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SIP Security

Roadmap

Basics

Terms, concepts, and so on

o Threats, security attributes, vulnerabilities
Network defenses
o Cryptogrpahic primitives
o Protocols: SSL/TLS, IPsec, PGP, S/MIME, etc.

VoIP Security

VoIP Risks, Threats, and Attacks

Best Practices for VoIP Security
Open Issues

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Proliferation of VoIP

People are moving from PSTN to VoIP service

VoIP is cheaper, more convenient and flexible, with diverse features
The number of residential VoIP subscribers worldwide is expected to grow
significantly
4.8 million in 2004
~288 million by 2013
Types of VoIP
Managed: Vonage, AT&T, Verizon, etc.
Unmanaged: PC-to-PC VoIP or Softphone (e.g., Skype)

Expectations on VoIP

Confidentiality Security the No.1 concern

Emergency calls: 15, 17,18 in France
Lawful surveillance
Quality of Service (OoS)

Latency

Jitter

Packet loss

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Key Functional Components of VoIP

VoIP signaling

VoIP voice stream

Responsible for establishing, managing, tearing down VoIP sessions

H.323 1996 ITU
MGCP (Media Gateway Control Protocol) 1999 IETF RFC
SIP (Session Initiation Protocol) 1999, 2002 IEFT RFC
o The dominant VoIP signaling protocol
RTP (Real-time Transport Protocol)
SRTP (Secure RTP)

VoIP billing

Indispensable for VoIP service providers (e.g., Vonage, AT&T)

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VoIP Protocol Stack

H.323 Stack Diagram

SIP Stack Diagram

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Infrastructure Attacks

7 Application

Eavesdropping/Sniffing

For subnets using broadcast technologies such as WiFi, each attached

systems NIC can capture any communication on the subnet by using tools like
Wireshark, tcpdump/windump

Routers and internal switches can look at/export traffic they forward

Tap a link is possible: insert a device to mirror physical signal

Disruption

Transport

With physical access to a subnetwork, attacker can overwhelm its signaling

(e.g., jam WiFis RF)

Network

Data Link

Or send messages that violate the Layer-2 protocols rules, e.g., send messages
> maximum allowed size, sever timing synchronization, ignore fairness rules

Routers & switches can simply drop traffic

Physical

Spoofing

With physical access to a subnetwork, attacker can create their own msg.

Root/administrator privileges may gain full control

Spoofing + eavesdropping is more powerful: attacker can understand exact

state of victims communication and craft their spoofed traffic to match it
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SIP Overview

SIP is an IETF standard

RFC 2543 of 1999 (obsolete)

RFC 3261 of 2002 an application-layer control (signaling) protocol for creating,

modifying, and terminating sessions with one or more participants

Text-based, very similar to HTTP

SIP sessions include

Internet telephone call

Multimedia conferences

Instant messaging

and so on

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SIP Functionalities

Learn, determine the location, availability of remote communicating party

Registration

Call routing

Call redirection

Establish a session between end-points

Call setup, transfer, termination

Negotiate the media capability

Use SDP (Session Description Protocol) to specify the media parameters (e.g.,
IP address, port number, codec)

SIP Components

User Agents owned/used by subscribers

User Agent Client (UAC) who initiates a call

User Agent Server (UAS) who receives a call

SIP Servers maintained by service providers

Proxy server: relays the signaling messages (and potential voice streams) between
the caller and callee

Location server: keeps where a subscribers UA is currently at (the IP address)

Registrar server: accept registration from subscribers about their current locations,
and keeps track subscribers whereabouts at location server

Redirect server: provides information about next hop to a subscriber

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SIP Messages

Request identified by a method name

REGISTER tell where the UAC is currently at

INVITE Initiate a call to someone

BYE terminate an established call

ACK acknowledge the receipt of some message

CANCEL quit from an ongoing call setup

OPTION to query the capability of a server

Response identified by a number similar to HTTP

1 Provisional 100 Trying, 180 Ringing

2 Successful 200 OK

3 Redirection 301 Moved permanently, 302 Moved Temporarily

4 Failure 403 Not Found, 410 Gone, 403 Forbidden

5 Server Failure 503 Service Unavailable

6 Global Failure 600 Busy Everywhere

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SIP Network Architecture

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SIP Registration

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SIP Message Flow of Normal Call Setup & Down

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Questions ?

