3. sip security

7/25/2019 3. SIP Security

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SIP Security

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Roadmap

Basics

Terms, concepts, and so on

o Threats, security attributes, vulnerabilities

Network defenseso 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 convenientand flexible, with diverse features

The number of residential VoIP subscribers worldwide is expected to grow

significantly

4.8million in 2004

~288million by 2013

Types of VoIP

Managed: Vonage, AT&T, Verizon, etc.

Unmanaged: PC-to-PC VoIP or Softphone (e.g., Skype)

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Expectations on VoIP

Confidentiality Securitythe No.1concern

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 Responsible for establishing, managing, tearing downVoIP sessions

H.3231996 ITU

MGCP (Media Gateway Control Protocol)1999 IETF RFC

SIP (Session Initiation Protocol)1999, 2002 IEFT RFC

o The dominant VoIP signaling protocol

VoIP voice stream

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

SIP Stack DiagramH.323 Stack Diagram6


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Infrastructure Attacks

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Physical

Data Link

Network

Transport

Application7

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Eavesdropping/Sniffing

For subnets using broadcast technologies such as WiFi, each attachedsystems 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

With physical access to a subnetwork, attacker can overwhelm its signaling(e.g., jamWiFisRF)

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 droptraffic 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 exactstate of victims communication and craft their spoofed traffic to match it

SIP Overview

SIP is an IETF standard

RFC 2543 of 1999 (obsolete)

RFC 3261 of 2002an 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)

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SIP Components

User Agentsowned/used by subscribers

User Agent Client (UAC)who initiatesa call

User Agent Server (UAS)who receivesa call

SIP Serversmaintained by service providers

Proxy server: relays the signaling messages (and potential voice streams) betweenthe 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

Requestidentified by a method name REGISTERtell where the UAC is currently at

INVITEInitiate a call to someone

BYEterminate an established call

ACKacknowledge the receipt of some message

CANCELquit from an ongoing call setup

OPTIONto query the capability of a server

Responseidentified by a number similar to HTTP

1Provisional100 Trying, 180 Ringing

2Successful200 OK

3Redirection301 Moved permanently, 302 Moved Temporarily

4Failure403 Not Found, 410 Gone, 403 Forbidden

5Server Failure503 Service Unavailable

6Global Failure600 Busy Everywhere11

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 fromwho it claims to becaller 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

o 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 providerno service stealing, toll fraud

Subscriberno overcharge

Availability

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

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VoIP Security: System-Centric Perspective

VoIP signaling security

Protect the authenticity, integrityof the signaling message

VoIP voice stream security

Protect the authenticity, integrityand confidentialityof thevoicecontent(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 Availability and DoS

Switch Default Password Vulnerability

Classical Wiretap Vulnerability

ARP Cache Poisoning & Floods

Web Server Interfaces

IP Phone Netmask Vulnerability

Extension to IP Address Mapping Vulnerability

DHCP Server Insertion

TFTP Server Insertion Attack

CPU Resource Consumption

Default Password

Exploitable Software Flaws

Account Lockout Vulnerability

Threats to VoIP Security

Identity and service theft

Steal minutesfrom VoIP service provider

Call at other subscribers expense

Service disruption against the following 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 packetsin the communication stream

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Threats to VoIP Security, cont

Call hijacking (Man-in-the-Middleattacks) Registration spoofing

Unauthorized call redirection

Taking over established VoIP call via re-INVITE

Interception and modification

Conversation alterationmix, 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, itmust register itself so that SIP proxy server know its current IP

address

What if some attacker send spoofedREGISTERmessage to the

registrar to trick the registrar into believing that the victim SIP phone isat an IP address chosen by the attacker

All the calls to the victim will reach attackers phone !

What about SIP digest authentication ?

The SIP registrar does require authenticationfor registration

Assuming the attacker does not know the secret password shared

between the victim SIP phone and the registrar, can attacker still spoofregistration ?YES !

