Unlike most conventional  real-time systems (e.g., SIP-based soft phones) WebRTC communications  are directly controlled by a Web server over some signalling protocol which may be XMPP , websockets , socket.io , Ajax etc . This poses new  challenges such as

• Web browser might expose a JavaScript APIs which allows  web server to place a video call itself.This may cause web pages to secretly record and stream the webcam activity from user’s computer
• malicious calling services can record the user’s conversation and misuse
• malicious webpages can lure users via advertising and execute auto calling services .
• Since JavaScript calling APIs are implemented as browser built-ins , un authorized access to these can also make user’s audio and camera streams vulnerable
• If program and APIs allow the server to instruct the browser to send arbitrary content, then they can be used to bypass firewalls or mount denial of service attacks.

WEB ATTACKERS are who induce users to visit their sites but do not control the network.NETWORK ATTACKERS are who are able to control network. When analyzing HTTP connections, we must assume that traffic is going to the attacker.

The Browser Threat Model

The browser acts as a TRUSTED COMPUTING BASE (TCB) both from the user’s perspective and to some extent from the  server’s.  HTML and JavaScript (JS) provided by the web server can execute scripts on browser and generate actions and events . However browser  operates in a sandbox that isolates these scripts both from the user’s computer and from server .

Access to Local Resources

The users computer may have lot of private and confidential data on the disk . Browser do make it mandatory that user must explicitly select the file and consent to its upload before doing file upload and transfer transactions . However still it is not very rare that misleading text and buttons can make users click files .  

Another way of accessing local resources is through downloading malicious files to users computer which are executable and may harm users computer .

SOP or Same Origin Policy

SOP  forces scripts from each site to run in their own, isolated, sandboxes.  It enables webpages and scripts from the same origin server to interact with each other’s JS variables, but prevents pages from the different origins or even iframes on the same page to not exchange information.

As part of SOP scripts are allowed to make HTTP requests via the  XMLHttpRequest() API to only those server which have same ORIGIN/domain as that of the originator .

CORS [Cross-Origin Resource Sharing ]

CORS enables multiple web services to intercommunicate . Therefore when a script from origin A executes what would otherwise be a forbidden cross-origin request, the browser instead contacts the target server B to determine whether it is willing to allow cross-origin requests from A.  If it is so willing, the browser then allows the request.  This consent verification process is designed to safely allow cross-origin requests.

Websockets

Even websockets overcome SOP and establish cross origin transport channels .

Once a WebSockets connection has been established from a script to a site, the script can exchange any traffic it likes without being required to frame it as a series of HTTP request/response transactions.

WebSockets use masking technique to randomize the bits that are being transmitted , thus making it more difficult to generate traffic which resembles a given protocol , thus making it difficult for inspection from flowing traffic .

JSONP

Jsonp is a hack designed to bypass origin restriction through script tag injection. A JSONp enabled server passes the response in user specified function

when we use <script> tags the domain limitation is ignored ie we can load scripts from any domain .  So when we need to fetch get exchange data just pass callback parameters through scripts . For example

</p>
<!-- /wp:paragraph -->

<!-- wp:paragraph -->
<p>function mycallback(data){<br>
// this is the callback function executed when script returns<br>
alert("hi"+ data);&lt;/span&gt;<br>
}</p>
<!-- /wp:paragraph -->

<!-- wp:paragraph -->
<p>var script = document.createElement('script');<br>
script.src = '//serverb.com/v1/getdata?callback=mycallback'<br>
document.head.appendChild(script) </p>
<!-- /wp:paragraph -->

<!-- wp:paragraph -->
<p>

There have been found vulnerabilities in the existing Java and Flash consent verification techniques and handshake.

The Security arising from ICE and TURN

ICE

Sender and receiver are able to share media stream after a offer answer handshake. But we already need one in order to do NAT hole-punching. Presuming the ICE server is malicious , in absence of transaction IDs by stun unknow to call scripts , it is not possible for the webpage of receiver to ascertain is the data is forged or original . Thus to prevent this the browser must generate hidden transaction Id’s and should not sharing with call scripts ,even via a diagnostic interface.

IP Location Privacy

As soon as the callee sends their ICE candidates, the caller learns the callee’s IP addresses.  The callee’s server reflexive address reveals a lot of information about the callee’s location.