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Security Requirements on VoIP

Authenticity

When A dials Bs number, the call with reach B

The incoming call is really from who it claims to be caller ID is authentic

Confidentiality, privacy and anonymity

No one other than the caller(s) and callee(s) should do the following
o

Have access to the conversation content

Even know that Alice and Bob have talked over VoIP

Integrity

The call signaling and content have not been tempered with

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Security Requirements on VoIP, cont

Temper resistant billing

Service provider no service stealing, toll fraud

Subscriber no overcharge

Availability

Be resilient to denial-of-service (DoS) attack

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VoIP Security: System-Centric Perspective

VoIP signaling security

VoIP voice stream security

Protect the authenticity, integrity of the signaling message

Protect the authenticity, integrity and confidentiality of the voice
content (including any keys pressed)

VoIP billing security

Prevent service theft, toll fraud, undercharge, overcharge

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VoIP Security Threats Taxonomy (Incomplete)

Threats

Confidentiality & privacy

Integrity

Switch Default Password Vulnerability

Classical Wiretap Vulnerability

DHCP Server Insertion

TFTP Server Insertion Attack

ARP Cache Poisoning & Floods

Web Server Interfaces

Availability and DoS

CPU Resource Consumption
Default Password
Exploitable Software Flaws
Account Lockout Vulnerability

IP Phone Netmask Vulnerability

Extension to IP Address Mapping Vulnerability

Threats to VoIP Security

Identity and service theft

Service disruption against the following

Steal minutes from VoIP service provider

Call at other subscribers expense
The VoIP infrastructure
Individual subscriber
Terminate established calls
Prevent certain calls from being established

Call tampering

Tampering a phone call in progress: spoil the quality of the call by

injecting noise packets in the communication stream
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Threats to VoIP Security, cont

Call hijacking (Man-in-the-Middle attacks)

Registration spoofing

Unauthorized call redirection

Taking over established VoIP call via re-INVITE

Interception and modification

Conversation alteration mix, change the RTP voice stream

Eavesdropping

Wiretap and traffic analysis

Obtain names, password and phone numbers (service and information theft)

VoIP fraud

Voice phishing (aka vishing): faking as trustworthy organization (e.g., bank)

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Threats to VoIP Security, cont

Annoyance

SPIT (Spam over Internet Telephony)

Nuisance call

Attack against others

Divert VoIP traffic to flood someone

Denial of Service (DoS): consuming bandwidth or overloading resource

VoIP based botnet

Worms, viruses, and malware

What if attacker compromises the softphone running in the laptop or
handheld ?

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Attack#1: VoIP Service Stealing Case

According to court records, Pena
sold more than 10 million
minutes of VoIP service that had
been stolen from 15
telecommunications providers.
Prosecutors valued the lost
minutes at $1.4 million.

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Attack#2: SIP Phone Registration Spoofing

Before a SIP UA (i.e., phone) can be used to make or receiver a call, it

must register itself so that SIP proxy server know its current IP
address
What if some attacker send spoofed REGISTER message to the
registrar to trick the registrar into believing that the victim SIP phone is
at an IP address chosen by the attacker

What about SIP digest authentication ?

All the calls to the victim will reach attackers phone !

The SIP registrar does require authentication for registration

Assuming the attacker does not know the secret password shared
between the victim SIP phone and the registrar, can attacker still spoof
registration ? YES !
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Attack#2: SIP Phone Registration Spoofing, cont

The SIP digest does NOT cover the IP address of the SIP phone
The registrar actually uses the source IP address of the packet
containing the REGISTRER message
The attacker could simply replay the legitimate REGISTER message
from different IP address

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Attack#3: SIP Phone Registration Hijacking

With a MITM, attacker could completely hijack calls to the victim

1
5
2
3
4

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Attack#4: Fake BYE to Terminate Established SIP Call

Attacker could terminate any established SIP call by sending fake BYE
message to the SIP phone(s) and/or SIP proxy
SIP proxy requires digest authentication, so it could detect fake
BYE and ignore it
Existing SIP phones however will honor any BYE message with
correct Call-ID

