Skip to main content

On Kamailio World 2018, part II

In the first part of my brain dump about this year's edition of Kamailio World I focused mainly on testing. Core developers and application designers want to be able to test the behaviour of Kamailio-based architectures with minimal effort and fast feedback.

A different dimension to testing, that I haven't mentioned in my previous post, was related to Fuzz testing. There were two presentations focused on this: Sandro Gauci's (The easiest way to understand who Sandro is: listen on port 5060 on the public Internet and wait a couple of minutes. You'll see a SIP request from a tool called sipvicious (aka friendly-scanner), a penetration testing tool Sandro wrote (and others misuse)) and Henning Westerholt, historical member of the Kamailio community.

Sandro's presentation focused around fuzzing approaches for RTC in general (slides), while Henning was more specifically focused on Kamailio.

Fuzzing is a sophisticated technique to verify the robustness of a software application, by sending input that can vary greatly from the typical or expected usage. The objective is to find weaknesses that can lead to crashes or other malfunctions, so that they can be fixed. Of course testing a server like Kamailio is even trickier than testing an application that can read from a file. It is a fascinating topic.

Kamailio proved to be very robust: Henning reported an average of  about 1 message every 44 million required to make Kamailio misbehave. The video of Henning's presentation is here (by the way, Pascom have done a great work this year too, providing a flawless video streaming and recording. It feels like we are a little spoiled, because we give it for granted and barely notice all the work behind it).

In terms of learning opportunities for architects and administrators of Kamailio-based infrastructure, I found very valuable Daniel's presentations around high-level scripting (with KEMI) to build the routing logic (Video and slides).

Remember that Lua may not be the most popular - apparently - but it's the one estimated to give you performances closest to the native routing language.

Another valuable presentation was around the Least Cost Routing techniques that the Kamailio environment makes available. (Video, and slides). Some solutions use out of the box modules (like lcr, carrierroute, drouting), some are more indirect (pdt, mtree, dialplan, prefix_route), and others are a combination of them. Must-see if you're working in that area.

Another learning goldmine has been Lorenzo Miniero's (author of Janus, a WebRTC conferencing framework (this definition is mine)) lecture about Privacy, Security and Authentication for WebRTC. (Video and slides) Lorenzo does talk fast, but no word is spoken in vain. Worst case, you can watch the video at 0.5 speed (smile). Interesting the case of double encryption for media.



I guess there's enough for a part III in the near future! To be continued. 
Labels:

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.
Read more

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...
Read more

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...
Read more