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

Extracting RTP streams from network captures

I needed an efficient way to programmatically extract RTP streams from a network capture. In addition I wanted to: save each stream into a separate pcap file. extract SRTP-negotiated keys if present and available in the trace, associating them to the related RTP (or SRTP if the negotiation succeeded) stream. Some caveats: In normal conditions the negotiation of SRTP sessions happens via a secure transport, typically SIP over TLS, so the exchanged crypto information may not be available from a simple network capture. There are ways to extract RTP streams using Wireshark or tcpdump; it’s not necessary to do it programmatically. All this said I wrote a small tool ( https://github.com/giavac/pcap_tool ) that parses a network capture and tries to interpret each packet as either RTP/SRTP or SIP, and does two main things: save each detected RTP/SRTP stream into a dedicated pcap file, which name contains the related SSRC. print a summary of the crypto information exchanged, if available. With ...
Read more

Testing SIP platforms and pjsip

There are various levels of testing, from unit to component, from integration to end-to-end, not to mention performance testing and fuzzing. When developing or maintaining Real Time Communications (RTC or VoIP) systems,  all these levels (with the exclusion maybe of unit testing) are made easier by applications explicitly designed for this, like sipp . sipp has a deep focus on performance testing, or using a simpler term, load testing. Some of its features allow to fine tune properties like call rate, call duration, simulate packet loss, ramp up traffic, etc. In practical terms though once you have the flexibility to generate SIP signalling to negotiate sessions and RTP streams, you can use sipp for functional testing too. sipp can act as an entity generating a call, or receiving a call, which makes it suitable to surround the system under test and simulate its interactions with the real world. What sipp does can be generalised: we want to be able to simulate the real world tha...
Read more