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Attack#5: VoIP CallerID Spoofing

Caller ID of SIP call is NOT trustworthy, and it is easy to spoof

Just modify the From field of INVITE message

There are companies that offer caller ID spoofing service to the public:
http://www.calleridspoofing.info/

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Billing of Managed VoIP

Billing is fundamental to VoIP service providers (e.g., Vonage, AT&T)

Certain VoIP calls are charged on a per minute basis: International call,
Service providers rely on accounting and billing for charging their
customers for the serve they provided

Billing has direct impact on each individual VoIP subscriber

Charges for the VoIP services they have chosen and used
NO Charges on the calls they have NOT made
NO Overcharge on the calls they have made

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Requirements of VoIP Billing

Billing needs to be reliable

Resilient to billing fraud

Provides consistent view on what the service providers has provided and
what the subscriber has received

Billing needs to be trustworthy

It will determine
o (a) how much money the service provider will make;
o (b) how much money the subscriber will pay
Inaccurate or corrupted bill will create disputes between the service
provider and its customers

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Billing and Signaling of VoIP

Existing VoIP billing is based on VoIP signaling, which determines

Any vulnerability in VoIP signaling could be a potential vulnerability

of VoIP billing

The caller and callee of the call ---- communication parties

When the call starts and ends ---- call duration
Where the call will be routed ---- call path

Manipulation of SIP messages

Fake SIP messages

VoIP signaling protocols

SIP the dominant VoIP signaling protocol

MGCP
H323
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VoIP Billing Vulnerabilities

The use public Internet for signaling and billing opens many doors for
attacks

MITM at any router or gateway along with VoIP signaling path

How vulnerable are those deployed commercial SIP-based VoIP

services ?

Vonage, AT&T, Verizon, Broadvoice

Reference R. Zhang, et al., Billing Attacks on SIP-Based VoIP Systems, in Proceedings of the first USENIX workshop on Offensive Technologies
(WOOT '07).

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VoIP Billing Attack #1: INVITE Replay billing attack

Callers SIP phone initiates a call by sending an INVITE message to its SIP
server, which tracks the INVITE message for billing and accounting
Attack: an attacker can replay some captured INVITE message and make calls
at others expense

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VoIP Billing Attack #2: FakeBusy Billing

BUSY message is not protected by SIP authentication

The MITM between a SIP phone and SIP server could do the following

Hijack the VoIP calls of targeted SIP phone

Establish the calls with bogus content

When both the caller side and callee side have MITM, the MITMs could control the
hijacked call duration while keeping the caller and callee unaware of the call

This could result in overcharge on calls the VoIP subscriber has actually made

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VoIP Billing Attack #3: ByeDelay and ByeDrop

Established SIP calls are terminated by BYE message

Can an MITM delay or simply drop the BYE message of established SIP call ?

The SIP server will think the call is still alive and just count on the time, thereby
prolonging the duration of established calls

This could lead to overcharge on calls the VoIP subscriber has made

Such a delay or drop of the BYE message does not involve any modification of
the BYE message: SIP authentication would not help here

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Root Causes Analysis of Billing Attacks

SIP security is largely relied on existing security mechanisms for

HTTP and SMTP

TLS or IPsec protect the SIP signaling path

S/MIME protects the integrity and confidentiality of SIP messages
No end-to-end protection

Weakness of SIP authentication

Only applies to a few SIP messages, e.g., INVITE, BYE, REGISTER,

while others like TRYING, RINGING, 200 OK, ACK and BUSY are
unprotected
Only a few SIP fields are protected, e.g., request-URI, username, realm,
while other important SIP fields (e.g., SDP, From, To) are unprotected
One-way protection: only applies to UAC so SIP servers
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Potential Mitigations Against VoIP Billing Attacks

INVITE replay

FakeBusy

Just need to implement the SIP authentication correctly

Implementation errors may lead to this attack
Simply correlating the SIP and RTP message wont help (Bogus RTP
streams with correct IP addresses, port numbers, and sequence numbers)
Full integrity protection of SIP, RTP could defeat FakeBusy

ByeDelay, ByeDrop

Hard to defend even with full SIP, RTP integrity protection

o

No modification of any SIP or RTP

Delay and drop could happen naturally (accidental anomalies)

Some heuristics might help

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Questions ?