To prevent server should suppress the start of ICE negotiation until the callee has answered.

Also user may hide their location entirely by forcing all traffic through a TURN server.

Communications Security

Goal of webrtc based call services should be to create channel which is secure  against both message recovery and message modification for all audio / video and data .

Threats from Screen Sharing

With the increasing requirement of screen sharing in web app and communication systems there is always a high threat of oversharing / exposing confidential passwords , pins , security details etc . This may either through some part of screen or some notification whihc pops up .

There is always the case when the user may believe he is sharing a window when in fact they are the entire desktop.

The attacker may request screensharing and make user open his webmail , payment settings or even net-banking accounts .

Long term access to camera and microphone

When user frequently uses a site he / she may want to give the site a long-term access to the camera and microphone ( indicated by ” Always allow on this site ” in chrome ). However the site may be hacked and thus initiate call on users’ computer automatically to secretly listen-in .

False UI shows cut off call while still being active

Unless the user checks his laptops glowing camera light LED or goes and monitors the traffic himself he would not know if there is active call in background, which according to him he had cut off . In such a case an attacker may pretend to cut a call shows red phone signs and supportive text but still keep the session and media stream active placing himself on mute .

During-Call Attack

Even if the calling service cannot directly access keying material ,it  can simply mount a man-in-the-middle attack on the connection. The idea is to mount a bridge capturing all the traffic.

To protect against this it is now mandatory to use https for using getusermedia and otherwise also recommended to keep webrtc comm services on https or use strict fingerprinting .
This section is derived from Security Considerations for WebRTC draft-ietf-rtcweb-security-08

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How can I make my WebRTC solution secure?

In one of my previous posts I have mentioned about Security threats to WebRTC Solution . It includes mainly 4 ways in which WebRTC Solution Providers and Users are vulnerable . It includes

1. Identity Management ,
2. Browser Security ,
3. Authentication and
4. Media encryption.

Since I have already covered these topics here( https://altanaitelecom.wordpress.com/2014/10/03/security-for-webrtc-applications/ ) I will not repeat the same here. This post is about making WebRTC secure so that they can be used inn area which require sensitive data to be communicated and need to be secure enough to withstand and hacks and attacks.

In the recent months everyone has been trying to get into the WebRTC  space but at the same time fearing that hackers might be able to listen in on conferences, access user data, or even private networks. Although development and usage around WebRTC is so simple , the security and encryption aspects of it are in the dim light.

So does existing WebRTC model offer security ?

We know that the forces behind WebRTC standardization are WHATWG, W3C, IETF and strong internet working groups . WebRTC security was already taken into consideration when standards were being build for it . The encryption methods and technologies like DTLS and SRTP were included to safeguard users from intrusions so that the information stays protected.

WebRTC media stack has native built-in features that address security concerns. The peer-to-peer media is already encrypted for privacy . Figure below:

For WebRTC to transfer real time data, the data is first encrypted using the DTLS (Datagram Transport Layer Security) method. This is a protocol built into all the WebRTC supported browsers from the start (Chrome, Firefox and Opera). On a DTLS encrypted connection, eavesdropping and information tampering cannot take place.

Other than DTLS, WebRTC also encrypts video and audio data via the SRTP (Secure Real-Time Protocol) method ensuring that IP communications – your voice and video traffic – can not be heard or seen by unauthorized parties.

What is SRTP ?

The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications.

Earlier models of VOIP communication such as SIP based calls had an option to use only RTP for communication thereby subjecting the endpoint users to lot of problem like compromising media Confidentiality  . However the WebRTC model mandates the use of SRTP hence ruling out insecurities of RTP completely. For encryption and decryption of the data flow SRTP utilizes the Advanced Encryption Standard (AES) as the default cipher.

What is DTLS ?

DTLS allows datagram-based applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. The DTLS protocol is based on the stream-oriented Transport Layer Security (TLS) protocol .

Together DTLS and SRTP enables the exchange of the cryptographic parameters so that the key exchange takes place in the media plane and are multiplexed on the same ports as the media itself without the need to reveal crypto keys in the SDP.

Today the browser acts as a TRUSTED COMPUTING BASE (TCB) where the HTML and JS act inside of a sandbox that isolates them both from the user’s computer.

A script cannot access user’s webcam , microphone , location , file , desktop capture without user’s explicit consent. When the user allows access, a red dot will appear on that tab, providing a clear indication to the user, that the tab has media access.