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Attack#2: SIP Phone Registration Spoofing, cont

The SIP digest does NOT cover the IP address of the SIP phoneThe registrar actually uses the source IP address of the packet

containing the REGISTRER message

The attacker could simply replaythe legitimate REGISTER messagefrom different IP address

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Attack#3: SIP Phone Registration Hijacking

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

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Attack#4: Fake BYE to Terminate Established SIP Call

Attacker could terminateany established SIP call by sending fake BYEmessageto the SIP phone(s) and/or SIP proxy

SIP proxy requires digest authentication, so it could detectfake

BYEand ignore it

Existing SIP phones however will honor anyBYE messagewith

correct Call-ID

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Attack#5: VoIP CallerID Spoofing

Caller ID of SIP call is NOTtrustworthy, and it is easy to spoof

Just modify the Fromfield 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 aper minute basis: International call,

Service providers rely on accounting and billingfor charging theircustomers 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 NOTmade

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 providedandwhat 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

Inaccurateor corruptedbill 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 The callerand callee of the call ---- communication parties

When the call startsand ends ---- call duration

Where the call will be routed ---- call path

Any vulnerability inVoIP signalingcould be a potential vulnerability

of VoIP billing

Manipulation of SIP messages

Fake SIP messages

VoIP signaling protocols

SIPthe 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 VoIPservices ?

Vonage, AT&T, Verizon, Broadvoice

ReferenceR. Zhang, et al., Billing Attacks on SIP-Based VoIP Systems, inProceedings 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 INVITEmessage to its SIPserver, which tracks the INVITEmessage for billingand accounting

Attack: an attacker can replaysome captured INVITEmessage 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

Hijackthe 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 thehijacked call durationwhile keeping the caller and callee unaware of the call

This could result in overchargeon calls the VoIP subscriber has actually made

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VoIP Billing Attack #3: ByeDelayandByeDrop

Established SIP calls are terminated by BYEmessage

Can an MITM delayor simply dropthe BYEmessage of established SIP call ?

The SIP server will think the call is still alive and just count on the time, thereby

prolonging the durationof established calls

This could lead to overchargeon 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 forHTTP and SMTP

TLS or IPsec protect the SIP signaling path

S/MIME protects the integrity and confidentiality of SIP messages

No end-to-endprotection

Weakness of SIP authentication

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

while others likeTRYING, RINGING, 200 OK,ACKand 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

Just need to implement the SIP authentication correctly

Implementation errors may lead to this attack

FakeBusy

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

o Delay and drop could happen naturally (accidental anomalies)

Some heuristics might help

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Questions ?

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Roadmap

Basics



Network defenses

o Cryptogrpahic primitives


VoIP Security



Open Issues

SIP Security Mechanisms

SIP does NOT define its own security mechanism, it reusesexistingsecurity mechanisms for HTTP, SMTP whenever possible

Two building blocks of SIP security mechanisms

Authentication: HTTP digest authentication

Encryption: IPsec, TLS, S/MIME

SIP authentication and encryption can NOT be applied to thewhole SIP messages from end-to-end

Intermediate SIP proxies need to modify and insert SIP message fields:

o Add viafield

o Change request URIdue 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 !)

Anti-replay protection

Assumesthe two parties involved share a secret password

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

Uses challenge/response

A valid response contains a Checksum (by default MD5RFC 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 confidentialityfor thetransmitted 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-endas well ashop-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 automatedcryptographic 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 mandatesthe 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(serveronly) and secure key distribution

Applied hop-by-hopbetween UAS/proxies or between proxies

Requires reliabletransport 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-endauthentication, integrity andconfidentiality for mostSIP 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 recommendsS/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) Defines a generic SIP authentication token by adding an S/MIME body to a SIP message

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

S/MIME AES Requirement for SIP 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

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 (tailoringweb pages for varying devices like PDAs, cell phones)

Secure information inserted by intermediaries (proxies sometimes need to modify a message)49

Secure RTP (from IETF Proceedings)

Security Parameter Index

(recommended for frequent re-

keying in large groups), 32-bit

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Preventative Solutions to VoIP Security: Firewall

What is firewall ?

