Skip to main content

The interesting case of Lua in RTC world

An interesting pattern that caught my attention is the role that Lua is gaining in the RTC (Real-Time Communications) world.

Lua is a small-footprint programming language, powerful while keeping a simple syntax.

I’ve been using Lua to script dialplan actions for FreeSWITCH since about 2014. It has provided me with a way to define relatively complex logic and speed up the definition of FS’ behaviour.

Delegating this type of logic to a scripting language had several advantages, such as:
  • It’s easier to read and understand than native dialplans or native routing logic.
  • Makes unit testing of the dialplan possible/easier.
  • Allows changing some pieces of logic easily, in many cases preventing expensive reload of modules or restart of applications.

I’ve been using Lua for Kamailio as well. Kamailio is an open source programmable SIP Proxy. In a specific case, some bits of the routing logic required regex processing and was expecting to change often: an ideal case for an external script to do that work.
When the logic changes, it’s sufficient to instruct Kamailio to reload the script, and from that moment on the new requests being processed will use the new logic.

Recent versions of Kamailio though add a framework called KEMI. This opens up new possibilities, and also provides support for many other scripting languages, python being the most popular, JS, Squirrel. Still, Lua appears to provide the fastest implementations (with no observable performance degradation) while others have limitations. Python, as you can imagine, provides a rich set of functions and libraries, but it’s not as performant and the reload mechanism currently has some issues.

Wireshark, a tool to capture and analyse network activity, exposes a useful API for Lua. You can use the API to define your own Wireshark dissector (which you’ll need to install as a plugin). This has performance limitations in comparison with dissectors written in C - and so it’s recommended for prototyping only - but still can solve your problem perfectly. Out of need, I wrote a Wireshark dissector for HEP, a binary protocol used in the Homer environment. Homer is an open source framework for the monitoring and analysis of Real-Time Communications.

Last weekend a new interesting case was presented by Lorenzo Miniero at FOSDEM. The target application was Janus, an open source framework to build WebRTC gateways.

Janus allows to build applications by defining the transport and business logic as plugins, on top of the core that implements the WebRTC stack.
It’s written in C and so far users needed to write plugins with that language. The Janus developers have introduced the possibility to write plugins in Lua.

In his presentation Lorenzo explains also in detail what approach is best to use for a Real-Time application to interact with single threaded language like Lua in an asynchronous context.

Just a funny note: Lua uses a double dash for commenting out a line: '--'. Be careful when you watch diffs in a terminal because a removed comment will start with '- --‘ and may not the easiest thing to interpret (experiences may vary depending on the terminal, of course!).



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...