If you’re asking this question, then chances are you either have an existing SIP infrastructure and are looking for a way to interconnect with Web Real-Time Communications (WebRTC) - enabled endpoints, or you have an existing web application and are looking for a way to interconnect with the telephone network.
In both cases, the goal is the same — interconnection — and that’s what SIP is all about.
Before we can answer the main question, though, we need to understand clearly what SIP actually is, and to do that, we need to get a little bit technical.
It takes trust to place your confidential business information and sensitive conversations into the care of video conferencing technology. Especially when you hear about security breaches in the news nearly every day.
In 2019, the group FaceTime bug was an example of how video conferences could become compromised. Although in that case the FaceTime group calling feature was disabled and a fix was quickly released, it raised questions about video conferencing security.
It is truly an exciting time for WebRTC developers, especially given the rapid adoption of extended reality (XR) collaboration platforms by Fortune 500 companies. Extended reality, the umbrella term encompassing AR, VR, and XR, has become a technology capable of wielding the power of flexible WebRTC-based live video, which in turn unlocks the capabilities of real-time multi-party live video and audio streaming.
As such, extended reality applications are bringing value to organizations seeking to improve their bottom lines and open up new business opportunities. …
End-to-end (E2E) encryption is a bit of a buzzword these days. Everyone wants it and every company is jumping into the ring to claim that they have it. It makes sense. Who doesn’t want a completely unhackable application? However, end-to-end encryption (especially for media within a browser) is extremely new and some limitations are often glossed over.
End-to-end (E2E) encryption in video conferencing is a way to secure data that prevents third parties or intermediary servers (SFUs, TURN Servers, Gateway, etc.) from accessing or tampering with it at every hop along the media pipeline.
One easy way to think of…
An often-overlooked and perhaps under-appreciated feature, sending and receiving text-based messages is a critical piece of many WebRTC-based applications. Audio and video tend to steal the limelight, but the ability to exchange messages adds an extra layer of expressiveness and interactivity that is simple and reliable, even on old devices and slow networks.
In some apps, like Slack, text messaging is the primary feature, while audio and video communication take a secondary role. …
The demand for on-demand content is skyrocketing. High-traffic sites like YouTube, Vimeo, and Netflix stream billions of pre-recorded videos every single day. A generation of “cord-cutters” walking away from expensive television subscription services combined with exponential growth in internet speeds and new opportunities for interactivity within a live broadcast has driven many live content producers to distribute online.
How to Reduce Latency
As the drive towards interactivity increases, so does the demand to reduce latency on the generation and distribution of live content. Traditional techniques using HTTP streaming, which have been adapted for live broadcast, generally involve latency that exceeds…
WebRTC has emerged as an undeniable force in the modern communication and video conferencing landscape, making a significant impact on industries that depend on real-time communications. Industries ranging from education to telecommunications have seen WebRTC become an industry standard for developing cost-effective and user-friendly video streaming technology. In a world where plugins were viewed increasingly as annoying and intrusive security risks, web developers suddenly had a brand new plugin-free tool in their belt allowing them to create immersive applications that brought people together in new and exciting ways. …
Chief Technology Officer for LiveSwitch Inc