Thursday, August 6, 2020

Exploiting Android Messengers with WebRTC: Part 3

Posted by Natalie Silvanovich, Project Zero

This is a three-part series on exploiting messenger applications using vulnerabilities in WebRTC. CVE-2020-6514 discussed in the blog post was fixed on July 14 with these CLs.This series highlights what can go wrong when applications don't apply WebRTC patches and when the communication and notification of security issues breaks down. 

Part 3: Which Messengers?

In Part 2, I described an exploit for WebRTC on Android. In this section, I explore which applications it works on.

The exploit

When writing the exploit, I originally altered the SCTP packets sent to the target device by altering the source of WebRTC and recompiling it. This wasn’t practical for attacking closed source applications, so I eventually switched to using Frida to hook the binary of the attacking device instead. Frida’s hooking functionality allows for code to be executed before and after a specific native function is called, which allowed my exploit to alter outgoing SCTP packets as well as inspect incoming ones. Functionally, it is equivalent to altering the source of the attacking client, but instead of the alterations being made in the source at compile time, they are made dynamically by Frida at run time. The source for the exploit is available here.

There are seven functions that the attacking device needs to hook, as follows.

usrsctp_conninput // receives incoming SCTP
DtlsTransport::SendPacket // sends outgoing SCTP
cricket::SctpTransport::SctpTransport // detects when SCTP transport is ready
calculate_crc32c // calculates checksum for SCTP packets
sctp_hmac // performs HMAC to guess secret key
sctp_hmac_m // signs SCTP packet
SrtpTransport::ProtectRtp // suppresses RTP to reduce heap noise

These functions can be hooked as symbols, or as offsets in the binary.

There are also three address offsets from the binary of the target device that are needed for the exploit to work.  The offset between the system function and the malloc function, as well as the offset between the gadget described in the previous post and the malloc function are two of these. These offsets are in libc, which is an Android system library, so they need to be determined based on the target device’s version of Android. The offset from the location of the cricket::SctpTransport vtable to the location of malloc in the global offset table is also needed. This must be determined from the binary that contains WebRTC in the application being attacked.

Note that the exploit scripts provided have a serious limitation: every time memory is read, it only works if bit 31 of the pointer is set. The reasons for this are explained in Part 2. The exploit script has an example of how to fix this and read any pointer using FWD_TSN chunks, but this is not implemented for every read. For testing purposes, I reset the device until the WebRTC library was mapped in a favorable location.

Android Applications

A list of popular Android applications that integrate WebRTC was determined by searching APK files on Google Play for a specific string in usrsctp. Roughly 200 applications with more than five million users appeared to use WebRTC. I evaluated these applications to determine whether they could plausibly be affected by the vulnerabilities in the exploit, and what the impact would be.

It turned out the ways applications use WebRTC are quite varied, but can be separated into four main categories.

  • Projection: the screen and controls of a mobile application is projected into a desktop browser with user consent for enhanced usability
  • Streaming: audio and video content is sent from one user to many users. Usually there is an intermediary server, so the sender does not need to manage possibly thousands of peers, and the content is recorded for later viewing
  • Browsers: all major browsers contain WebRTC to implement the JavaScript WebRTC API
  • Conferencing: two or more users communicate via audio or video in real time

The impact of the vulnerabilities used in the exploit is different for each of these categories. Projection is low risk, as a lot of user interaction is required to set up the WebRTC connection, and the user has access to both sides of the connection in the first place, so there is little to gain by compromising the other side. 

Streaming is also fairly low risk. While it’s possible that some applications use peer-to-peer connections when a stream has a low number of viewers, they usually use an intermediary server that terminates the WebRTC connection from the sending peer, and starts new connections with the receiving peers. This means that the attacker usually cannot send malformed packets directly to a peer. Even with a set-up where streaming is performed peer-to-peer, user interaction is required for the target to view the stream, and there’s often no way to limit who can access a stream. For this reason, streaming applications that use WebRTC are probably not useful for targeted attacks. Of course, it is possible that these vulnerabilities affect the servers used by streaming services, but this was not investigated in this research.

Browsers are almost certainly vulnerable to most bugs in WebRTC, because they allow a large amount of control over how it is configured. To exploit such a bug in a browser, an attacker would need to set up a host that acts like the other peer in the peer-to-peer connection, and convince the target to visit a webpage that starts a call to that host. In this case, the vulnerability would have a similar impact to other memory corruption vulnerabilities in JavaScript.

