Brief

All posts tagged Brief

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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Losing customers because of issues with your network service is a bad thing. Sure you can gather data and try to prevent, but isn’t it better to prevent issues in the first place? What are the most common pitfalls to look out for? What’s a good benchmark? What WebRTC-specific user experience elements should you spend your limited resources focusing on? No service can be perfect, so what is a reasonable error rate? These are all tough questions to answers without decent industry data.

Fortunately Lasse Lumiaho and Varun Singh from callstats.io have agreed to share some stats from their WebRTC monitoring service to help you answer these questions.  Their service does not monitor every WebRTC service,  but their 100+ customer base does provide a statistically meaningful sample to help give you some practical metrics for planning and comparison.

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I Think I'm Being Watched

There has been more noise about WebRTC making it possible to track users. We have covered some of the nefarious uses of WebRTC and look out for it before. After reading a blog post on this topic covering some allegedly new unaddressed issues a week ago I decided to ignore it after some discussion on the mozilla IRC channel. But this has some up on a the twitter-sphere again and Tsahi said ‘ouch’, here are my thoughts.

Claims

The blog post (available here) makes a number of claims about how certain Chrome behavior makes fingerprinting easier:

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Chrome, Firefox, and Edge are all on the same party line. Image from Pillow Talk (1959)

For the first time, Chrome, Firefox and Edge can “talk” to each other via WebRTC and ORTC. Check the demo on Microsoft’s modern.ie testdrive.

tl;dr: don’t worry, audio works. codec interop issue…

Feature Interoperability Notes
ICE yes Edge requires end-of-candidate signaling
DTLS yes
audio yes using G.722, Opus or G.711 codecs
video no standard H.264 is not supported in Edge yet
DataChannels no Edge does not support dataChannels

As a reader of this blog, you probably know what WebRTC is but let me quote this:

WebRTC is a new set of technologies that brings clear crisp voice, sharp high-definition (HD) video and low-delay communication to the web browser.

In order to succeed, a web-based communications platform needs to work across browsers. Thanks to the work and participation of the W3C and IETF communities in developing the platform, Chrome and Firefox can now communicate by using standard technologies such as the Opus and VP8 codecs for audio and video, DTLS-SRTP for encryption, and ICE for networking.

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Two weeks ago Philipp Hancke,  lead WebRTC developer of Talky and part of the &yet‘s WebRTC consulting team, started a series of posts about detailed examinations he is doing on several major VoIP deployments to see if and how they may be using WebRTC. Please see that post on WhatsApp for some background on the series and below for another great analysis – this time on Facebook Messenger. {“editor”: “chad hart“}

Last week, Facebook announced support for video chats in their Messenger app. Given that Messenger claims to account for 10% of global mobile VoIP traffic, this made in a very interesting target for further investigation. As part of the series of deconstructions, the full analysis (another fifteen pages, using the full range of analysis techniques demonstrated earlier) is available for download here, including the wireshark dumps.

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One of our first posts was a Wireshark analysis of Amazon’s Mayday service to see if it was actually using WebRTC. In the very early days of WebRTC, verifying a major deployment like this was an important milestone for the WebRTC community. More recently, Philipp Hancke – aka Fippo – did several great posts analyzing Google Hangouts and Mozilla’s Hello service in Firefox. These analyses validate that WebRTC can be successfully deployed by major companies at scale. They also provide valuable insight for developers and architects on how to build a WebRTC service.

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A couple of decades ago if you bought something of any reasonable complexity, odds are it came with a call center number you had to call in case something went wrong. Perhaps like the airline industry, economic pressures on contact centers shifted their modus operandi from customer delight to cost reduction. Unsurprisingly this has not done well for contact center public sentiment. Its no wonder the web came along to augment and replace much of this experience –  but no where near all it. Today, WebRTC offers a unique opportunity for contact centers to combine their two primary means of customer interaction – the web and phone calls – and entirely change the dynamic to the benefit of both sides.

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MS Elephant

What about IE?

As WebRTC has matured to a state where it’s first implementations are ready for companies to launch real services around it, the readiness of various companies to adopt WebRTC has fanned out quite a bit. Some are already charging ahead as early adopters, while others are playing it conservative. Of those in the conservative camp, one of the common doubts that gives them pause is: “What about IE?”

When speaking to those interested in WebRTC, but concerned about Internet Explorer (IE), many times we’ve tried to assure them not to worry: our friends in Redmond won’t be too far behind. We often point to the undeniably significant contributions from Microsoft to WebRTC, especially considering that they bring to the table two titans of VoIP industry (Lync and Skype). We highlight some of their early IE WebRTC demos (using beta code) as signs of progress. We’ve rationalized the absence of  a Microsoft equivalent to what Chrome and Firefox are shipping, by noting the slower release cycle for IE. However, we’ve come to realize that to some, IE support is a really big deal.

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As discussed in previous posts, WebRTC standards do not specify a signaling protocol. In general this decision is positive by giving developers the freedom to select (or invent) the protocol that best suits the particular WebRTC application’s needs. This can also reduce the time to market since standards compliance-related tasks are minimized. WebRTC media and data protocols from the provider to the user are standardized, so the lack of a standardized signaling protocol does not hurt interoperability of subscribers within the same network service. The calling party just has to have a URL from the called party to download its web app and to establish a WebRTC session with them. They both connect to the same web server and will then utilize the same signaling scheme. This is a new paradigm that is often difficult to embrace for traditional telephony developers who are used using the SIP protocols for handling all signaling, including the User to Network Interface (UNI) and Network-to-Network Interface (NNI).

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