest security standards (DTLS-SRTP). The encryption of media data is an integral part of the solution and, unlike common VoIP applications, cannot be deactivated or bypassed.
NAT & firewall traversal
Every network is different. In large networks in the enterprise environment, in mobile networks or home networks, the network quality can be affected by a large number of factors (firewall, deep packet inspection, limited bandwidth, defective hardware, congestion, packet loss or high latency). In addition to the necessary technologies such as ICE/STUN/TURN, we also have the necessary experience and analysis tools.
Native development
WebRTC is primarily considered a browser technology. However, Google’s WebRTC stack is also available as an open source library and can be integrated and extended in native mobile apps.
WebRTC server
We have experience with various WebRTC servers and WebRTC-enabled PBXs such as Janus, Kurento, Asterisk or Freeswitch.
Janus consulting
The Janus WebRTC Server is the Swiss army knife for real-time communication. It offers a wide range of applications and can be extended via plug-ins. We can implement customised extensions for your specific use case.
Interoperability with legacy devices
Classic VoIP or RTP devices such as legacy PBXs or IP cams that were originally designed for use in the LAN can also be made fit for use in the cloud with WebRTC.
Further services
Mobile app development
The smartphone is our ubiquitous multifunctional tool. Specific apps for iOS and Android make our everyday and professional lives easier. Would you like to transfer your business case into the mobile world? We develop the optimal app for your individual application.
Conception & evaluation
Innovation and technology consulting for startups, spin-offs and the disruptive development of new business fields. Whether concept creation, UX design, a functional proof of concept or the final development of a market-ready app β we are there for you.
Convinced?
Come have coffee with us!