Conferencing is the highest risk usage of WebRTC, but the actual impact of a vulnerability depends on a lot of how users of an application contact each other. The highest risk design is an application where any user can contact any other user based on an identifier. Some applications require the callee to have interacted in a specific way with the caller before a call can be made, which makes users harder to contact a target and generally reduces risk. Some applications require users to enter a code or visit a link to start a call, which has a similar effect. There is also a large group of applications where it is difficult or impossible to call a specific user, for example chat roulette applications, and applications which have features that allow a user to start a call to customer support. 

For this research, I focused on conferencing applications that allow users to contact specific other users. This reduced my list of 200 applications to 14 applications, as follows.

Installs on Play Store
OK and TamTam (similar apps by same vendor)

This list was compiled on June 18, 2020. Note that a few applications were removed because their server was not operational on that day, or they were very difficult to test (for example, required watching multiple ads to make a single call).

One application tested will not be identified in this blog post, as a serious additional vulnerability was discovered in the process of testing that has not yet been fixed or reached its disclosure deadline. This blog post will be updated when the disclosure deadline has passed. Update (2020-10-14): The affected application was Mocha. We discovered this vulnerability.

Testing the Exploit

The following section describes my attempts to test the exploit against the above applications. Please note that due to the number of applications, limited time was spent on each, so there is no guarantee that every attack against WebRTC was considered. While I am very confident that applications found to be exploitable are indeed exploitable, I am less confident about applications found to be not exploitable. If you need to know whether a specific application is vulnerable for the purposes of protecting users, please contact the vendor instead of relying on this post.


I started off by testing Signal because it is the only open-source application on this list. Signal integrates WebRTC as a part of a library called ringrtc. I built ringrtc and then Signal with symbols, and then hooked the needed symbols with the Frida script on the attacker device. I tried the exploit and it worked about 90% of the time!

This attack did not require any user interaction with the target device because Signal starts the WebRTC connection before an incoming call is answered, and this connection can accept incoming RTP and SCTP. The exploit is not 100% reliable on Signal and other targets because Bug 376 requires that a freed heap allocation is replaced with the next allocation of the same size performed by the thread, and occasionally another thread will do an allocation of the same size in the meantime. Failure results in a crash that is usually not evident to the user because the process respawns, but a missed call will appear.

This exploit was performed on Signal 4.53.6 which was released on January 13, 2020, as Bug 376 had already been patched in Signal by the time I finished the exploit. CVE-2020-6514 was also fixed in later versions, and ASCONF has also been disabled in usrsctp, so the code that caused Bug 376 is no longer reachable. Signal has also recently implemented a feature that requires user interaction for the WebRTC connection to be started when the caller is not in the callee’s contacts. Signal has also stopped using SCTP in their beta version, and plans to add this change to the release client once it is tested. The source for this exploit is available here.

Google Duo

Duo was also an interesting target, as it is preinstalled on so many Android devices. It dynamically links the Android WebRTC library, with no obvious modifications. I reverse engineered this library in IDA to find the location of all the functions that needed to be hooked, and then modified the Frida script to hook them based on their offsets from an exported symbol. I also modified the offset between the cricket::SctpTransport vtable and the global offset table, as it was different than in Signal. The exploit also worked on Duo. Source for the Duo exploit is available here.

This vulnerability did not require any user interaction, as like Signal, Duo starts the WebRTC connection before a call is answered.

The exploit was tested on version 68.0.284888502.DR68_RC09 which was released on December 15, 2019. The vulnerability has since been fixed. Also, at the time this application was released, it was possible for Duo to call any Android device with Google Play Services installed, regardless of whether Duo had been installed. This is no longer the case. A user now needs to set up Duo and have the caller in their contacts for an incoming call to be received.

Google Hangouts

While Google Hangouts uses WebRTC, it does not use data channels, and does not exchange SDP in order to set up calls, so there is no obvious way to enable them from a peer. For that reason, the exploit does not work on Hangouts.

Facebook Messenger

Facebook Messenger is another interesting target. It has a large number of users, and according to its documentation, any user can call any other user based on their mobile number. Facebook Messenger integrates WebRTC into a library called, which dynamically links to usrsctp from another library, Facebook Messenger downloads these libraries dynamically as opposed to including them in the APK, so it is difficult to identify the version I examined, but it was downloaded on June 22, 2020. 