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Roadmap

Basics

Terms, concepts, and so on

o Threats, security attributes, vulnerabilities
Network defenses
o Cryptogrpahic primitives
o Protocols: SSL/TLS, IPsec, PGP, S/MIME, etc.

VoIP Security

VoIP Risks, Threats, and Attacks

Best Practices for VoIP Security
Open Issues

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SIP Security Mechanisms

SIP does NOT define its own security mechanism, it reuses existing
security mechanisms for HTTP, SMTP whenever possible
Two building blocks of SIP security mechanisms

SIP authentication and encryption can NOT be applied to the

whole SIP messages from end-to-end

Authentication: HTTP digest authentication

Encryption: IPsec, TLS, S/MIME

Intermediate SIP proxies need to modify and insert SIP message fields:
o Add via field
o Change request URI due to call-redirection

SIP Security is hop-by-hop

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HTTP Digest Authentication

Provides one-way authentication: identify UAC to a UAS or SIP

proxy (but does not identify UAS or SIP proxy !)

Assumes the two parties involved share a secret password

Anti-replay protection
Usually hard-coded in the UAC (SIP phone adaptor)

Uses challenge/response

A valid response contains a Checksum (by default MD5 RFC 2617, extended in
RFC 3310)

Response=F(nonce,

username, password, realm, SIP-method, request-URI)

Must be supported by all SIP compliant UA and SIP servers

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Message Flow of SIP Authentication

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Unauthenticated INVITE Message

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407 Authenticated Required Message

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Authenticated INVITE Message

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SIP Security Mechanism: IPsec

Can provide authentication, integrity and confidentiality for the

transmitted data

Suited to securing SIP hosts in a SIP VPN scenario (SIP user agents/proxies) or
between administrative SIP domains

Works for all UDP, TCP and SCTP based SIP signaling

Supports end-to-end as well as hop-by-hop scenarios

No SIP component is required to support IPsec

Key management issues

No default cipher suite for IPsec defined in SIP (RFC 3261 does not describe that)

One accepted protocol: Internet Key Exchange (IKE), which provides automated
cryptographic key exchange and management mechanisms and is used to negotiate
security associations (SA) (particularly used in the establishment of VPNs)

Setting up security association with every UA is too expensive !

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SIP Security Mechanism: TLS

RFC 3261 mandates the use of TLS for all SIP compliant servers

E.g., proxies, redirect servers, registrars

UAs are strongly recommended to support TLS

Protects SIP signaling messages against loss of integrity,

confidentiality, and anti-replay
Provides integrated key-management with authentication (server
only) and secure key distribution
Applied hop-by-hop between UAS/proxies or between proxies
Requires reliable transport stack (TCP-based signaling)

Not applicable to UDP-based SIP signaling

Universal Resource Indicator (URI) -- sips:
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SIP Security Mechanism: S/MIME

Can provide some degree of end-to-end authentication, integrity and

confidentiality for most SIP header fields

Excluding Request-URI, Record-Route, Route, Max-Forwards, and Proxy-Authorization

Specifies triple-DES as the required minimum encryption algorithm

Require PKI (Public-Key Infrastructure) to be effective

PKI includes services and protocols for managing public keys, often through the use of
Certification Authority (CA) and Registration Authority (RA) components (e.g., key
registration, certificate revocation, key selection, trust evaluation)

Incurring additional overhead

RFC 3261 recommends S/MIME to be used for UAS

If S/MIME is used to tunnel messages, TCP connection is recommended (larger

messages)

Authentication, integrity, and confidentiality protection of signaling data can be realized

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Security Enhancements for SIP (IETF Drafts)

SIP authenticated identity body (AIB)

SIP authenticated identity management

Upgrades that of RFC 3261(3DES); higher throughput, lower computational complexity,

and lower memory requirement, thus more suitable for mobile or embedded devices

Security mechanism agreement for SIP

For authenticating end users and to distribute authenticated identities with an AIB

S/MIME AES Requirement for SIP

Defines a generic SIP authentication token by adding an S/MIME body to a SIP message

