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Articles and security news about vulnerabilities and attacks affecting VoIP and WebRTC by Enable Security.

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Bug discovery diaries: uncovering sngrep overflow issues with blackbox fuzzing

Executive summary (TL;DR) During OpenSIPIt, we crashed sngrep by mistake while briefly fuzzing OpenSIPS. Later on we setup a docker environment to reproduce the issue, identified the actual bugs and reported them upstream. If you want to learn the simple steps to do this, you actually have to read the rest of the post :-) sngrep crash during the live OpenSIPit event Last year we participated in OpenSIPIt’s interoperability testing event which was held between the 14th and 15th of September 2020.…

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SIPVicious PRO beta release contains SIP fuzzer and better automation

We just made SIPVicious PRO v6.0.0-beta.1 available to our beta testers. This latest release brings a new SIP fuzzer and enhancements for automation to your favourite RTC offensive security toolset. We have the following highlights with this release: New fuzzing tools - sip fuzz method. This used to be in a separate internal tool called gasoline (see our toolset page); this now been polished and has joined the SVPRO toolset; this has been used to identify vulnerabilities in Kamailio (advisory), sngrep (advisory 1 and 2) and other SIP servers.…

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How doing QA testing for SIPVicious PRO led to an Asterisk DoS

Executive summary (TL;DR) While heavily testing SIPVicious PRO for bugs, we encountered an unexpected crash in Asterisk. We reported this to the Asterisk team, who recently issued a fix. If you’re a vendor, you too can beta test SIPVicious PRO! How the Asterisk crash was found We test our software as much as we can because, like any other software, ours contains bugs too! When it comes to SIPVicious PRO, one of our quality assurance tests is to run it against instances of Asterisk and Kamailio and check for expected results.…

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ClueCon Weekly with Sandro Gauci, demonstration of SIP Digest Leak

Published on Oct 16, 2020 in , ,

ClueCon weekly is a regular video by the people behind Freeswitch and Signalwire, hosted by the very friendly David Duffet. I had the pleasure of recording an interview and a presentation with David a few weeks back. If you would like a summary of what the video chat was about, scroll down to the points below. Otherwise, hope you enjoy the chat as much as I did! Summary Here’s an outline of what went on:…

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RTC Security chat at Kamailio World Online with Daniel and Olle

It’s been a month already since the Kamailio World RTC security chat! The conversation included Daniel-Constantin Mierla and Olle E. Johansson from the Kamailio project and myself. Daniel is the lead developer of Kamailio, can be found at ASIPTO while Olle is behind Edvina.net. If you don’t have time to watch the entire conversation, the following is my summary of this discussion: Introductions and discussions After introductions from Daniel, I took lead to briefly mention what we at Enable Security have been up to, including our work on SIPVicious PRO, our research on WebRTC security especially regarding the TURN server abuse vulnerability, our work on DoS in VoIP and WebRTC infrastructure and finally, research on how Kamailio may be (mis)configured to introduce vulnerabilities.…

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The great Kamailio security debate and some misconceptions debunked

Published on Sep 22, 2020 in ,

Introduction The Kamailio community has always been very welcoming to us since our first connection in 2015 where I gave a dangerous demo showing the open-source version of SIPVicious scanning the Internet and discovering all sorts of SIP devices. Since then, we’ve been contributing through presentations at Kamailio World each year, highlighting various security concerns for the RTC community and the occasional security report and advisory urging people to upgrade their Kamailio.…

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Smuggling SIP headers past Session Border Controllers FTW!

Published on Sep 1, 2020 in , , ,

Executive summary (TL;DR) SIP Header smuggling is a thing; in some cases it may be super-bad. It affected Kamailio and we have published a Github project to easily demonstrate and test this for yourself. Kamailio has since fixed the issue in release 5.4.0 but similar issues are likely to affect other SBCs. Usage of special SIP headers When it comes to trusted SIP networks, one of the primary ways that information is passed across different hops is through SIP headers.…

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Kamailio World Online SIP and VoIP Security Panel

On 2nd September, 14:00-14:30 Berlin time, the author of this post is joining Olle E. Johansson to chat at Kamailio World online about (guess what?) SIP and VoIP security, and recommendations on how working from home impacts security. I very much look forward to our discussions that will be streamed live on the Kamailio World youtube channel! My arguments will likely be turned into an opinion piece later on, but they’ll likely steer towards the following thoughts:…

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Bug bounty bout report 0x01 - WebRTC edition

Published on Jun 16, 2020 in , ,

Read the full report here. In April 2020, in between SIPVicious PRO development and VoIP Pentesting and WebRTC, we dedicated some days to bug bounties and vulnerability disclosure programs to see what comes out of it. Our focus was on those that have WebRTC infrastructure in scope. In the end, we reported 3 vulnerabilities to 4 different vendors, for 6 different products. So finally, after making sure that the affected vendors have addressed these security issues and have agreed with publication, we are putting out a compiled report!…

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Attacking a real VoIP System with SIPVicious OSS

Published on Jun 8, 2020 in , ,

Recently, we put out a target server on the Internet at demo.sipvicious.pro which hosts a Kamailio Server handling SIP over UDP, TCP, TLS as well as WebSockets. Behind that, the observant reader will soon discover that an Asterisk server handles the voicemail and echo services. This is actually a fully functioning (real) VoIP system that’s ready to be attacked. Therefore, in combination, these software packages allow us to reproduce a number of common security vulnerabilities affecting VoIP and WebRTC systems.…

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