sfu

All posts tagged sfu

Multi-party calling architectures are a common topic here at webrtcHacks, largely because group calling is widely needed but difficult to implement and understand. Most would agree Scalable Video Coding (SVC) is the most advanced, but the most complex multi-party calling architecture.

To help explain how it works we have brought in not one, but two WebRTC video architecture experts. Sergio Garcia Murillo is a long time media server developer and founder of Medooze. Most recently, and most relevant for this post, he has been working on an open source SFU that leverages VP9 and SVC (the first open source project to do this that I am aware of). In addition, frequent webrtcHacks guest author and renown video expert Gustavo Garcia Bernando joins him.

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Dealing with multi-party video infrastructure can be pretty daunting. The good news is the technology, products, and standards to enable economical multiparty video in WebRTC has matured quite a bit in the past few years. One of the key underlying technologies enabling some of this change is called simulcast. Simulcast has been an occasional sub-topic here at webrtcHacks in the past and it is time we gave it more dedicated attention.

To do that we asked Oscar Divorra Escoda, Tokbox’s Senior Media Scientist and Media Cloud Engineering Lead to walk us through it. Tokbox was one of the first to market with a SFU and Oscar shares some of his learnings below.

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Earlier this month Fippo published a post analyzing Slack’s new WebRTC implementation. He did not have direct access or a team account to do a thorough deep dive – not to mention he is supposed to be taking some off this month. That left many with some open questions? Is there more to the TURN network? How does multi-party calling work? How exactly is Slack using the Janus gateway? Fortunately WebRTC has an awesomely active and capable community that quickly picked up the slack (pun intended).

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Atlassian’s HipChat acquired BlueJimp, the company behind the Jitsi open source project. Other than for positive motivation, why should WebRTC developers care? Well, Jitsi had its Jitsi Video Bridge (JVB) which was one of the few open source Selective Forwarding Units (SFU) projects out there. Jitsi’s founder and past webrtcHacks guest author, Emil Ivov, was a major advocate for this architecture in both the standards bodies and in the public. As we have covered in the past, SFU’s are an effective way to add multiparty video to WebRTC. Beyond this one component, Jitsi was also a popular open source project for its VoIP client, XMPP components, and much more.

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Gustavo Garcia Bernardo

Gustavo Garcia Bernardo

WebRTC and its peer-to-peer capabilities are great for one-to-one communications. However, when I discuss with customers use cases and services that go beyond one-to-one, namely one-to-many or many-to-many, the question arises: “OK, but what architecture shall I use for this?”. Some service providers want to reuse the multicast support they have in their networks (we are having fun doing some experiments with this), some are exploring simulcast-based solutions, others are considering centralised solutions like MCUs/mixers, and a bunch of them are simply willing to place the burden on the endpoint by using some variation of a mesh-based topology.   The folks at TokBox (a Telefónica Digital company) have great experience with multiparty conferencing solutions.  I thought it would be great to have my friend Gustavo Garcia Bernardo (Cloud Architect at TokBox) to share here his take on the topic.

At TokBox, Gustavo is responsible for architecture, design, and development of cloud components. This includes Mantis, the cloud-scaling infrastructure for the OpenTok, which uses the WebRTC platform. Before joining TokBox, Gustavo spent more than 10 years building VoIP products at Telefónica and driving early adoption of WebRTC in telco products. In fact, I’ve known Gustavo for 8 years now and the first time I met him it was preparing a proposal for a European Commission-funded research project on P2PSIP. Since then we’ve been collaborating in the IETF doing some work in the context of P2PSIP, ALTO and SIP related activities. A couple of years ago, while I was working with Acme Packet (now Oracle), we worked together designing and launching Telefonica’s Digital TuMe and TuGo.  Lately we have both shifted our focus towards WebRTC.

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