Figure depicting browser asking for user’s consent to access Media devices for WebRTC .

Figure depicting Media Capture active on browser with red dot .

we know that XMLHttpRequest() API can be used to secretly send data from one origin to other and this can be used to secretly send information without user’s knowledge . However now , SAME ORIGIN POLICY (SOP) in browser’s prevents server A from mounting attacks on server B via the user’s browser, which protects both the user (e.g., from misuse of his credentials) and the server B (e.g., from DoS attack).

In-spite of all this ,  the security challenges with Web Server based WebRTC service are many for example :

1. If the both the peers have WebRTC browser then one can place a WebRTC call to callee anytime this might result in denial of service .
2. Since the media is p2p and also can override firewalls settings through TURN server , it can result in unwanted data being send to peer .
3. One may secretly make calls to users through website and extract information .
4. Threat from screen sharing, for example user might mistakenly share his internet banking screen or some confidential information.
5. Giving long-term access to the camera and microphone for certain sites is also a concern . for example : since next time you visit a site that has access to your microphone and camera , they can secretly be viewing youe webcam and microphone inputs .
6. Clever use of User Interface to mask a ongoing call can mislead the user into believing that call has been cut while it is secretly still ongoing.
7. Network attackers can modify an HTTP connection through my Wifi router or hotspot to inject an IFRAME (or a redirect) and then forge the response to initiate a call to himself.
8. As WebRTC doesn’t yet have an congestion control mechanism , it can eat up a large chunk of user’s bandwidth.
9. By visiting chrome://webrtc-internals/ in chrome browser alone , one can view the full traces of all webRTC communication happening through his browser . The traces contain all kinds of details like signalling server used , relay servers , TURN servers , peer IP , frame rates etc .

Ofcourse other challenges that arrive with any other webservice based architecture are also applicable here such as :

1. Malicious Websites which automatically execute the attacker’s scripts.
2. User can be induced to download harmful executable files and run them.
3. Improper use of W3C Cross-Origin Resource Sharing (CORS) to bypass SAME ORIGIN POLICY (SOP) .

Best practices to make your VOIP Solution more secure

A simple WebRTC architecture is shown in the figure below :

By following the simple steps described below one can ensure a more secure WebRTC implementation . The same applies to healthcare and banking firms looking forth to use WebRTC as a communication solution for their portals .

1. Ensure that the signalling platform is over a secure protocol such as SIP / HTTPS / WSS .

2. User’s that can participate in a call , should be pre registered / Authenticated with a registrar service. Unauthenticated entities should be kept away from session’s reach .

2. Make sure that ICE values are masked thereby not rendering the caller/ callee’s IP and location to each other through tracing in chrome://webrtc-internals/ or packet detection in Wirehsark on user’s end.

3. Also since media is p2p , the media contents like audio video channel are between peers directly in full duplex. Thus

4. As the signalling server maintains the number of peers , it should be consistently monitored for addition of suspicious peers in a call session. If the number of peers actually present on signalling server is more that the number of peers interacting on WebRTC page then it means that someone is eavesdropping secretly and should be terminated from session access by force.

5. It is observed these days that users simply agree to all permissions request from browser without actually consciously giving consent . Therefore user’s should be made aware of API in websites which ask for undue permissions . For example permission to :

6. To protect against Man-In-The-Middle (MITM) attack the media path should be monitored regularly for no suspicious relay.

7. Third party API should be thoroughly verified before sending their data on WebRTC DataChannel.

8. Before Desktop Sharing user’s should be properly notified and advised to close any screen containing sensitive information .

What happens if your VOIP solution is on the verge of being compromised ?

As the media connections are p2p , even if we restart the signalling server , it will not affect the ongoing media sessions . Only the time duration ( probably 3 – 4 minutes ) it takes to restart the server , is when the users will not be able to connect to signalling server for creating new sessions .

Most browsers today like Google Chrome and Mozilla Firefox have a goof record of auto-updating themselves withing 24 hours of a vulnerability of threat occurring .

If a call is confirmed to be compromised , it should be within the power of Web Application server rendering the WebRTC capable page to cut off the call .

References :

• https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage
• https://datatracker.ietf.org/doc/draft-ietf-rtcweb-sdp/
• https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/