Skip to main content

FOSDEM 2015 - part I

It's that time of the year when experts of Open Source software meet in frosty Brussels for two intense days of talks, conversations, and a good quantity of beer: FOSDEM.

Being just 2 hours away from London with the Eurostar, the relevance/effort ratio is very high. Additionally, the event is free and held over the weekend, so it has a low impact on the normal job activity (although on the other hand it does have some on your family time, but you can't have everything, unless your kids are big enough to join you, which I'd recommend).

Right after settling down in a lovely flat (AirBnB is a great choice) with three friends I started planning the sessions to follow. There are about twenty parallel sessions, so you must cherry pick. For day 1 I was oriented towards Configuration Management and Lightning Talks tracks. Day 2 had Virtualisation and Testing And Automation in my radar.

As it turns out, FOSDEM is such a success that the rooms are filled incredibly fast and many have even long queues outside. It was definitely the case for the Configuration Management dev room, which I had to skip in favour of Infrastructure As A Service.

So I got a basic understanding of Apache Mesos, for resource management of distributed systems, and GlusterFS, a distributed file system and general-purpose storage platform.
Apache Mesos will come back to attention on Day 2, when I'll learn more about CoreOS Rocket for container orchestration (more on this on a following post).

This year there wasn't any track dedicated to RTC (Real Time Communication), but nevertheless the topic was around a lot (not to mention the Internet Of Things track, which I gather has been very successful).


The Python devroom though offered an interesting cross between Python and WebRTC, with Saúl's "Python, WebRTC and you" (slides here). As the reader may know already, WebRTC doesn't mandate a signalling protocol, so this talk has been a good opportunity to show how conceptually easy is to interact with WebRTC's APIs (using rtcninja as wrapper), while at the same time managing signalling between browsers with a combination of Python3 and asyncio.

This talk alone, from a teaching perspective, is very valuable. You can see:
- The power of WebRTC: as long as the browser is compatible, you're just a few dozens of lines of code away from a new application.
- You can use the web server you prefer: web content servicing and signalling are just tools around the RTC.
- Unless you need some complex feature, you can leverage pure P2P and remove the server infrastructure from the equation.
- WebSocket support is increasing [1], making the life of WebRTC architects easier.

Of course things get more complicated for multiparty conferences, but there's something about this on a following post.

[1] The popular node.js platform has an ad-hoc module for WebSocket support too. I wrote about it and its usage inside a docker container in a previous post.


Popular posts from this blog

Troubleshooting TURN

  WebRTC applications use the ICE negotiation to discovery the best way to communicate with a remote party. I t dynamically finds a pair of candidates (IP address, port and transport, also known as “transport address”) suitable for exchanging media and data. The most important aspect of this is “dynamically”: a local and a remote transport address are found based on the network conditions at the time of establishing a session. For example, a WebRTC client that normally uses a server reflexive transport address to communicate with an SFU. when running inside the home office, may use a relay transport address over TCP when running inside an office network which limits remote UDP targets. The same configuration (defined as “iceServers” when creating an RTCPeerConnection will work in both cases, producing different outcomes.

VoIP calls encoded with SILK: from RTP to WAV

SILK is a codec defined by Skype, but can be found in many VoIP clients, like CSipSimple . It comes in different flavours (sample rates and frame sizes), from narrowband (8 KHz) to wideband (24 KHz). Since Wireshark doesn't allow you to decode an RTP stream carrying SILK frames, I was curious to find a programmatic way to do it. In fact, this has also allowed to me to earn a "tumbleweed" badge in stackoverflow . You may argue that a Wireshark plugin would be the right solution, but that's probably for another day. Initially I thought it was sufficient to read the specification for RTP payload when using SILK ; the truth is that I had to reverse engineer a solution by looking at SILK SDK's test vectors. There, I discovered that a file containing SILK audio doesn't have the file header indicated in the IETF draft ("!#SILK"), but a slightly different one ("!#SILK_V3"). More importantly, each encoded frame is not preced...

Extracting Opus from a pcap file into an audible wav

From time to time I need to verify that the audio inside a trace is as expected. Not much in terms of quality, but more often content and duration. A few years ago I wrote a small program to transform a pcap into a wav file - the codec in use was SILK. These days I'm dealing with Opus , and I have to say things are greatly simplified, in particular if you consider opus-tools , a set of utilities to handle opus files and traces. One of those tools, opusrtp , can do live captures and write the interpreted payload into a .opus file. Still, what I needed was to achieve the same result but from a pcap already existing, i.e. "offline". So I come up with a small - quite shamlessly copy&pasted - patch to opusrtc, which is now in this fork . Once you have a pcap with an RTP stream with opus (say in input.pcap ) you can retrieve the .opus equivalent (in rtpdump.opus ) with: ./opusrtp --extract input.pcap Then you can generate an audible wav file with: ./opusd...