TLS, HTTP Digest, IPsec with IKE, manually keyed IPSec without IKE, S/MIME

End-to-Middle, Middle-to-Middle, Middle-to-end Security

Enables intermediaries to use some of the SIP message header and body when end-to-end
security is applied: logging services for enterprises, firewall traversal, transcoding (tailoring
web pages for varying devices like PDAs, cell phones)

Secure information inserted by intermediaries (proxies sometimes need to modify a message)

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Secure RTP (from IETF Proceedings)

Security Parameter Index

(recommended for frequent rekeying in large groups), 32-bit

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Preventative Solutions to VoIP Security: Firewall

What is firewall ?

To block unauthorized access while permitting authorized communications, e.g.,

block all FTP traffic (port 21), allow all HTTP traffic (port 80)

Packet filter (header=<IP address, port No.,, protocol type>, 5-tuple), application
layer firewall, stateful firewall (maintains connection records and investigate
application data: NOT for a static port)

Needs: a logical segmentation of the network (Voice/Data)

A crucial functionality for a network containing PC-based phones

RTP makes use of dynamic UDP ports (1024-65534): prevent UDP DoS attack

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Collision between Voice and Data Traffic

PC-Based IP phones (data) require access to the (voice) segment to

place calls, leave messages, etc.,
IP Phones and call managers (voice) accessing voice mail (data)
Users (data) accessing the proxy server (voice)
The proxy server (voice) accessing network resources (data)
Traffic from IP Phones (voice) to the call processing manager (voice)
or proxy server (voice) must pass through the firewall because such
contacts use the data segment as an intermediary

Reference J. Halpern, IP Telephony Security in Depth, White Paper, Cysco Systems, 2002

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SIP-Aware Firewall

Thanks to the texting encoding of SIP, standard parsing tools (Perl,

lex and yacc) can be used
Must parse the SIP exchanges in order to discover which packets
contain the media, i.e., understand the SIP exchanges and extract the
relevant information
Must be stateful and monitor SIP traffic to determine which RTP
ports are to be opened and made available to which addresses (similar
case to H.323 but simpler call setup and header parsing)

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Case Study: Mitigations Against Flooding Type of DoS

Measure the difference between the numbers of attempted

connections and completed handshakes

Hellinger distance based anomaly detection

By B. Reynolds and D. Ghosal, Proc. of NDSS 2003

by Sengar et al (Proc. of IWQoS 2006)

SIP-aware firewall (SAFW), proposed by Columbia U. & Verizon

RTP pinhole/port filtering

Return routability check based on null-authentication

o

State machine sequencing

o

Could filter out those SIP messages with spoofed source

Filter out-of-state SIP messages

Maintain dialogue state (source, contact)

o

Only accept BYE with legitimate address

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Case Study: SIP-aware Application Layer Firewall

Sipd: SIP redirect, forking proxy
and registration server that provides
name mapping, user location and
scripting services
ASM -- Application Server Module
DPPM Deep Packet Processing
Module
FCP -- Firewall Control Protocol
CAM Content Addressable
Memory

Reference E. Yardeni, H. Schulzrinne, and G. Ormazabal, Large Scale SIP-aware Application Layer Firewall, Technical Report.
Columbia Univ.

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Case Study: SIP-aware Application Layer Firewall, cont

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Case Study: SIP-aware Application Layer Firewall, cont

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NAT: Network Address Translation

What is NAT ?

Can hide internal network address and enable several endpoints within a LAN
to share the same global IP address
Reduce the protected points of access and simplifies network management
Full cone NAT, restricted cone NAT, port restricted cone, symmetric NAT
Widely used, the need will not be alleviated by IPv6

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Preventative Solutions to VoIP Security: VPN

VPN: Virtual Private Networks

Tunnels between two endpoints that allow for data to be securely transmitted
between the nodes

IPsec ESP tunnel helps to traverse a public domain (the Internet) in a private manner,
although local VPN tunnel (within an intranet) is more secure and faster

VoIPsec: VoIP using IPsec

Reduces the threat of MIMA, packet sniffers, various voice traffic analysis, but

Brings negative effect on QoS (encryption at the end points)

Causes encryption/decryption latency (Secure Real Time Protocol, MIKEY via AES)

Bottleneck: scheduling and the lack of QoS in the crypto-engine

o

Better scheduling schemes: e.g., give OoS packets higher priority and prior to the
encryption phase

Expanded packet size (compression of packet size)

IPsec and NAT incompatibility (UDP encapsulation, RSIP, IPv6 tunnel broker)
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IDS for VoIP Network

What is IDS Intrusion Detection Systems ?