To block unauthorizedaccesswhile permitting authorized communications, e.g.,blockall FTP traffic (port 21), allowall HTTP traffic (port 80)

Packet filter (header=, 5-tuple), application

layer firewall, stateful firewall (maintainsconnection records and investigateapplication 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

52ReferenceJ. 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,

lexand yacc) can be used

Must parse the SIP exchanges in order to discover which packets

contain the media, i.e., understand the SIP exchangesand extracttherelevant information

Must be statefuland monitor SIP traffic to determine which RTPportsare 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 attemptedconnections and completed handshakes

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

Hellinger distance based anomaly detection

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 Could filter out those SIP messages with spoofed source

State machine sequencing

o Filter out-of-state SIP messages

Maintain dialogue state (source, contact)

o Only accept BYE with legitimate address

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Sipd: SIP redirect, forking proxyand registration server that provides

name mapping, user locationand

scriptingservices

ASM-- Application Server Module

DPPMDeep Packet Processing

Module

FCP-- Firewall Control Protocol

CAMContent Addressable

Memory

ReferenceE. Yardeni, H. Schulzrinne, and G. Ormazabal , Large Scale SIP-aware Application Layer Firewall, Technical Report.

Columbia Univ.

Case Study: SIP-aware Application Layer Firewall

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Case Study: SIP-aware Application Layer Firewall, cont


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Case Study: SIP-aware Application Layer Firewall, cont

NAT: Network Address Translation

What is NAT ?

Can hide internalnetwork address and enable several endpoints within a LANto share the same global IP address

Reduce theprotected points of access and simplifies network management

Full cone NAT, restricted cone NAT, port restricted cone, symmetric NAT

Widely used, the needwill notbe alleviated by IPv6

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VPN: Virtual Private Networks Tunnels between two endpoints that allow for data to be securely transmitted

between the nodes

IPsec ESP tunnelhelps 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 theencryption phase

Expanded packet size (compression of packet size)

IPsec and NAT incompatibility (UDP encapsulation, RSIP, IPv6 tunnel broker)

Preventative Solutions to VoIP Security: VPN

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What is IDSIntrusion 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 atDSN 2004, by Wu et al.)o Sending fake BYE message to only one end-point will leave an orphan RTP stream from

the other end-point

o Stateful, cross protocol correlation could detect this

o What if attacker sendsbogus BYEto both end-points ?

o 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

o What if some attack do follow the SIP state machine ?o CallerID spoofing, Registration hijacking

IDS for VoIP Network

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Questions ?

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Roadmap

Basics



Network defenseso Cryptogrpahic primitives


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 locationlocation-independent

Its likely that we will interact with some end-point that has NO prior

established trust at all

We have to allow an unknownperson calls us from an unknownSIP

phone at an previously unknownIP address

We need to be able to call an unknownperson with a previously

unknownIP 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 securitymechanisms 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 SIP, MGCP, H.323

Voice stream

o RTP, SRTP

Security

o HTTP digestincorporated 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/publickeys?

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 ?

Its easy to spoof caller ID

How to determine the real origin of the call ?

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

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 un-

trusted 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 SIP-

aware 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 hostsin 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 unsolicitedincoming trafficit does not know how to

translate that into private IP & port

VoIP needs to support unsolicited incoming calls from previouslyunknownand 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

INVITEor 200 OK messages

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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)

o Uses TURN server on the public Internet

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

o Does NOT work with Symmetric (bidirectional) NAT

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

o Susceptible to port scan

TURN (Traversal Using Relay NAT)

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

o Works with symmetric (bidirectional) NAT

o May open door for MITM attack

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

TURN73


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 foro Registration hijacking

o Call hijacking

o MITM attack

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Challenge #8: Challenge/Response with SIP Forking

Challenge/Responsewould 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 testagainst SPIT

Before accepts a call, the calleesSIP 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 attacksimilar 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-houseto outsourcing(Virtual PBX)

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

however,

0-dayvulnerabilities are introduced, leading to novel attacks

Operating/deployment environment is challenging

Virtual machinesmulti-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

3. sip security

Documents