Skip to main content

ejabberd clustering: a clever idea for Mnesia replication setup

The process of configuring a cluster of ejabberd entities serving the same domain is apparently simple (considering the ejabberd Installation and Operation Guide).

Assuming you have a node running, you create a second node with the same Erlang cookie, then setup Mnesia replication between the two. Assuming both ejabberd instances have a similar configuration setup and can connect to each other over the network, you're pretty much done.
This can be generalized and made N times to build your cluster.

ejabberd uses Mnesia as its internal DB. Although you can easily move to MySQL as storage backend, it's important to note that  Mnesia is still necessary to successfully build the cluster.

ejabberdctl is the control script that can be used to start, stop, restart ejabberd. It can be also used to attach a debug console to a running ejabberd instance, or execute any command exposed by an ejabberd module.

The idea presented in Easy ejabberd clustering procedure is to extend ejabberdctl with an additional command ("attach") which hides the complexity of connecting to an Erlang node and setup Mnesia replication. I've found it an interesting idea, firstly because it becomes easier to automate the clustering configuration, secondly because goes in the right direction of hiding the setup complexities behind a single control tool.


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...