The system designed for identifying and responding to intrusive events or incidents

Two types: misuse-based IDS (signature-based) and anomaly-based IDS

Example Systems
SCIDIVE (Reported at DSN 2004 , by Wu et al.)

Sending fake BYE message to only one end-point will leave an orphan RTP stream from
the other end-point

Stateful, cross protocol correlation could detect this

What if attacker sends bogus BYE to both end-points ?

No orphan RTP stream (Happen naturally if both sides hang up at about the same time)

Interactive protocol state machine based IDS (Reported at DSN 2006, By Sengar et al.)
o

Could detect those attacks that do not follow the SIP state machine

What if some attack do follow the SIP state machine ?

CallerID spoofing, Registration hijacking

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Questions ?

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Roadmap

Basics

Terms, concepts, and so on

o Threats, security attributes, vulnerabilities
Network defenses
o Cryptogrpahic primitives
o Protocols: SSL/TLS, IPsec, PGP, S/MIME, etc.

VoIP Security

VoIP Risks, Threats, and Attacks

Best Practices for VoIP Security
Open Issues

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Challenges in Securing VoIP

Open architecture
Real-time constraints
Multiple protocols
Key management issues
Emergency services

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Challenge #1: Open Architecture

VoIP network is as open as the Internet

The end-points (SIP phone) of VoIP could be anywhere, and they can
freely change their location location-independent

Its likely that we will interact with some end-point that has NO prior
established trust at all

We have to allow an unknown person calls us from an unknown SIP

phone at an previously unknown IP address

We need to be able to call an unknown person with a previously

unknown IP address

How do we authenticate a caller/callee we dont know at all ?

How do we secure VoIP in such a case
Trust is NOT transitive !
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Challenge #2: Real-Time Constraints

Unlike other Internet applications such as Web, VoIP has stringent

real-time constraints

End-to-end delay should be no more than 150 ms

Each packets of voice stream should be delivered at constant (e.g.,

once every 20 ms or 30 ms): the processing of each RTP packet should
never be > 20 ms

The call setup time should not be too long: caller expect to hear ring
tone or voice within seconds after dialing

All these put an upper bound on the total time that all the security
mechanisms in all the SIP proxies, RTP servers and SIP phones can
use
More generally, QoS-critical

Packet loss, Jitter, and Latency

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Challenge #3: Multiple Protocols

VoIP involves a number of different protocols

Signaling
o

Voice stream
o

RTP, SRTP

Security
o

SIP, MGCP, H.323

HTTP digest incorporated in SIP, TLS, IPsec, Key management

In security, the whole system is as strong as its weakest point !

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Challenge #4: Key Management

How could we establish session key with someone we dont know at

all
Can we expect every SIP phone to have its own private/public keys?

How do we authenticate the public key of someone we dont know

Can we expect PKI for SIP phones ?

Scalability issue for the SIP servers

A SIP server may need to serve tens (or even hundreds) of thousands of
subscribers

Dynamically establishing and managing keys with many concurrent

callers/callees may overwhelm the SIP server

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Challenge #5: Emergency Call

How to provide authenticated caller ID ?

How to determine the real origin of the call ?

Its easy to spoof caller ID

VoIP phones are highly mobile, they can be used from anywhere on the
Internet

How to route an emergency call to the appropriate PSAP (Public

Safety Answering Point) ?
How to keep VoIP emergency call connection with PSAP ?