The library appears to use a version of WebRTC that is roughly six years old, so it was before the class cricket::SctpTransport existed. That said, the analogous class cricket::DataMediaChannel appeared to be vulnerable to CVE-2020-6514. The library appears to be more modern, but contains the vulnerable code for Bug 376. That said, it does not appear to be possible to reach this code from Facebook Messenger, as it is set to use RTP data channels as opposed to SCTP data channels, and does not accept attempts to change the channel type via Session Description Protocol (SDP). While it is not clear whether the motivation behind this design is security, this is a good example of how restricting attacker access to features can reduce an application’s vulnerability. Facebook also waits until a call is answered before starting the WebRTC connection, which further reduces the exploitability of any WebRTC vulnerabilities that affect it.

Interestingly, Facebook Messenger also contains a more modern version of WebRTC in a library called, but it does not appear to be used by the application. It is possible to get Facebook Messenger to use the alternate library by setting a system property on Android, but I could not find a way an attacker could cause a device to switch libraries.


Like Facebook Messenger, while Viber version appeared to contain the vulnerable code, but the application disables SCTP when the PeerConnectionFactory is created. This means an attacker cannot reach the vulnerable code.


VK is a social networking app released by in which users have to explicitly allow specific other users to contact them before each user is allowed to call them. I tested my exploit against VK, and it required some modifications to work. To start, VK doesn’t use data channels as a part of its WebRTC connection, so I had to enable it. To do this, I wrote a Frida script that hooks nativeCreateOffer in Java, and makes a call to createDataChannel before the offer is created. This was sufficient to enable SCTP on both devices, as the target device determines whether to enable SCTP based on the SDP provided by the attacker. The version of WebRTC was also older than the one I wrote the exploit for. WebRTC doesn’t contain any version information, so it is difficult to tell for sure, but the library appeared to be at least one year old based on log entries. This meant that some of the offsets in the ‘fake object’ used by the exploit were different. With a few changes, I was able to exploit VK.

VK sends an SDP offer to a target device to start a call, but the target does not return the SDP answer until the user has accepted the call, which means this exploit requires the target to answer the call before the WebRTC connection is started. This means the exploit will not work unless the target manually answers the call. In the video below, the exploit takes a fair amount of time to run after the user has answered. This is due to how I designed the exploit, and not due to fundamental limitations of the vulnerabilities it uses. In particular, the exploit waits for usrsctp to generate specific packets even though they could be generated more quickly by the exploit script, and also uses delays to avoid packet reordering when responses could be checked instead. It is likely that with enough effort, this exploit could run in less than five seconds. Also note that I altered the exploit to work with a single incoming call, as opposed to two incoming calls in the exploits above, as it is not realistic to expect a target to answer a call twice in quick succession. This didn’t require substantial changes to how the exploit works, though it does make the exploit code more complex and difficult to debug.

Regardless, the requirement that a user must choose to accept calls from an attacker before they can call, alongside the requirement that the user answer the call and stay on the line for a few seconds makes this exploit substantially less useful against VK compared to applications without these features.

Testing was performed against VK 6.7 (5631). Like Facebook, VK dynamically downloads its version of WebRTC, so it is difficult to specify its version, however testing was performed on July 13, 2020. VK has since updated their servers so that a user cannot start a call with SDP that contains data channels, so the exploit no longer works. Note that VK does not use WebRTC for two-party calls, only group calls, so I tested this exploit using a group call. The source for the exploit is available here.

OK and TamTam

OK and Tamtam are similar messaging applications released by the same vendor, also They use a dynamically downloaded version of WebRTC that is identical to the one used by VK. Since the library is exactly the same, my exploit also worked on OK, and I didn’t bother also testing TamTam because it is so similar.

Like VK, OK and TamTam do not return the SDP answer until the target has answered the call by interacting with the phone, so this is not a fully remote exploit on OK and TamTam. OK also requires users to choose to accept messages from another user before the user can call them. TamTam is a bit more liberal, for example, if a user verifies a phone number, any user who has their phone number can contact them.

Testing was performed on version 20.7.7 of OK on Monday, July 13. SDP-only testing was performed on TamTam version 2.14.0. Since then, the servers for these applications have been updated so that SDP containing data channels cannot be used to start a call, so the exploit no longer works.

Discord has documented its use of WebRTC thoroughly. The application uses an intermediary server for WebRTC connections, which means that it is not possible for a peer to send raw SCTP to another peer, which is required for the exploit to work. Discord also requires several clicks to enter a call. For these reasons, Discord is not affected by the vulnerabilities discussed in this post.