PSTN emergency call can only be terminated by PSAP

VoIP emergency call an be terminated by the caller through power cycle

the SIP phone

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Challenge #6: Firewall Traversal

Firewall problem with VoIP

Firewall needs to block traffic based on source, destination, and

traffic type

VoIP needs to allow unsolicited incoming calls from unknown and untrusted sources

Conflicting requirements: effects on QoS by introducing latency and

jitter

SIP-aware firewall

Allows Sip signaling message and corresponding RTP comes in

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Challenge #6: Firewall Traversal

Any exploits of SIP or RTP could compromise the security of SIPaware firewall
How to support such VoIP calls without compromising the firewall
filtering policy ?
Solutions to call setup (excessive latency is not tolerated !): applies to
NAT too

Application level gateways (ALGs, embedded software): latency,

congestion, expensive

Middlebox solutions (independent in-path system): setup cost, single

point attack

Session Border Controllers(SBC): firewall/NAT traversal, call admission

control, Service Level Agreement monitor..

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Challenge #7: NAT Traversal

NAT (Network Address Translation)

Dynamically maps internal, private IP & port with external public IP &
port

Allows multiple hosts in a private network to share one public IP

address

Most residential IP phones are behind NAT

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Challenge #7: NAT Traversal, cont

NAT problem with VoIP

NAT blocks unsolicited incoming traffic it does not know how to

translate that into private IP & port

VoIP needs to support unsolicited incoming calls from previously

unknown and un-trusted sources

SIP phones behind NAT will use its private IP for REGISTER, INVITE,
and 200 OK

SIP hones behind NAT will dynamically choose the port number for
receiving the incoming RTP stream and specify it in SDP part of
INVITE or 200 OK messages

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Challenge #7: NAT Traversal, cont

Existing NAT traversal solutions

UPnP (Universal Plug and Play): queries the NAT device, does not work
with cascading NAT

STUN (simple Traversal of UDP through NAT)

Uses TURN server on the public Internet

Works with Full Cone, Restricted Cone, and Port Restricted Cone NAT

Does NOT work with Symmetric (bidirectional) NAT

Does not support TCP (SIP RFC 3261 mandates TCP support)

Susceptible to port scan

TURN (Traversal Using Relay NAT)

o

Relies on a TURN server in the middle of the signaling and media path

Works with symmetric (bidirectional) NAT

May open door for MITM attack

ICE (Interactive Connectivity Establishment): make use of STUN and

TURN
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Challenge #7: NAT Traversal, cont

Security implications of NAT in VoIP

Makes it hard to enforce the integrity from end-to-end

Makes it hard to validate/authenticate the IP address of SIP phone

Open door for

o

Registration hijacking

Call hijacking

MITM attack

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Challenge #8: Challenge/Response with SIP Forking

Challenge/Response would prevent replay attack, but it is difficult to

work with SIP forking

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Challenge #9: SPIT (Spam on Internet Telephony)

VoIP spam could be more disturbing than email spam

People are used to respond to a phone call immediately

What if some spammer keeps ringing your phone all day long ?
o

Could make your phone virtually unusable

VoIP spam is harder to block than email spam

Caller ID is not trustworthy
Voice content based filtering is difficult

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Challenge #9: SPIT, cont

Reverse Turing test against SPIT

Before accepts a call, the callees SIP phone (or proxy) asks the caller
some questions that is easy for human to answer but difficult for machine

However, such a voice Turing test scheme could be abused to launch

DoS attack similar to Smurf attack

Attacker send INVITE to lots of VoIP subscribers with spoofed IP address

of the victim

The proxies of those VoIP subscribers will send RTP streams to the
victims

The victim could be overwhelmed by those RTP streams

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Challenge #10: Voice Phishing (aka Vishing)

Attackers started using VoIP in phishing

Voice phishing attack on PayPal

Voice phishing attack on Santa Barbara Bank & Trust

This is essentially exploits peoples trust in landline telephone

How to make VoIP as trustworthy as landline telephone ?

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New Challenge: VoIP in the Cloud

VoIP servers are migrated to the Clouds

New system architecture: in-house to outsourcing (Virtual PBX)

Easier management, less upfront capital costs (e.g., Amazon EC2),

however,

0-day vulnerabilities are introduced, leading to novel attacks

Operating/deployment environment is challenging

Virtual machines multi-tenant environments

(Multiple) Protocol-level protections are remained

Still, the trade-off between Quality of Service (QoS) and security

Service Oriented Architecture (SOA) Service Level Agreement (SLA)

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Questions

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