JioChat  is a messaging application that allows for any user to call any other user based on phone number. Analyzing version, it appeared that its WebRTC integration contained both vulnerabilities, and the app exchanges the SDP offer and answer before the callee accepts the incoming call, so I expected the exploit to work without user interaction. However, this was not the case when I tested it, and it turns out that JioChat uses a different strategy to prevent the WebRTC connection from starting until the callee has accepted the call. I was able to easily bypass this strategy, and get the exploit to work on JioChat.

Unfortunately, JioChat’s connection delay strategy introduced another vulnerability, which has been fixed, but the disclosure period has not expired for. For this reason, details of how to bypass it will not be shared in this blog post. The source for the exploit without this functionality is available here. JioChat has recently updated their servers so that SDP containing data channels cannot be used to start a call, meaning that the exploit no longer works on JioChat.

Slack and ICQ

Slack and ICQ are similar in that they both integrate WebRTC, but do not use the transport features of the library (note that Slack doesn’t integrate WebRTC directly for audio calls, it integrates Amazon Chime, which integrates WebRTC). They both use WebRTC for audio processing only, but implement their own transport layer and do not use WebRTC’s RTP and SCTP implementations. For this reason, they are not vulnerable to the bugs discussed in this blog post, and many other WebRTC bugs.


BOTIM has an unusual design that prevents the exploit from working. Instead of calling createOffer and exchanging SDP, each peer generates its own SDP based on a small amount of information from the peer. SCTP is not used by this application by default, and it was not possible to use SDP to turn it on. Therefore, it was not possible to use this exploit. BOTIM does appear to have a mode where it exchanges SDP with a peer, but I could not figure out how to enable it.

Other Application

The exploit worked in a fully remote fashion on one other application, but setting up the exploit revealed an obvious additional serious vulnerability in the application. Details of the exploit’s behavior on the application will be released after the disclosure period has expired for the vulnerability.


The Risk of WebRTC

Out of the 14 applications analyzed, WebRTC enabled a fully remote exploit on four applications, and a one-click exploit on two more. This highlights the risk of including WebRTC in a mobile application. WebRTC does not pose a substantially different risk than other video conferencing solutions, but the decision to include video conferencing in an application introduces a large remote attack surface that wouldn’t be there otherwise. WebRTC is one of the few fully remote attack surfaces of a mobile application, and of Android in general. It is likely the highest risk component in almost every application that uses it for video conferencing.

Video conferencing is vital to the functionality of some applications, but in others it is an ‘extra’ that is rarely used. Low usage does not make video conferencing any less of a risk to users. It is important for software makers to consider whether video conferencing is a truly necessary part of their application, with a full understanding of the risk it presents to users.

WebRTC Patching

This research showed that many applications fall behind with regard to applying security updates to WebRTC. Bug 376 was fixed in September of 2019, yet only two of the 14 applications analyzed had patched it. There were several factors that led to this.

To start, usrsctp does not have a formal process for identifying and communicating vulnerabilities. Instead, bug 376 was fixed like any other bug, so the code was not pulled into WebRTC until March 10, 2020.  Even after it was patched, the bug was not noted on the Security Notes for the Chrome Stable channel, which is where WebRTC tells developers to look for security updates. This means that developers of applications that use an older version of WebRTC and cherry-pick fixes, or applications that include usrsctp separately from WebRTC would not be aware of the need to apply this patch.

This is not the full story though, as many applications include WebRTC as an unmodified library, and there have been other WebRTC vulnerabilities included in the Chrome Security Notes since March 2020. Another contributing factor is that until 2019, WebRTC did not provide any security patching guidance to integrators, in fact, their website inaccurately said that no vulnerabilities had ever been reported in the library, which occurred because WebRTC security bugs are generally filed in the Chromium bug tracker, and there was no process for considering these bug’s impact on non-browser integrators at the time. Many of the applications I analyzed had versions of WebRTC that predated this, so it is likely that the legacy of this incorrect guidance still causes applications to not update WebRTC. While WebRTC has done a lot to make it easier for integrators to patch WebRTC, for example allowing large integrators to apply for advance notice of vulnerabilities, there is still likely a long tail of integrators who have only seen the old guidance. Of course, there is no guarantee that integrators would have followed better guidance if it was available, but considering that for a long time it was very difficult for an integrator to know when and how to update WebRTC even if they wanted to, it is likely it would have had an impact.

Integrators also have a responsibility to keep WebRTC up to date with security fixes, and many of them have failed in this area. It was surprising to see so many versions of WebRTC that are well over a year old. Developers should monitor every library they integrate for security updates, and apply them promptly.

Application Design

Application design affects the risk posed by WebRTC, and many applications researched were designed well. The easiest, and most important way to limit the security impact of WebRTC is to avoid starting the WebRTC connection until the callee has accepted the call by interacting with the device. This turns an exploit that can compromise any user quickly into an exploit that requires user interaction, and won’t be successful on every target. It also makes lower quality vulnerabilities not practically exploitable, because while a fully remote exploit can be attempted many times without the user noticing, an exploit that requires a user to answer a call needs to work in a small number of tries.

Starting the WebRTC connection late has a performance impact, and precludes certain features, like giving the callee a preview of the call. Of the applications that the exploit worked on, two started the connection without user interaction, and two required user interaction. JioChat and the application we are not yet identifying tried to use unique tricks to delay the connection until the user accepted the call without performance impact, but introduced vulnerabilities as a result. Developers should be aware that the best way to delay a WebRTC connection is to avoid calling setRemoteDescription until the user has accepted the call.  Other methods might not actually delay the connection and can cause other security problems.

Another way to reduce the security risk of WebRTC is to limit who an attacker can call, for example by requiring that the callee have the user in their contact list, or only allowing calls between users that have agreed to be able to message each other in the application. Like delaying the connection, this greatly reduces the targets an attacker can reach without a lot of effort.

Finally, integrators should limit the features of WebRTC an attacker can use to the features the application needs. Many applications were not vulnerable to this specific exploit because they had effectively disabled SCTP. Others did not use SCTP, but did not disable it in a way that prevented attackers from using it, and I was able to enable it. The best way to disable a feature in WebRTC is to remove it at compile time, which is supported for certain codecs. It is also possible to disable certain features through the PeerConnection and PeerConnectionFactory, and this is also very effective. Features can also be disabled by filtering SDP, but it is important to make sure that the filter is robust and tested thoroughly.


I wrote an exploit for WebRTC for Android involving two vulnerabilities in usrsctp. This exploit was fully remote on Signal, Google Duo, JioChat and one other application, and required user interaction on VK, OK and TamTam. Seven other messengers were not affected because they effectively disabled SCTP. Several applications used versions of WebRTC that did not include patches for either of the vulnerabilities used in the exploit. One remains unpatched. Low patch uptake is partially a result of WebRTC historically providing poor patching guidance. Integrators can reduce the risk of WebRTC by requiring user interaction to start a WebRTC connection, limiting who users can call easily and disabling unused features. They should also consider whether video conferencing is an important and necessary feature of their application.

Vendor Response

The software vendors mentioned in this blog post were given a chance to review this post before it was posted publicly, and some provided responses, as follows.


The WebRTC bug that was used both to bypass ASLR and move the instruction pointer has been fixed. WebRTC no longer passes the SctpTransport pointer directly into usrsctp, using an opaque identifier that is mapped to a SctpTransport instead, with invalid values being ignored. We have identified and patched every affected Google product and reached out to 50 applications and integrators using WebRTC, including all applications analyzed in this post. For all applications and integrators who have not yet patched the vulnerability, we recommend updating to the WebRTC M85 branch, or patching the following two commits: 1, 2.

User security is of the highest priority for all Group products, which include VK, OK, TamTam and others. Acting on the information we received regarding the vulnerability, we immediately started the process of updating our mobile apps to the latest version of WebRTC. This update is currently underway. We have also implemented algorithms on our servers that no longer allow this vulnerability to be exploited in our products. This action allowed us to fix the issue for all of our users within 3 hours of receiving the information with an exploit demonstration.


We appreciate the effort that went into finding these bugs and improving the security of the WebRTC ecosystem. Signal had already shipped a defensive patch that protected users from this exploit prior to its discovery. In addition to routine updates of our calling libraries, we continue to take proactive steps to mitigate the impact of future WebRTC bugs.


We're pleased to see that this report concludes that Slack is not impacted by the referenced WebRTC vulnerabilities and exploits. Upon learning about this risk, we undertook additional diligence and confirmed that the entirety of our Calls service is not impacted by the vulnerabilities and findings described here.

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